Introduction
Audio on the web has been fairly primitive up to this point and until
very recently has had to be delivered through plugins such as Flash and
QuickTime. The introduction of the audio
element in HTML5
is very important, allowing for basic streaming audio playback. But, it
is not powerful enough to handle more complex audio applications. For
sophisticated web-based games or interactive applications, another
solution is required. It is a goal of this specification to include the
capabilities found in modern game audio engines as well as some of the
mixing, processing, and filtering tasks that are found in modern
desktop audio production applications.
The APIs have been designed with a wide variety of use cases [webaudio-usecases] in mind. Ideally, it should be able to support any use case which could reasonably be implemented with an optimized C++ engine controlled via script and run in a browser. That said, modern desktop audio software can have very advanced capabilities, some of which would be difficult or impossible to build with this system. Apple’s Logic Audio is one such application which has support for external MIDI controllers, arbitrary plugin audio effects and synthesizers, highly optimized direct-to-disk audio file reading/writing, tightly integrated time-stretching, and so on. Nevertheless, the proposed system will be quite capable of supporting a large range of reasonably complex games and interactive applications, including musical ones. And it can be a very good complement to the more advanced graphics features offered by WebGL. The API has been designed so that more advanced capabilities can be added at a later time.
Features
The API supports these primary features:
-
Modular routing for simple or complex mixing/effect architectures.
-
High dynamic range, using 32-bit floats for internal processing.
-
Sample-accurate scheduled sound playback with low latency for musical applications requiring a very high degree of rhythmic precision such as drum machines and sequencers. This also includes the possibility of dynamic creation of effects.
-
Automation of audio parameters for envelopes, fade-ins / fade-outs, granular effects, filter sweeps, LFOs etc.
-
Flexible handling of channels in an audio stream, allowing them to be split and merged.
-
Processing of audio sources from an
audio
orvideo
media element
. -
Processing live audio input using a
MediaStream
fromgetUserMedia()
. -
Integration with WebRTC
-
Processing audio received from a remote peer using a
MediaStreamTrackAudioSourceNode
and [webrtc]. -
Sending a generated or processed audio stream to a remote peer using a
MediaStreamAudioDestinationNode
and [webrtc].
-
-
Audio stream synthesis and processing directly using scripts.
-
Spatialized audio supporting a wide range of 3D games and immersive environments:
-
Panning models: equalpower, HRTF, pass-through
-
Distance Attenuation
-
Sound Cones
-
Obstruction / Occlusion
-
Source / Listener based
-
-
A convolution engine for a wide range of linear effects, especially very high-quality room effects. Here are some examples of possible effects:
-
Small / large room
-
Cathedral
-
Concert hall
-
Cave
-
Tunnel
-
Hallway
-
Forest
-
Amphitheater
-
Sound of a distant room through a doorway
-
Extreme filters
-
Strange backwards effects
-
Extreme comb filter effects
-
-
Dynamics compression for overall control and sweetening of the mix
-
Efficient real-time time-domain and frequency-domain analysis / music visualizer support.
-
Efficient biquad filters for lowpass, highpass, and other common filters.
-
A Waveshaping effect for distortion and other non-linear effects
-
Oscillators
Modular Routing
Modular routing allows arbitrary connections between different AudioNode
objects. Each node can have inputs and/or outputs.
A source node has no inputs and a single output.
