W3C

Web Audio API

W3C Working Draft 15 March 02 August 2012

This version:
http://www.w3.org/TR/2012/WD-webaudio-20120315/ http://www.w3.org/TR/2012/WD-webaudio-20120802/
Latest published version:
http://www.w3.org/TR/webaudio/
Previous version:
http://www.w3.org/TR/2011/WD-webaudio-20111215/ http://www.w3.org/TR/2012/WD-webaudio-20120315/
Editor:
Chris Rogers, Google <crogers@google.com>

Abstract

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

The introductory section covers the motivation behind this specification.

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

Status of this Document

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 W3C technical reports index at http://www.w3.org/TR/.

This is the second third public Working Draft of the Web Audio API specification. It has been produced by the W3C Audio Working Group , which is part of the W3C WebApps Activity.

Please send comments about this document to < public-audio@w3.org > ( public archives of the W3C audio mailing list). Web content and browser developers are encouraged to review this draft.

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.

This document was produced by a group operating under the 5 February 2004 W3C Patent Policy . W3C maintains a public list of any patent disclosures 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 Essential Claim(s) must disclose the information in accordance with section 6 of the W3C Patent Policy .

Table of Contents

1. Introduction

This section is informative.

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

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

1.1. Features

The API supports these primary features:

1.2. Modular Routing

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

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

modular routing

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

ECMAScript
 
            var context = new AudioContext();
            function playSound() {
                var source = context.createBufferSource();
                source.buffer = dogBarkingBuffer;
                source.connect(context.destination);
                source.noteOn(0);
            }

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

modular routing2

TODO: add Javascript example code here ...

1.3. API Overview

The interfaces defined are:

2. Conformance

Everything in this specification is normative except for examples and sections marked as being informative.

The keywords “ MUST ”, “ MUST NOT ”, “ REQUIRED ”, “ SHALL ”, “ SHALL NOT ”, “ RECOMMENDED ”, “ MAY ” and “ OPTIONAL ” in this document are to be interpreted as described in Key words for use in RFCs to Indicate Requirement Levels [RFC2119] .

The following conformance classes are defined by this specification:

conforming implementation

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

3. Terminology and Algorithms

This specification includes algorithms (steps) as part of the definition of methods. Conforming implementations (referred to as "user agents" from here on) MAY use other algorithms in the implementation of these methods, provided the end result is the same.

4. The Audio API

4.1. The AudioContext Interface

This interface represents a set of AudioNode objects and their connections. It allows for arbitrary routing of signals to the AudioDestinationNode (what the user ultimately hears). Nodes are created from the context and are then connected together. In most use cases, only a single AudioContext is used per document. An AudioContext is constructed as follows:

        var context = new AudioContext();
// For implementation WebKit this will be: var context = new webkitAudioContext();
IDL
 
        interface AudioContext {
            readonly attribute AudioDestinationNode destination;
            readonly attribute float sampleRate;
            readonly attribute float currentTime;
            readonly attribute AudioListener listener;
            readonly attribute unsigned long activeSourceCount;

            AudioBuffer createBuffer(in unsigned long numberOfChannels, in unsigned long length, in float sampleRate);
            AudioBuffer createBuffer(in ArrayBuffer buffer, in boolean mixToMono);

            AudioBuffer createBuffer(in unsigned long numberOfChannels, in unsigned long length, in float sampleRate)
                raises(DOMException);
            AudioBuffer createBuffer(in ArrayBuffer buffer, in boolean mixToMono)
                raises(DOMException);

            
            void decodeAudioData(in ArrayBuffer audioData,
                                 in [Callback] AudioBufferCallback successCallback,
                                 in [Optional, Callback] AudioBufferCallback errorCallback)
                raises(DOMException);
            
            // AudioNode creation          
            AudioBufferSourceNode createBufferSource();
            JavaScriptAudioNode createJavaScriptNode(in short bufferSize, in short numberOfInputs, in short numberOfOutputs);

            MediaElementAudioSourceNode createMediaElementSource(in HTMLMediaElement mediaElement)
                raises(DOMException);
            MediaStreamAudioSourceNode createMediaStreamSource(in MediaStream mediaStream)
                raises(DOMException);
            JavaScriptAudioNode createJavaScriptNode(in unsigned long bufferSize,
                                                     in [Optional] unsigned long numberOfInputChannels = 2,
                                                     in [Optional] unsigned long numberOfOutputChannels = 2)
                raises(DOMException);

            RealtimeAnalyserNode createAnalyser();
            AudioGainNode createGainNode();
            DelayNode createDelayNode(in [Optional] double maxDelayTime);
            BiquadFilterNode createBiquadFilter();
            AudioPannerNode createPanner();
            ConvolverNode createConvolver();
            AudioChannelSplitter createChannelSplitter();
            AudioChannelMerger createChannelMerger();

            AudioChannelSplitter createChannelSplitter(in [Optional] unsigned long numberOfOutputs = 6)
                raises(DOMException);
            AudioChannelMerger createChannelMerger(in [Optional] unsigned long numberOfInputs = 6);
                raises(DOMException);

            DynamicsCompressorNode createDynamicsCompressor();
            Oscillator createOscillator();
            WaveTable createWaveTable(in Float32Array real, in Float32Array imag)
                raises(DOMException);

        }

4.1.1. Attributes

destination

An AudioDestinationNode with a single input representing the final destination for all audio (to be rendered to the audio hardware). All AudioNodes actively rendering audio will directly or indirectly connect to destination .

sampleRate

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.

currentTime

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.

listener

An AudioListener which is used for 3D spatialization .

activeSourceCount

The number of AudioBufferSourceNodes that are currently playing.

4.1.2. Methods and Parameters

The createBuffer method

Creates an AudioBuffer of the given size. The audio data in the buffer will be zero-initialized (silent). An exception will be thrown if the numberOfChannels or sampleRate are out-of-bounds.

The numberOfChannels parameter determines how many channels the buffer will have. An implementation must support at least 32 channels.

The length parameter determines the size of the buffer in sample-frames.

The sampleRate 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.

The createBuffer from ArrayBuffer method

Creates an AudioBuffer given the audio file data contained in the ArrayBuffer. The ArrayBuffer can, for example, be loaded from an XMLHttpRequest with the new responseType and response attributes.

The buffer parameter contains the audio file data (for example from a .wav file).

The mixToMono parameter determines if a mixdown to mono will be performed. Normally, this would not be set.

The decodeAudioData method

Asynchronously decodes the audio file data contained in the ArrayBuffer. The ArrayBuffer can, for example, be loaded from an XMLHttpRequest with the new responseType and response attributes. Audio file data can be in any of the formats supported by the audio element.

The decodeAudioData() method is preferred over the createBuffer() from ArrayBuffer method because it is asynchronous and does not block the main JavaScript thread.

audioData is an ArrayBuffer containing audio file data.

successCallback 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.

errorCallback is a callback function which will be invoked if there is an error decoding the audio file data.

The createBufferSource method

Creates an AudioBufferSourceNode .

The createMediaElementSource method

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

The createMediaStreamSource method

Creates a MediaStreamAudioSourceNode 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.

The createJavaScriptNode method

Creates a JavaScriptAudioNode for direct audio processing using JavaScript. An exception will be thrown if bufferSize or numberOfInputChannels or numberOfOutputChannels are outside the valid range.

The bufferSize parameter determines the buffer size in units of sample-frames. It must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently the onaudioprocess event handler is called and how many sample-frames need to be processed each call. Lower values for bufferSize will result in a lower (better) latency . Higher values will be necessary to avoid audio breakup and glitches . The value chosen must carefully balance between latency and audio quality.

The numberOfInputs numberOfInputChannels parameter (defaults to 2) and determines the number of inputs. channels for this node's input. Values of up to 32 must be supported.

The numberOfOutputs numberOfOutputChannels parameter (defaults to 2) and determines the number of outputs. channels for this node's output. Values of up to 32 must be supported.

It is invalid for both numberOfInputs numberOfInputChannels and numberOfOutputs numberOfOutputChannels to be zero.

The createAnalyser method

Creates a RealtimeAnalyserNode .

The createGainNode method

Creates an AudioGainNode .

The createDelayNode method

Creates a DelayNode representing a variable delay line. The initial default delay time will be 0 seconds.

The maxDelayTime parameter is optional and specifies the maximum delay time allowed for the delay line. If not specified, the maximum delay time defaults to 1 second.

The createBiquadFilter method

Creates a BiquadFilterNode representing a second order filter which can be configured as one of several common filter types.

The createPanner method

Creates an AudioPannerNode .

The createConvolver method

Creates a ConvolverNode .

The createChannelSplitter method

Creates an AudioChannelSplitter representing a channel splitter. An exception will be thrown for invalid parameter values.

The numberOfOutputs parameter determines the number of outputs. Values of up to 32 must be supported. If not specified, then 6 will be used.

The createChannelMerger method

Creates an AudioChannelMerger representing a channel merger. An exception will be thrown for invalid parameter values.

The numberOfInputs parameter determines the number of inputs. Values of up to 32 must be supported. If not specified, then 6 will be used.

The createDynamicsCompressor method

Creates a DynamicsCompressorNode .

The createOscillator method

Creates an Oscillator .

The createWaveTable method

Creates a WaveTable representing a waveform containing arbitrary harmonic content. The real and imag parameters must be of type Float32Array 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 Fourier series representing the partials of a periodic waveform. The created WaveTable will be used with an Oscillator and will represent a normalized time-domain waveform having maximum absolute peak value of 1. Another way of saying this is that the generated waveform of an Oscillator 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 WaveTable will be normalized on creation, the real and imag parameters represent relative values.

The real parameter represents an array of cosine terms (traditionally the A terms). In audio terminology, the first element (index 0) is the DC-offset of the periodic waveform 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.

The imag parameter represents an array of sine terms (traditionally the B terms). The first element (index 0) should be set to zero (and will be ignored) since this term does not exist in the Fourier series. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.

4.1.3. Lifetime

Once created, an AudioContext will not be garbage collected. It will live until the document goes away.

4.2. The AudioNode Interface

AudioNodes are the building blocks of an AudioContext . This interface represents audio sources, the audio destination, and intermediate processing modules. These modules can be connected together to form processing graphs for rendering audio to the audio hardware. Each node can have inputs and/or outputs. An AudioSourceNode has no inputs and a single output. An AudioDestinationNode 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.

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

IDL
 
    interface AudioNode {
        void connect(in AudioNode destination, in unsigned long output = 0, in unsigned long input = 0);
        void disconnect(in int output = 0);

        void connect(in AudioNode destination, in [Optional] unsigned long output = 0, in [Optional] unsigned long input = 0)
            raises(DOMException);
        void connect(in AudioParam destination, in [Optional] unsigned long output = 0)
            raises(DOMException);
        void disconnect(in [Optional] unsigned long output = 0)
            raises(DOMException);

        readonly attribute AudioContext context;
        readonly attribute unsigned long numberOfInputs;
        readonly attribute unsigned long numberOfOutputs;
    }

4.2.1. Attributes

context

The AudioContext which owns this AudioNode.

numberOfInputs

The number of inputs feeding into the AudioNode. This will be 0 for an AudioSourceNode.

numberOfOutputs

The number of outputs coming out of the AudioNode. This will be 0 for an AudioDestinationNode.

4.2.2. Methods and Parameters

The connect to AudioNode method

Connects the AudioNode to another AudioNode.

The destination parameter is the AudioNode to connect to.

The output parameter is an index describing which output of the AudioNode from which to connect. An out-of-bound value throws an exception.

The input parameter is an index describing which input of the destination AudioNode to connect to. An out-of-bound value throws an exception.

It is possible to connect an AudioNode output to more than one input with multiple calls to connect(). Thus, "fanout" is supported.

It is possible to connect an AudioNode to another AudioNode which creates a cycle . 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 DelayNode in the cycle or an exception will be thrown.

The connect to AudioParam method

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

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

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

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

The destination parameter is the AudioParam to connect to.

The output parameter is an index describing which output of the AudioNode from which to connect. An out-of-bound value throws an exception.

The disconnect method

Disconnects an AudioNode's output.

The output parameter is an index describing which output of the AudioNode to disconnect. An out-of-bound value throws an exception.

4.2.3. Lifetime

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

  1. A normal JavaScript reference obeying normal garbage collection rules.
  2. A playing reference for an AudioSourceNode . Please see details for each specific AudioSourceNode sub-type. For example, both AudioBufferSourceNodes and OscillatorNodes maintain a playing reference to themselves while they are in the SCHEDULED_STATE or PLAYING_STATE.
  3. A connection reference which occurs if another AudioNode is connected to it.
  4. A tail-time reference which an AudioNode maintains on itself as long as it has any internal processing state which has not yet been emitted. For example, a ConvolverNode has a tail which continues to play even after receiving silent input (think about clapping your hands in a large concert hall and continuing to hear the sound reverberate throughout the hall). Some AudioNodes have this property. Please see details for specific nodes.

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

When an AudioNode has no references it will be deleted. But before it is deleted, the implementation must disconnect the node from any other AudioNodes which it is connected to. In this way it releases all connection references (3) it has to other nodes.

Regardless of any of the above references, an AudioNode will be deleted when its AudioContext is deleted.

4.3. The AudioSourceNode Interface

This is an abstract interface representing an audio source, an AudioNode which has no inputs and a single output:

      numberOfInputs  : 0
      numberOfOutputs : 1

Subclasses of AudioSourceNode will implement specific types of audio sources.

IDL
 
    interface AudioSourceNode : AudioNode {
    }

4.4. The AudioDestinationNode Interface

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

      numberOfInputs  : 1
      numberOfOutputs : 0
IDL
 
    interface AudioDestinationNode : AudioNode {
        readonly attribute unsigned long numberOfChannels;

        readonly attribute unsigned long maxNumberOfChannels;
        attribute unsigned long numberOfChannels;

    }

4.4.1. Attributes

maxNumberOfChannels

The maximum number of channels that the numberOfChannels attribute can be set to. An AudioDestinationNode representing the audio hardware end-point (the normal case) can potentially output more than 2 channels of audio if the audio hardware is multi-channel. maxNumberOfChannels is the maximum number of channels that this hardware is capable of supporting. If this value is 0, then this indicates that maxNumberOfChannels may not be changed. This will be the case for an AudioDestinationNode in an OfflineAudioContext .

numberOfChannels

The number of channels of the destination's input. This value will default to 2, and may be set to any non-zero value less than or equal to maxNumberOfChannels . 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 numberOfChannels to 8, and render 8-channels of output.