A destination node has one input and no outputs. Other nodes such as
filters can be placed between the source and destination nodes. The
developer doesn’t have to worry about low-level stream format
details when two objects are connected together; 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
:
Illustrating this simple routing, here’s a simple example playing a single sound:
const context= new AudioContext(); function playSound() { const source= context. createBufferSource(); source. buffer= dogBarkingBuffer; source. connect( context. destination); source. start( 0 ); }
Here’s a more complex example with three sources and a convolution reverb send with a dynamics compressor at the final output stage:
let context; let compressor; let reverb; let source1, source2, source3; let lowpassFilter; let waveShaper; let panner; let dry1, dry2, dry3; let wet1, wet2, wet3; let mainDry; let mainWet; function setupRoutingGraph() { context= new AudioContext(); // Create the effects nodes. lowpassFilter= context. createBiquadFilter(); waveShaper= context. createWaveShaper(); panner= context. createPanner(); compressor= context. createDynamicsCompressor(); reverb= context. createConvolver(); // Create main wet and dry. mainDry= context. createGain(); mainWet= context. createGain(); // Connect final compressor to final destination. compressor. connect( context. destination); // Connect main dry and wet to compressor. mainDry. connect( compressor); mainWet. connect( compressor); // Connect reverb to main wet. reverb. connect( mainWet); // Create a few sources. source1= context. createBufferSource(); source2= context. createBufferSource(); source3= context. createOscillator(); source1. buffer= manTalkingBuffer; source2. buffer= footstepsBuffer; source3. frequency. value= 440 ; // Connect source1 dry1= context. createGain(); wet1= context. createGain(); source1. connect( lowpassFilter); lowpassFilter. connect( dry1); lowpassFilter. connect( wet1); dry1. connect( mainDry); wet1. connect( reverb); // Connect source2 dry2= context. createGain(); wet2= context. createGain(); source2. connect( waveShaper); waveShaper. connect( dry2); waveShaper. connect( wet2); dry2. connect( mainDry); wet2. connect( reverb); // Connect source3 dry3= context. createGain(); wet3= context. createGain(); source3. connect( panner); panner. connect( dry3); panner. connect( wet3); dry3. connect( mainDry); wet3. connect( reverb); // Start the sources now. source1. start( 0 ); source2. start( 0 ); source3. start( 0 ); }
Modular routing also permits the output of AudioNode
s to be routed to an AudioParam
parameter that controls the behavior
of a different AudioNode
. In this scenario, the
output of a node can act as a modulation signal rather than an
input signal.
function setupRoutingGraph() { const context= new AudioContext(); // Create the low frequency oscillator that supplies the modulation signal const lfo= context. createOscillator(); lfo. frequency. value= 1.0 ; // Create the high frequency oscillator to be modulated const hfo= context. createOscillator(); hfo. frequency. value= 440.0 ; // Create a gain node whose gain determines the amplitude of the modulation signal const modulationGain= context. createGain(); modulationGain. gain. value= 50 ; // Configure the graph and start the oscillators lfo. connect( modulationGain); modulationGain. connect( hfo. detune); hfo. connect( context. destination); hfo. start( 0 ); lfo. start( 0 ); }
API Overview
The interfaces defined are:
-
An AudioContext interface, which contains an audio signal graph representing connections between
AudioNode
s. -
An
AudioNode
interface, which represents audio sources, audio outputs, and intermediate processing modules.AudioNode
s can be dynamically connected together in a modular fashion.AudioNode
s exist in the context of anAudioContext
. -
An
AnalyserNode
interface, anAudioNode
for use with music visualizers, or other visualization applications. -
An
AudioBuffer
interface, for working with memory-resident audio assets. These can represent one-shot sounds, or longer audio clips. -
An
AudioBufferSourceNode
interface, anAudioNode
which generates audio from an AudioBuffer. -
An
AudioDestinationNode
interface, anAudioNode
subclass representing the final destination for all rendered audio. -
An
AudioParam
interface, for controlling an individual aspect of anAudioNode
's functioning, such as volume. -
An
AudioListener
interface, which works with aPannerNode
for spatialization. -
An
AudioWorklet
interface representing a factory for creating custom nodes that can process audio directly using scripts. -
An
AudioWorkletGlobalScope
interface, the context in which AudioWorkletProcessor processing scripts run. -
An
AudioWorkletNode
interface, anAudioNode
representing a node processed in an AudioWorkletProcessor. -
An
AudioWorkletProcessor
interface, representing a single node instance inside an audio worker. -
A
BiquadFilterNode
interface, anAudioNode
for common low-order filters such as:-
Low Pass
-
High Pass
-
Band Pass
-
Low Shelf
-
High Shelf
-
Peaking
-
Notch
-
Allpass
-
-
A
ChannelMergerNode
interface, anAudioNode
for combining channels from multiple audio streams into a single audio stream. -
A
ChannelSplitterNode
interface, anAudioNode
for accessing the individual channels of an audio stream in the routing graph. -
A
ConstantSourceNode
interface, anAudioNode
for generating a nominally constant output value with anAudioParam
to allow automation of the value. -
A
ConvolverNode
interface, anAudioNode
for applying a real-time linear effect (such as the sound of a concert hall). -
A
DelayNode
interface, anAudioNode
which applies a dynamically adjustable variable delay. -
A
DynamicsCompressorNode
interface, anAudioNode
for dynamics compression. -
A
GainNode
interface, anAudioNode
for explicit gain control. -
An
IIRFilterNode
interface, anAudioNode
for a general IIR filter. -
A
MediaElementAudioSourceNode
interface, anAudioNode
which is the audio source from anaudio
,video
, or other media element. -
A
MediaStreamAudioSourceNode
interface, anAudioNode
which is the audio source from aMediaStream
such as live audio input, or from a remote peer. -
A
MediaStreamTrackAudioSourceNode
interface, anAudioNode
which is the audio source from aMediaStreamTrack
. -
A
MediaStreamAudioDestinationNode
interface, anAudioNode
which is the audio destination to aMediaStream
sent to a remote peer. -
A
PannerNode
interface, anAudioNode
for spatializing / positioning audio in 3D space. -
A
PeriodicWave
interface for specifying custom periodic waveforms for use by theOscillatorNode
. -
An
OscillatorNode
interface, anAudioNode
for generating a periodic waveform. -
A
StereoPannerNode
interface, anAudioNode
for equal-power positioning of audio input in a stereo stream. -
A
WaveShaperNode
interface, anAudioNode
which applies a non-linear waveshaping effect for distortion and other more subtle warming effects.