4.5. The AudioParam Interface

AudioParam is a parameter controlling controls an individual aspect of an AudioNode 's functioning, such as volume. The parameter can be set immediately to a particular value using the "value" attribute. Additionally, value changes can be scheduled to happen at very precise times, 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.

Some synthesis and processing AudioNodes have AudioParams as attributes whose values must be taken into account on a per-audio-sample basis. For other AudioParams , sample-accuracy is not important and the value changes can be sampled more coarsely. Each individual AudioParam will specify that it is either an a-rate parameter which means that its values must be taken into account on a per-audio-sample basis, or it is a k-rate parameter whose value changes must be taken into account at least at a 3ms resolution, but can be more precise than this.

Because practical implementations will use block processing, and will process a fixed number of sample-frames at a time ( block-size sample-frames). For each block, the value of a k-rate parameter will be sampled at the time of the very first sample-frame, and that value will be used for the entire block.

IDL
 
    interface AudioParam {
        attribute float value;
        readonly attribute float minValue;
        readonly attribute float maxValue;
        readonly attribute float defaultValue;
        readonly attribute DOMString name;
                 
        
        readonly attribute short units;

        // Parameter automation. 
        void setValueAtTime(in float value, in float time);
        void linearRampToValueAtTime(in float value, in float time);
        void exponentialRampToValueAtTime(in float value, in float time);
        // Exponentially approach the target value with a rate having the given time constant. 
        void setTargetValueAtTime(in float targetValue, in float time, in float timeConstant);
        // Sets an array of arbitrary parameter values starting at time for the given duration. 
        // The number of values will be scaled to fit into the desired duration. 
        void setValueCurveAtTime(in Float32Array values, in float time, in float duration);
        
        // Cancels all scheduled parameter changes with times greater than or equal to startTime. 
        void cancelScheduledValues(in float startTime);
    }

4.5.1. Attributes

value

The parameter's floating-point value. If a value is set outside the allowable range described by minValue and maxValue an no exception is thrown. thrown, because these limits are just nominal and may be exceeded.

minValue

Minimum Nominal minimum value. The value attribute must not may be set lower than this value.

maxValue

Maximum Nominal maximum value. The value attribute must may be set lower higher than this value.

defaultValue

Initial value for the value attribute

4.5.2. Methods and Parameters

An name AudioParam maintains a time-ordered event list which is initially empty. The name 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 parameter. event list by adding a new event into the list of a type specific to the method. Each event has a time associated with it, and the events will always be kept in time-order in the list. These methods will be called automation methods:

Represents The following rules will apply when calling these methods:

The setValueAtTime method

Schedules a parameter value change at the given time (relative to the AudioContext .currentTime). time.

The value parameter is the value the parameter will change to at the given time.

The time parameter is the time (relative to in the AudioContext .currentTime). same time coordinate system as AudioContext.currentTime.

If there are no more events after this SetValue event, then for t > = time, v(t) = value. In other words, the value will remain constant.

If the next event (having time T1) after this SetValue event is not of type LinearRampToValue or ExponentialRampToValue , then, for t: time <= t < T1, v(t) = value. In other words, the value will remain constant during this time interval, allowing the creation of "step" functions.

If the next event after this SetValue event is of type LinearRampToValue or ExponentialRampToValue then please see details below.

The linearRampToValueAtTime method

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

The value parameter is the value the parameter will linearly ramp to at the given time.

The time parameter is the time (relative to in the AudioContext .currentTime). same time coordinate system as AudioContext.currentTime.

The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the time parameter passed into this method) will be calculated as:


      v(t) = V0 + (V1 - V0) * ((t - T0) / (T1 - T0))

Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.

If there are no more events after this LinearRampToValue event then for t > = T1, v(t) = V1

The exponentialRampToValueAtTime method

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

The value 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.

The time parameter is the time (relative to in the AudioContext .currentTime). same time coordinate system as AudioContext.currentTime.

The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the time parameter passed into this method) will be calculated as:


      v(t) = V0 * (V1 / V0) ^ ((t - T0) / (T1 - T0))

Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.

If there are no more events after this ExponentialRampToValue event then for t > = T1, v(t) = V1

The setTargetValueAtTime method

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.

The targetValue parameter is the value the parameter will *start* start changing to at the given time.

The time parameter is the time (relative to in the AudioContext .currentTime). same time coordinate system as AudioContext.currentTime.

The timeConstant 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.

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

During the time interval: T0 <= t < T1 , where T0 is the time parameter and T1 represents the time of the event following this event (or infinity if there are no following events):


      v(t) = V1 + (V0 - V1) * exp(-(t - T0) / timeConstant)

Where V0 is the initial value (the .value attribute) at T0 (the time parameter) and V1 is equal to the targetValue parameter.

The setValueCurveAtTime method

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.

The values parameter is a Float32Array representing a parameter value curve. These values will apply starting at the given time and lasting for the given duration.

The time parameter is the starting time for the curve values (relative to in the AudioContext .currentTime). same time coordinate system as AudioContext.currentTime.

The duration parameter is the time-constant value amount of first-order filter (exponential) approach time in seconds (after the time parameter) where values will be calculated according to the target value. values parameter..

During the time interval: time <= t < time + duration , values will be calculated:

      v(t) = values[N * (t - time) / duration], where N is the length of the values array.
The cancelScheduledValues method

Cancels all scheduled parameter changes with times greater than or equal to startTime.

The startTime 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.

4.5.3. AudioParam Automation Example

Example
AudioParam automation
ECMAScript
 
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);

4.6. AudioGain

This interface is a particular type of AudioParam which specifically controls the gain (volume) of some aspect of the audio processing. The unit type is "linear gain". The nominal minValue is 0.0, and although the 0, but may be set negative for phase inversion. The nominal maxValue is 1.0, 1, but higher values are allowed (no exception thrown).

IDL
 
  interface AudioGain : AudioParam {
  };

4.7. The AudioGainNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

which changes the gain of (scales) multiplies the incoming input audio signal by the (possibly time-varying) gain 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 certain amount. The default amount constant gain change of 1.

As with other AudioParams , the gain 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 1.0 (no multiplied by the gain change). parameter's value for the specific time corresponding to that audio sample. This multiplied value represents the PCM audio sample for the output.

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 AudioGainNode gain is one values and being copied into the corresponding channel of the building blocks for creating mixers . output.

The implementation must make gain changes to the audio stream smoothly, without introducing noticeable clicks or glitches. This process is called "de-zippering".

IDL
 
    interface AudioGainNode : AudioNode {
        AudioGain gain;
    }

4.7.1. Attributes

gain

An AudioGain object representing the amount of gain to apply. The default value ( gain.value ) is 1.0 1 (no gain change). See AudioGain for more information. This parameter is a-rate

4.8. The DelayNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

which delays the incoming audio signal by a certain amount. The default amount is 0.0 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.

IDL
 
    interface DelayNode : AudioNode {
        AudioParam delayTime;
    }

4.8.1. Attributes

delayTime

An AudioParam object representing the amount of delay (in seconds) to apply. The default value ( delayTime.value ) is 0.0 0 (no delay). The minimum value is 0.0 0 and the maximum value is currently 1.0 (but this determined by the maxDelayTime argument to the AudioContext method createDelayNode . This parameter is arbitrary and could be increased). k-rate

4.9. The AudioBuffer Interface

This interface represents a memory-resident audio asset (for one-shot sounds and other short audio clips). Its format is non-interleaved IEEE 32-bit linear PCM with a nominal range of -1.0 -1 -> +1.0. +1. It can contain one or more channels. It is analogous to a WebGL texture. Typically, it would be expected that the length of the PCM data would be fairly short (usually somewhat less than a minute). For longer sounds, such as music soundtracks, streaming should be used with the audio element and MediaElementAudioSourceNode .

An AudioBuffer may be used by one or more AudioContexts.

IDL
 
    interface AudioBuffer {
                 
        attribute AudioGain gain;
        

        readonly attribute float sampleRate;
        readonly attribute long length;
        // in seconds          
        readonly attribute float duration;  
        readonly attribute int numberOfChannels;
        Float32Array getChannelData(in unsigned long channel);
    }

4.9.1. Attributes

gain The amount of gain to apply when using this buffer in any AudioBufferSourceNode . The default value is 1.0.
sampleRate

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

length

Length of the PCM audio data in sample-frames.

duration

Duration of the PCM audio data in seconds.

numberOfChannels

The number of discrete audio channels.

4.9.2. Methods and Parameters

The getChannelData method

Gets direct access to Returns the Float32Array representing the PCM audio data stored in an AudioBuffer. for the specific channel.

The channel parameter is an index representing the particular channel to get data for. An index value of 0 represents the first channel. This index value MUST be less than numberOfChannels or an exception will be thrown.

4.10. The AudioBufferSourceNode Interface

This interface represents an audio source from an in-memory audio asset in an AudioBuffer . It generally will be used for short audio assets which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways). The playback state of an AudioBufferSourceNode goes through distinct stages during its lifetime in this order: UNSCHEDULED, SCHEDULED, PLAYING, FINISHED. UNSCHEDULED_STATE, SCHEDULED_STATE, PLAYING_STATE, FINISHED_STATE. The noteOn() method causes a transition from the UNSCHEDULED UNSCHEDULED_STATE to SCHEDULED state. SCHEDULED_STATE. Depending on the time argument passed to noteOn(), a transition is made from the SCHEDULED SCHEDULED_STATE to PLAYING state, PLAYING_STATE, at which time sound is first generated. Following this, a transition from the PLAYING PLAYING_STATE to FINISHED state FINISHED_STATE happens when either the buffer's audio data has been completely played (if the loop attribute is false), or when the noteOff() method has been called and the specified time has been reached. Please see more details in the noteOn() and noteOff() description. Once an AudioBufferSourceNode has reached the FINISHED state it will no longer emit any sound. Thus noteOn() and noteOff() may not be issued multiple times for a given AudioBufferSourceNode.

    numberOfInputs  : 0
    numberOfOutputs : 1
IDL
 
    interface AudioBufferSourceNode : AudioSourceNode {
        const unsigned short UNSCHEDULED_STATE = 0;
        const unsigned short SCHEDULED_STATE = 1;
        const unsigned short PLAYING_STATE = 2;
        const unsigned short FINISHED_STATE = 3;
        readonly attribute unsigned short playbackState;

        // Playback this in-memory audio asset     
        // Many sources can share the same buffer     
        attribute AudioBuffer buffer;
        readonly attribute AudioGain gain;

        attribute AudioParam playbackRate; 
        attribute boolean loop;
        void noteOn(in double when);
        void noteGrainOn(in double when, in double grainOffset, in double grainDuration);
        void noteOff(in double when);
    }

4.10.1. Attributes

buffer playbackState

Represents the audio asset The playback state, initialized to be played. UNSCHEDULED_STATE.

gain buffer

The default gain at which to play back Represents the buffer. The default gain.value is 1.0. audio asset to be played.

playbackRate

The speed at which to render the audio stream. The default playbackRate.value is 1.0. 1. This parameter is a-rate

loop

Indicates if the audio data should play in a loop.

4.10.2. Methods and Parameters

The noteOn method

Schedules a sound to playback at an exact time.

The when parameter describes at what time (in seconds) the sound should start playing. This time It is relative to the currentTime attribute of in the AudioContext. same time coordinate system as AudioContext.currentTime. If 0 is passed in for this value or if the value is less than currentTime , then the sound will start playing immediately. Either noteOn or noteGrainOn (but not both) may only be called one time and must be called before noteOff is called or an exception will be thrown.

The noteGrainOn method

Schedules a portion of a sound to playback at an exact time.

The when parameter describes at what time (in seconds) the sound should start playing. This time It is relative to the currentTime attribute of in the AudioContext. same time coordinate system as AudioContext.currentTime. If 0 is passed in for this value or if the value is less than currentTime , then the sound will start playing immediately.

The grainOffset parameter describes the offset in the buffer (in seconds) for the portion to be played.

The grainDuration parameter describes the duration of the portion (in seconds) to be played. Either noteOn or noteGrainOn (but not both) may only be called one time and must be called before noteOff is called or an exception will be thrown.

The noteOff method

Schedules a sound to stop playback at an exact time.

The when parameter describes at what time (in seconds) the sound should stop playing. This time It is relative to the currentTime attribute of in the AudioContext. same time coordinate system as AudioContext.currentTime. If 0 is passed in for this value or if the value is less than currentTime , then the sound will stop playing immediately. noteOff must only be called one time and only after a call to noteOn or noteOff , or an exception will be thrown.

4.11. The MediaElementAudioSourceNode Interface

This interface represents an audio source from an audio or video element. The element's audioSource attribute implements this.

    numberOfInputs  : 0
    numberOfOutputs : 1
IDL
 
    interface MediaElementAudioSourceNode : AudioSourceNode {
    }

4.12. The JavaScriptAudioNode Interface

This interface is an AudioNode which can generate, process, or analyse audio directly using JavaScript.

    numberOfInputs  : 1
    numberOfOutputs : 1

The JavaScriptAudioNode is constructed with a bufferSize which must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently the onaudioprocess event handler is called and how many sample-frames need to be processed each call. Lower numbers for bufferSize will result in a lower (better) latency . Higher numbers will be necessary to avoid audio breakup and glitches . The value chosen must carefully balance between latency and audio quality.

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

    var node = context.createJavaScriptNode(bufferSize, numberOfInputChannels, numberOfOutputChannels);
IDL
 
    interface JavaScriptAudioNode : AudioNode {
        attribute EventListener onaudioprocess;
        
        readonly attribute long bufferSize;
    }

4.12.1. Attributes

onaudioprocess

An event listener which is called periodically for audio processing. An event of type AudioProcessingEvent will be passed to the event handler.

bufferSize

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

4.13. The AudioProcessingEvent Interface

This interface is a type of Event which is passed to the onaudioprocess event handler used by JavaScriptAudioNode .

The event handler processes audio from the input (if any) by accessing the audio data from the inputBuffer attribute. The audio data which is the result of the processing (or the synthesized data if there are no inputs) is then placed into the outputBuffer .

IDL
 
    interface AudioProcessingEvent : Event {
        JavaScriptAudioNode node;
        readonly attribute float playbackTime;
        readonly attribute AudioBuffer inputBuffer;
        readonly attribute AudioBuffer outputBuffer; 
    }

4.13.1. Attributes

node

The JavaScriptAudioNode associated with this processing event.

playbackTime

The time when the audio will be played. This time is played in relation to the context's currentTime attribute. same time coordinate system as AudioContext.currentTime. playbackTime allows for very tight synchronization between processing directly in JavaScript with the other events in the context's rendering graph.

inputBuffer

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

outputBuffer

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

4.14. The AudioPannerNode Interface

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

numberOfInputs : 1
    numberOfInputs  : 1

    numberOfOutputs : 1

The audio stream from the input will be either mono or stereo, depending on the connection(s) to the input.