There are also several features that have been deprecated from the Web Audio API but not yet removed, pending implementation experience of their replacements:
-
A
ScriptProcessorNode
interface, anAudioNode
for generating or processing audio directly using scripts. -
An
AudioProcessingEvent
interface, which is an event type used withScriptProcessorNode
objects.
1. The Audio API
1.1. The BaseAudioContext
Interface
This interface represents a set of AudioNode
objects and their connections. It allows for arbitrary routing of
signals to an AudioDestinationNode
. Nodes are
created from the context and are then connected together.
BaseAudioContext
is not instantiated directly,
but is instead extended by the concrete interfaces AudioContext
(for real-time rendering) and OfflineAudioContext
(for offline rendering).
BaseAudioContext
are created with an internal slot [[pending promises]]
that is an
initially empty ordered list of promises.
Each BaseAudioContext
has a unique media element event task source.
Additionally, a BaseAudioContext
has several private slots [[rendering thread state]]
and [[control thread state]]
that take values from AudioContextState
, and that are both initially set to "suspended"
, and a private slot [[render quantum size]]
that is an unsigned integer.
enum {
AudioContextState "suspended" ,"running" ,"closed" };
Enum value | Description |
---|---|
"suspended "
| This context is currently suspended (context time is not proceeding, audio hardware may be powered down/released). |
"running "
| Audio is being processed. |
"closed "
| This context has been released, and can no longer be used to process audio. All system audio resources have been released. |
enum {
AudioContextRenderSizeCategory "default" ,"hardware" };
Enumeration description | |
---|---|
"default "
| The AudioContext’s render quantum size is the default value of 128 frames. |
"hardware "
|
The User-Agent picks a render quantum size that is best for the
current configuration.
Note: This exposes information about the host and can be used for fingerprinting. |
callback DecodeErrorCallback =undefined (DOMException );
error callback DecodeSuccessCallback =undefined (AudioBuffer ); [
decodedData Exposed =Window ]interface BaseAudioContext :EventTarget {readonly attribute AudioDestinationNode destination ;readonly attribute float sampleRate ;readonly attribute double currentTime ;readonly attribute AudioListener listener ;readonly attribute AudioContextState state ;readonly attribute unsigned long renderQuantumSize ; [SameObject ,SecureContext ]readonly attribute AudioWorklet audioWorklet ;attribute EventHandler onstatechange ;AnalyserNode createAnalyser ();BiquadFilterNode createBiquadFilter ();AudioBuffer createBuffer (unsigned long ,
numberOfChannels unsigned long ,
length float );
sampleRate AudioBufferSourceNode createBufferSource ();ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);ChannelSplitterNode createChannelSplitter (optional unsigned long numberOfOutputs = 6);ConstantSourceNode createConstantSource ();ConvolverNode createConvolver ();DelayNode createDelay (optional double maxDelayTime = 1.0);DynamicsCompressorNode createDynamicsCompressor ();GainNode createGain ();IIRFilterNode createIIRFilter (sequence <double >,
feedforward sequence <double >);
feedback OscillatorNode createOscillator ();PannerNode createPanner ();PeriodicWave createPeriodicWave (sequence <float >,
real sequence <float >,
imag optional PeriodicWaveConstraints = {});
constraints ScriptProcessorNode createScriptProcessor (optional unsigned long bufferSize = 0,optional unsigned long numberOfInputChannels = 2,optional unsigned long numberOfOutputChannels = 2);StereoPannerNode createStereoPanner ();WaveShaperNode createWaveShaper ();Promise <AudioBuffer >decodeAudioData (ArrayBuffer ,
audioData optional DecodeSuccessCallback ?,
successCallback optional DecodeErrorCallback ?); };
errorCallback
1.1.1. Attributes
audioWorklet
, of type AudioWorklet, readonly-
Allows access to the
Worklet
object that can import a script containingAudioWorkletProcessor
class definitions via the algorithms defined by [HTML] andAudioWorklet
. currentTime
, of type double, readonly-
This is the time in seconds of the sample frame immediately following the last sample-frame in the block of audio most recently processed by the context’s rendering graph. If the context’s rendering graph has not yet processed a block of audio, then