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

IDL
 
    interface AudioPannerNode : AudioNode {
        // Panning model          
        const unsigned short EQUALPOWER = 0;
        const unsigned short HRTF = 1;
        const unsigned short SOUNDFIELD = 2;
        
        // Distance model 
        const unsigned short LINEAR_DISTANCE = 0;
        const unsigned short INVERSE_DISTANCE = 1;
        const unsigned short EXPONENTIAL_DISTANCE = 2;
        // Default for stereo is HRTF 
        attribute unsigned short panningModel;
        // Uses a 3D cartesian coordinate system 
        void setPosition(in float x, in float y, in float z);
        void setOrientation(in float x, in float y, in float z);
        void setVelocity(in float x, in float y, in float z);
        // Distance model and attributes          
        attribute unsigned short distanceModel;
        attribute float refDistance;
        attribute float maxDistance;
        attribute float rolloffFactor;
        // Directional sound cone          
        attribute float coneInnerAngle;
        attribute float coneOuterAngle;
        attribute float coneOuterGain;
        // Dynamically calculated gain values          
        readonly attribute AudioGain coneGain;
        readonly attribute AudioGain distanceGain;
    };

4.14.1. Constants

EQUALPOWER

A simple and efficient spatialization algorithm using equal-power panning.

HRTF

A higher quality spatialization algorithm using a convolution with measured impulse responses from human subjects. This panning method renders stereo output.

SOUNDFIELD

An algorithm which spatializes multi-channel audio using sound field algorithms.

LINEAR_DISTANCE

A linear distance model as defined in the OpenAL specification. which calculates distanceGain according to:

1 - rolloffFactor * (distance - refDistance) / (maxDistance - refDistance)
INVERSE_DISTANCE

An inverse distance model as defined in the OpenAL specification. which calculates distanceGain according to:

refDistance / (refDistance + rolloffFactor * (distance - refDistance))
EXPONENTIAL_DISTANCE

An exponential distance model as defined in the OpenAL specification. which calculates distanceGain according to:

pow(distance / refDistance, -rolloffFactor)

4.14.2. Attributes

listener

Represents the listener whose position and orientation is used together with the panner's position and orientation to determine how the audio will be spatialized.

panningModel

Determines which spatialization algorithm will be used to position the audio in 3D space. See the constants for the available choices. The default is HRTF .

distanceModel

Determines which algorithm will be used to reduce the volume of an audio source as it moves away from the listener.

refDistance

A reference distance for reducing volume as source move further from the listener.

maxDistance

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

rolloffFactor

Describes how quickly the volume is reduced as source moves away from listener.

coneInnerAngle

A parameter for directional audio sources, this is an angle, inside of which there will be no volume reduction.

coneOuterAngle

A parameter for directional audio sources, this is an angle, outside of which the volume will be reduced to a constant value of coneOuterGain .

coneOuterGain

A parameter for directional audio sources, this is the amount of volume reduction outside of the coneOuterAngle .

4.14.3. Methods and Parameters

The setPosition method

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

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

The setOrientation method

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

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

The setVelocity method

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 x, y, z parameters describe a direction vector indicating direction of travel and intensity.

4.15. The AudioListener Interface

This interface represents the position and orientation of the person listening to the audio scene. All AudioPannerNode objects spatialize in relation to the AudioContext's listener . See this section for more details about spatialization.

IDL
 
    interface AudioListener {
                 
        attribute float gain;
                 

        // same as OpenAL (default 1)          
        attribute float dopplerFactor;
        // in meters / second (default 343.3)          
        attribute float speedOfSound;
        // Uses a 3D cartesian coordinate system 
        void setPosition(in float x, in float y, in float z);
        void setOrientation(in float x, in float y, in float z, in float xUp, in float yUp, in float zUp);
        void setVelocity(in float x, in float y, in float z);
    };

4.15.1. Attributes

gain A linear gain used in conjunction with AudioPannerNode objects when spatializing.
dopplerFactor

A constant used to determine the amount of pitch shift to use when rendering a doppler effect.

speedOfSound

The speed of sound used for calculating doppler shift. The default value is 343.3 meters / second.

4.15.2. Methods and Parameters

The setPosition method

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

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

The setOrientation method

Describes which direction the listener is pointing in the 3D cartesian coordinate space. Both a front vector and an up vector are provided.

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

The xUp, yUp, zUp parameters represent an up direction vector in 3D space.

The setVelocity method

Sets the velocity vector of the listener. This vector controls both the direction of travel and the speed in 3D space. This velocity relative an audio source's velocity is used to determine how much doppler shift (pitch change) to apply.

The x, y, z parameters describe a direction vector indicating direction of travel and intensity.

4.16. The ConvolverNode Interface

This interface represents a processing node which applies a linear convolution effect given an impulse response. Normative requirements for multi-channel convolution matrixing are described here .

    numberOfInputs  : 1
    numberOfOutputs : 1
IDL
 
    interface ConvolverNode : AudioNode {
        

        attribute AudioBuffer buffer;
        attribute boolean normalize;
        // attribute ImpulseResponse response;

    };

4.16.1. Attributes

buffer

A mono mono, stereo, or multi-channel audio buffer 4-channel AudioBuffer containing the (possibly multi-channel) impulse response used by the convolver. ConvolverNode. At the time when this attribute is set, the buffer and the state of the normalize attribute will be used to configure the ConvolverNode with this impulse response having the given normalization.

normalize

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

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

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

 
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;
}

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

4.17. The RealtimeAnalyserNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1    

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

IDL
 
    interface RealtimeAnalyserNode : AudioNode {
        // Real-time frequency-domain data         
        void getFloatFrequencyData(in Float32Array array);
        void getByteFrequencyData(in Uint8Array array);
        // Real-time waveform data         
        void getByteTimeDomainData(in Uint8Array array);
        attribute unsigned long fftSize;
        readonly attribute unsigned long frequencyBinCount;
        attribute float minDecibels;
        attribute float maxDecibels;
        attribute float smoothingTimeConstant;
    };

4.17.1. Attributes

fftSize

The size of the FFT used for frequency-domain analsis. analysis. This must be a power of two.

frequencyBinCount

Half the FFT size.

minDecibels

The minimum power value in the scaling range for the FFT analysis data for conversion to unsigned byte values.

maxDecibels

The maximum power value in the scaling range for the FFT analysis data for conversion to unsigned byte values.

smoothingTimeConstant

A value from 0.0 0 -> 1.0 1 where 0.0 0 represents no time averaging with the last analysis frame.

4.17.2. Methods and Parameters

The getFloatFrequencyData method

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.

The array parameter is where frequency-domain analysis data will be copied.

The getByteFrequencyData method

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.

The array parameter is where frequency-domain analysis data will be copied.

The getByteTimeDomainData method

Copies the current time-domain (waveform) data into the passed unsigned byte array. If the array has fewer elements than the frequencyBinCount, the excess elements will be dropped.

The array parameter is where time-domain analysis data will be copied.

4.18. The AudioChannelSplitter Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 6 // number of "active" (non-silent) outputs is determined by number of channels in the input

    numberOfOutputs : Variable N (defaults to 6) // number of "active" (non-silent) outputs is determined by number of channels in the input

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 AudioChannelSplitter 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 6 outputs, supporting up N outputs (determined by the numberOfOutputs parameter to 5.1 output (note: this upper limit of the AudioContext method createChannelSplitter() ), The default number is 6 if this value is arbitrary and could be increased to support 7.2, and higher). not provided. Any outputs which are not "active" will output silence and would typically not be connected to anything.

Example:

channel splitter

Please note that in this example, the splitter does not interpret the channel identities (such as left, right, etc.), but simply splits out channels in the order that they are input.

One application for AudioChannelSplitter is for doing "matrix mixing" where individual gain control of each channel is desired.

IDL
 
    interface AudioChannelSplitter : AudioNode {
    };

4.19. The AudioChannelMerger Interface

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

numberOfInputs : 6 // number of connected inputs may be less than this
    numberOfInputs  : Variable N (default to 6)  // number of connected inputs may be less than this
    numberOfOutputs : 1

This interface represents an AudioNode for combining channels from multiple audio streams into a single audio stream. It has 6 inputs, 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 AudioChannelMerger 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.

Example:

channel merger

Please note that in this example, the merger does not interpret the channel identities (such as left, right, etc.), but simply combines channels in the order that they are input.

Be aware that it is possible to connect an AudioChannelMerger in such a way that it outputs an audio stream with a large number of channels greater than the maximum supported by the system (currently 6 channels for 5.1). audio hardware. In this case, if the case where such an output is connected to anything else the AudioContext .destination (the audio hardware), then an exception the extra channels will be thrown indicating an error condition. ignored. Thus, the AudioChannelMerger should be used in situations where the numbers number of input channels is well understood.

IDL
 
    interface AudioChannelMerger : AudioNode {
    };

4.20. The DynamicsCompressorNode Interface

DynamicsCompressorNode is an AudioNode processor implementing a dynamics compression effect.

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

    numberOfInputs  : 1
    numberOfOutputs : 1
IDL
    			 
    interface DynamicsCompressorNode : 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
    }
        

4.20.1. Attributes

All parameters are k-rate

threshold

The decibel value above which the compression will start taking effect.

knee

A decibel value representing the range above the threshold where the curve smoothly transitions to the "ratio" portion.

ratio

the decibel value above which the compression will start taking effect. The amount of dB change in input for a 1 dB change in output.

reduction

A read-only decibel value for metering purposes, representing the current amount of gain reduction that the compressor is applying to the signal.

attack

The amount of time to reduce the gain by 10dB.

release

The amount of time to increase the gain by 10dB.

4.21. The BiquadFilterNode Interface

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

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

    numberOfInputs  : 1
    numberOfOutputs : 1
IDL
 
    interface BiquadFilterNode : AudioNode {
        // Filter type.
        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;
        attribute unsigned short type;
        readonly attribute AudioParam frequency; // in Hertz
        readonly attribute AudioParam Q; // Quality factor
        readonly attribute AudioParam gain; // in Decibels
        void getFrequencyResponse(in Float32Array frequencyHz,
                                  in Float32Array magResponse,
                                  in Float32Array phaseResponse);
    }

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 reference by Robert Bristow-Johnson.

All parameters are k-rate

4.21.1 LOWPASS

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

frequency
The cutoff frequency above which the frequencies are attenuated
Q
Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked.
gain
Not used in this filter type

4.21.2 HIGHPASS

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

frequency
The cutoff frequency below which the frequencies are attenuated
Q
Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked.
gain
Not used in this filter type

4.21.3 BANDPASS

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

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

4.21.4 LOWSHELF

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

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

4.21.5 HIGHSHELF

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

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

4.21.6 PEAKING

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

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

4.21.7 NOTCH

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

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

4.21.8 ALLPASS

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

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

4.21.9. Methods

The getFrequencyResponse method

Given the current filter parameter settings, calculates the frequency response for the specified frequencies.

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

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

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

4.22. The WaveShaperNode Interface

WaveShaperNode is an AudioNode processor implementing non-linear distortion effects.

Non-linear waveshaping distortion is commonly used for both subtle non-linear warming, or more obvious distortion effects. Arbitrary non-linear shaping curves may be specified.

    numberOfInputs  : 1
    numberOfOutputs : 1
IDL
 
    interface WaveShaperNode : AudioNode {
        attribute Float32Array curve;
    }

4.22.1. Attributes

curve

The shaping curve used for the waveshaping effect. The input signal is nominally within the range -1 -> +1. Each input sample within this range will index into the shaping curve with a signal level of zero corresponding to the center value of the curve array. Any sample value less than -1 will correspond to the first value in the curve array. Any sample value less greater than +1 will correspond to the last value in the curve array.

4.23. The Oscillator Interface

Oscillator 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 WaveTable object.

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

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

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

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

    numberOfInputs  : 0
    numberOfOutputs : 1
IDL
 
    interface Oscillator : AudioSourceNode {
        // Type constants.
        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;
        attribute unsigned short type;
        
        const unsigned short UNSCHEDULED_STATE = 0;
        const unsigned short SCHEDULED_STATE = 1;
        const unsigned short PLAYING_STATE = 2;
        const unsigned short FINISHED_STATE = 3;
        readonly attribute unsigned short playbackState;
        readonly attribute AudioParam frequency; // in Hertz
        readonly attribute AudioParam detune; // in Cents
        void noteOn(in double when);
        void noteOff(in double when);
        void setWaveTable(in WaveTable waveTable);
    }

4.23.1. Attributes

type

The shape of the periodic waveform. It may directly be set to any of the type constant values except for CUSTOM. The setWaveTable() method can be used to set a custom waveform, which results in this attribute being set to CUSTOM.

playbackState

defined as in AudioBufferSourceNode .

frequency

The frequency (in Hertz) of the periodic waveform. This parameter is a-rate

detune

A detuning value (in Cents) which will offset the frequency by the given amount. This parameter is a-rate

4.23.2. Methods and Parameters

The setWaveTable method

Sets an arbitrary custom periodic waveform given a WaveTable .

The noteOn method

defined as in AudioBufferSourceNode .

The noteOff method

defined as in AudioBufferSourceNode .

4.24. The WaveTable Interface

WaveTable represents an arbitrary periodic waveform to be used with an Oscillator . Please see createWaveTable() and setWaveTable() and for more details.

IDL
 
    interface WaveTable {
    }

4.25. The MediaStreamAudioSourceNode Interface

This interface represents an audio source from a MediaStream . The first AudioMediaStreamTrack from the MediaStream will be used as a source of audio.

    numberOfInputs  : 0
    numberOfOutputs : 1
IDL
 
    interface MediaStreamAudioSourceNode : AudioSourceNode {
    }

5. Integration with the audio and video elements

A MediaElementAudioSourceNode can be created from an HTMLMediaElement using an AudioContext method.

ECMAScript
 
var mediaElement = document.getElementById('mediaElementID');
var sourceNode = context.createMediaElementSource(mediaElement);
sourceNode.connect(filterNode);

6. Mixer Gain Structure

Background

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

Summing Inputs

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

unity gain summing junction

In cases where the channel layouts of the outputs do not match, an up-mix will occur to the highest number of channels.

Gain Control

But many times, it's important to be able to control the gain for each of the output signals. The AudioGainNode gives this control:

mixer architecture new

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

Example: Mixer with Send Busses

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

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

mixer gain structure

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 AudioGainNode is used:

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

Example
ECMAScript
 
var context = 0;
var compressor = 0;
var reverb = 0;
var delay = 0;
var s1 = 0;
var s2 = 0;
var source1 = 0;
var source2 = 0;
var g1_1 = 0;
var g2_1 = 0;
var g3_1 = 0;
var g1_2 = 0;
var g2_2 = 0;
var g3_2 = 0;
// Setup routing graph 
function setupRoutingGraph() {
    context = new AudioContext();
    compressor = context.createDynamicsCompressor();
    // Send1 effect 
    reverb = context.createConvolver();
    // Convolver impulse response may be set here or later 
    // Send2 effect 
    delay = context.createDelayNode();
    // Connect final compressor to final destination 
    compressor.connect(context.destination);
    // Connect sends 1 & 2 through effects to main mixer 
    s1 = context.createGainNode();
    reverb.connect(s1);
    s1.connect(compressor);
    
    s2 = context.createGainNode();
    delay.connect(s2);
    s2.connect(compressor);
    // Create a couple of sources 
    source1 = context.createBufferSource();
    source2 = context.createBufferSource();
    source1.buffer = manTalkingBuffer;
    source2.buffer = footstepsBuffer;
    // Connect source1 
    g1_1 = context.createGainNode();
    g2_1 = context.createGainNode();
    g3_1 = context.createGainNode();
    source1.connect(g1_1);
    source1.connect(g2_1);
    source1.connect(g3_1);
    g1_1.connect(compressor);
    g2_1.connect(reverb);
    g3_1.connect(delay);
    // Connect source2 
    g1_2 = context.createGainNode();
    g2_2 = context.createGainNode();
    g3_2 = context.createGainNode();
    source2.connect(g1_2);
    source2.connect(g2_2);
    source2.connect(g3_2);
    g1_2.connect(compressor);
    g2_2.connect(reverb);
    g3_2.connect(delay);
    // We now have explicit control over all the volumes g1_1, g2_1, ..., s1, s2 
    g2_1.gain.value = 0.2;  // For example, set source1 reverb gain 
     // Because g2_1.gain is of type "AudioGain" which is an "AudioParam", 
     // an automation curve could also be attached to it. 
     // A "mixing board" UI could be created in canvas or WebGL controlling these gains. 
}


7. Dynamic Lifetime

Background

This section is informative. Please see AudioContext lifetime and AudioNode lifetime for normative requirements

In addition to allowing the creation of static routing configurations, it should also be possible to do custom effect routing on dynamically allocated voices which have a limited lifetime. For the purposes of this discussion, let's call these short-lived voices "notes". Many audio applications incorporate the ideas of notes, examples being drum machines, sequencers, and 3D games with many one-shot sounds being triggered according to game play.

In a traditional software synthesizer, notes are dynamically allocated and released from a pool of available resources. The note is allocated when a MIDI note-on message is received. It is released when the note has finished playing either due to it having reached the end of its sample-data (if non-looping), it having reached a sustain phase of its envelope which is zero, or due to a MIDI note-off message putting it into the release phase of its envelope. In the MIDI note-off case, the note is not released immediately, but only when the release envelope phase has finished. At any given time, there can be a large number of notes playing but the set of notes is constantly changing as new notes are added into the routing graph, and old ones are released.

The audio system automatically deals with tearing-down the part of the routing graph for individual "note" events. A "note" is represented by an AudioBufferSourceNode , which can be directly connected to other processing nodes. When the note has finished playing, the context will automatically release the reference to the AudioBufferSourceNode , which in turn will release references to any nodes it is connected to, and so on. The nodes will automatically get disconnected from the graph and will be deleted when they have no more references. Nodes in the graph which are long-lived and shared between dynamic voices can be managed explicitly. Although it sounds complicated, this all happens automatically with no extra JavaScript handling required.

Example

Example
dynamic allocation

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:

ECMAScript
 
	var context = 0;
	var compressor = 0;
	var gainNode1 = 0;
	var streamingAudioSource = 0;

var context = 0;
var compressor = 0;
var gainNode1 = 0;
var streamingAudioSource = 0;

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

// Initial setup of the "long-lived" part of the routing graph  
function setupAudioContext() {
    context = new AudioContext();

	    compressor = context.createDynamicsCompressor();
	    gainNode1 = context.createGainNode();

    compressor = context.createDynamicsCompressor();
    gainNode1 = context.createGainNode();

        // Create a streaming audio source.
	    var audioElement = document.getElementById('audioTagID');
	    streamingAudioSource = context.createMediaElementSource(audioElement);
	    streamingAudioSource.connect(gainNode1);

    // Create a streaming audio source.
    var audioElement = document.getElementById('audioTagID');
    streamingAudioSource = context.createMediaElementSource(audioElement);
    streamingAudioSource.connect(gainNode1);

	    gainNode1.connect(compressor);
	    compressor.connect(context.destination);
	}

    gainNode1.connect(compressor);
    compressor.connect(context.destination);
}

	
	
	function playSound() {
	    var oneShotSound = context.createBufferSource();
	    oneShotSound.buffer = dogBarkingBuffer;

// Later in response to some user action (typically mouse or key event) 
// a one-shot sound can be played. 
function playSound() {
    var oneShotSound = context.createBufferSource();
    oneShotSound.buffer = dogBarkingBuffer;

	    
	    var lowpass = context.createLowPass2Filter();
	    var panner = context.createPanner();
	    var gainNode2 = context.createGainNode();

    // Create a filter, panner, and gain node. 
    var lowpass = context.createBiquadFilter();
    var panner = context.createPanner();
    var gainNode2 = context.createGainNode();

	    
	    oneShotSound.connect(lowpass);
	    lowpass.connect(panner);
	    panner.connect(gainNode2);
	    gainNode2.connect(compressor);

    // Make connections 
    oneShotSound.connect(lowpass);
    lowpass.connect(panner);
    panner.connect(gainNode2);
    gainNode2.connect(compressor);

	    
	    oneShotSound.noteOn(context.currentTime + 0.75);
	}

    // Play 0.75 seconds from now (to play immediately pass in 0)
    oneShotSound.noteOn(context.currentTime + 0.75);
}


8. Channel Layouts

It's important to define the channel ordering (and define some abbreviations) for different layouts.

The channel layouts are clear:

  Mono
    0: M: mono
    
  Stereo
    0: L: left
    1: R: right

A more advanced implementation can handle channel layouts for quad and 5.1:

  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

Other layouts can also be considered.

9. Channel up-mixing and down-mixing

For now, only considers cases for mono, stereo, quad, 5.1. Later other channel layouts can be defined.

Up Mixing

Consider what happens when converting an audio stream with a lower number of channels to one with a higher number of channels. This can be necessary when mixing several outputs together where the channel layouts differ. It can also be necessary if the rendered audio stream is played back on a system with more channels.

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;

Down Mixing

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

  
Mono down-mix:
    2 -> 1 : stereo to mono
        output = 0.5 * (input.L + input.R);
    4 -> 1 : quad to mono
        output = 0.25 * (input.L + input.R + input.SL + input.SR);
    5.1 -> 1 : 5.1 to mono
        ???
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
???

10. Event Scheduling

Need more detail here, but for now:

11. Spatialization / Panning

Background

A common feature requirement for modern 3D games is the ability to dynamically spatialize and move multiple audio sources in 3D space. Game audio engines such as OpenAL, FMOD, Creative's EAX, Microsoft's XACT Audio, etc. have this ability.

Using an AudioPannerNode , an audio stream can be spatialized or positioned in space relative to an AudioListener . An AudioContext will contain a single AudioListener . Both panners and listeners have a position in 3D space using a right-handed cartesian coordinate system. AudioPannerNode objects (representing the source stream) have an orientation vector representing in which direction the sound is projecting. Additionally, they have a sound cone representing how directional the sound is. For example, the sound could be omnidirectional, in which case it would be heard anywhere regardless of its orientation, or it can be more directional and heard only if it is facing the listener. AudioListener objects (representing a person's ears) have an orientation and up vector representing in which direction the person is facing. Because both the source stream and the listener can be moving, they both have a velocity vector representing both the speed and direction of movement. Taken together, these two velocities can be used to generate a doppler shift effect which changes the pitch.

During rendering, the AudioPannerNode calculates an azimuth and elevation . These values are used internally by the implementation in order to render the spatialization effect. See the Panning Algorithm section for details of how these values are used.

The following algorithm must be used to calculate the azimuth and elevation :

 
// 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;

Panning Algorithm

mono-> stereo and stereo-> stereo panning must be supported. mono-> stereo processing is used when all connections to the input are mono. Otherwise stereo-> stereo processing is used.

The following algorithms can must be implemented:

Distance Effects

Sources farther

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

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


v = panner.position - listener.position

distance = sqrt(dot(v, v))

distance will then be used to calculate distanceGain which depends on the distanceModel attribute. See the Constants section for details of how this is calculated for each distance model.

As part of its processing, the AudioPannerNode scales/multiplies the input audio signal by distanceGain to make distant sounds quieter than and nearer ones. Different rolloff curves are assignable per-source: linear, inverse, exponential. ones louder.

Sound Cones

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

Doppler Shift

12. Linear Effects using Convolution

Background

Convolution 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.

Each unique effect is defined by an impulse response . An impulse response can be represented as an audio file and can be recorded from a real acoustic space such as a cave, or can be synthetically generated through a great variety of techniques.

Motivation for use as a Standard

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.

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.

Implementation Guide

Linear convolution can be implemented efficiently. Here are some notes describing how it can be practically implemented.

Reverb Effect (with matrixing)

This section is normative.

Single channel convolution operates on a mono audio source, using a mono impulse response. But to achieve a more spacious sound, multi-channel audio sources and impulse responses must be considered. Audio sources and playback systems can be stereo, 5.1, or more channels. 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.

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 ConvolverNode objects in conjunction with an AudioChannelMerger .

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, K, N and M are all less than 1 or equal to 2. Similarly, the matrixing for 5.1 2 and other playback configurations can be defined. K is 1, 2, or 4.

reverb matrixing

Recording Impulse Responses

This section is informative.

impulse response

The most modern 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.
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.

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.

Tools

Two command-line tools have been written:
generate_testtones generates an exponential sine-sweep test-tone and its inverse. Another tool convolve 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.

% 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

Recording Setup

recording setup

Audio Interface: Metric Halo Mobile I/O 2882



microphones speaker

microphone speaker

Microphones: AKG 414s, Speaker: Mackie HR824


The Warehouse Space

warehouse

13. JavaScript Synthesis and Processing

This section is informative.

The Mozilla project has conducted Experiments 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 AudioNode called JavaScriptAudioNode .

Here are some interesting examples where direct JavaScript processing can be useful:

Custom DSP Effects

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.

Educational Applications

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.

JavaScript Performance

JavaScript has a variety of performance issues 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 modular approach .

14. Realtime Analysis

15. Performance Considerations

15.1. Latency: What it is and Why it's Important

latency

For web applications, the time delay between mouse and keyboard events (keydown, mousedown, etc.) and a sound being heard is important.

This time delay is called latency and is caused by several factors (input device latency, internal buffering latency, DSP processing latency, output device latency, distance of user's ears from speakers, etc.), and is 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.

15.2. Audio Glitching

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

15.3. Hardware Scalability

The system should gracefully degrade to allow audio processing under resource constrained conditions without dropping audio frames.

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 glitches .

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.

Here's a list of some techniques which can be used to limit CPU usage:

15.3.1. CPU monitoring

In order to avoid audio breakup, CPU usage must remain below 100%.

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 cpuUsage attribute of AudioNode for use by JavaScript.

In cases where the measured CPU usage is near 100% (or whatever threshold is considered too high), then an attempt to add additional AudioNodes into the rendering graph can trigger voice-dropping.

15.3.2. Voice Dropping

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 the AudioSourceNode through any effect processing nodes which apply uniquely to that voice.

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.

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:

15.3.3. Simplification of Effects Processing

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 convolution effect 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.

In addition to the convolution effect. The AudioPannerNode 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.

15.3.4. Sample Rate

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 AudioContext 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.

15.3.5. Pre-flighting

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.

TODO: add specification and more detail here

15.3.6. Authoring for different user agents

JavaScript code can use information about user-agent to scale back any more intensive processing it may normally run on a more powerful machine.

15.3.7. Scalability of Direct JavaScript Synthesis / Processing

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.

15.4. JavaScript Issues with real-time Processing and Synthesis:

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. 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.

16. Example Applications

This section is informative.

Please see the demo page for working examples.

Here are some of the types of applications a web audio system should be able to support:

Basic Sound Playback

Simple and low-latency playback of sound effects in response to simple user actions such as mouse click, roll-over, key press.


3D Environments and Games

quake beach demo

An HTML5 version of Quake has already been created. Audio features such as 3D spatialization and convolution for room simulation could be used to great effect.

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.



box2d 8-ball

Box2D 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.

A virtual pool game with multi-sampled sound effects has also been created.


Musical Applications

garageband shiny drum machine tonecraft

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 GarageBand can be written.

Music Visualizers

music visualizer

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.

Educational Applications

javascript processing

A variety of educational applications can be written, illustrating concepts in music theory and computer music synthesis and processing.


Artistic Audio Exploration

There are many creative possibilites for artistic sonic environments for installation pieces.


17. Security Considerations

This section is informative.

18. Privacy Considerations

This section is informative . 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 JavaScriptAudioNode interface. The information could subsequently be used to create a fingerprint of the client.

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, perhaps probably via the getUserMedia() API .

19. Requirements and Use Cases

Please see Example Applications

A.References

A.1 Normative references

[RFC2119]
S. Bradner. Key words for use in RFCs to Indicate Requirement Levels. Internet RFC 2119. URL: http://www.ietf.org/rfc/rfc2119.txt

A.2 Informative references

No informative references.

B.Acknowledgements

Special thanks to the W3C Audio Working Group . Members of the Working Group are (at the time of writing, and by alphabetical order):
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.);

C. Web Audio API Change Log

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: Oscillator 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 AudioChannelSplitter and AudioChannelMerger
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 Oscillator 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
* RealtimeAnalyserNode no longer requires its output be connected to anything
* update AudioChannelMerger section describing numberOfOutputs (defaults to 6 but settable in constructor)
* update AudioChannelSplitter 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 createDelayNode()
* 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 JavaScriptAudioNode section
* Add distance model constants for AudioPannerNode 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

summary:     add convolution