currentTime
has a value of zero.In the time coordinate system of
currentTime
, the value of zero corresponds to the first sample-frame in the first block processed by the graph. Elapsed time in this system corresponds to elapsed time in the audio stream generated by theBaseAudioContext
, which may not be synchronized with other clocks in the system. (For anOfflineAudioContext
, since the stream is not being actively played by any device, there is not even an approximation to real time.)All scheduled times in the Web Audio API are relative to the value of
currentTime
.When the
BaseAudioContext
is in the "running
" state, the value of this attribute is monotonically increasing and is updated by the rendering thread in uniform increments, corresponding to one render quantum. Thus, for a running context,currentTime
increases steadily as the system processes audio blocks, and always represents the time of the start of the next audio block to be processed. It is also the earliest possible time when any change scheduled in the current state might take effect.currentTime
MUST be read atomically on the control thread before being returned. destination
, of type AudioDestinationNode, readonly-
An
AudioDestinationNode
with a single input representing the final destination for all audio. Usually this will represent the actual audio hardware. AllAudioNode
s actively rendering audio will directly or indirectly connect todestination
. listener
, of type AudioListener, readonly-
An
AudioListener
which is used for 3D spatialization. onstatechange
, of type EventHandler-
A property used to set an event handler for an event that is dispatched to
BaseAudioContext
when the state of the AudioContext has changed (i.e. when the corresponding promise would have resolved). The event type of this event handler isstatechange
. An event that uses theEvent
interface will be dispatched to the event handler, which can query the AudioContext’s state directly. A newly-created AudioContext will always begin in thesuspended
state, and a state change event will be fired whenever the state changes to a different state. This event is fired before thecomplete
event is fired. sampleRate
, of type float, readonly-
The sample rate (in sample-frames per second) at which the
BaseAudioContext
handles audio. It is assumed that allAudioNode
s in the context run at this rate. In making this assumption, sample-rate converters or "varispeed" processors are not supported in real-time processing. The Nyquist frequency is half this sample-rate value. state
, of type AudioContextState, readonly-
Describes the current state of the
BaseAudioContext
. Getting this attribute returns the contents of the[[control thread state]]
slot. renderQuantumSize
, of type unsigned long, readonly-
Getting this attribute returns the value of
[[render quantum size]]
slot.
1.1.2. Methods
createAnalyser()
-
Factory method for an
AnalyserNode
.No parameters.Return type:AnalyserNode
createBiquadFilter()
-
Factory method for a
BiquadFilterNode
representing a second order filter which can be configured as one of several common filter types.No parameters.Return type:BiquadFilterNode
createBuffer(numberOfChannels, length, sampleRate)
-
Creates an AudioBuffer of the given size. The audio data in the buffer will be zero-initialized (silent). A
NotSupportedError
exception MUST be thrown if any of the arguments is negative, zero, or outside its nominal range.Arguments for the BaseAudioContext.createBuffer() method. Parameter Type Nullable Optional Description numberOfChannels
unsigned long
✘ ✘ Determines how many channels the buffer will have. An implementation MUST support at least 32 channels. length
unsigned long
✘ ✘ Determines the size of the buffer in sample-frames. This MUST be at least 1. sampleRate
float
✘ ✘ Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. An implementation MUST support sample rates in at least the range 8000 to 96000. Return type:AudioBuffer
createBufferSource()
-
Factory method for a
AudioBufferSourceNode
.No parameters.Return type:AudioBufferSourceNode
createChannelMerger(numberOfInputs)
-
Factory method for a
ChannelMergerNode
representing a channel merger. AnIndexSizeError
exception MUST be thrown ifnumberOfInputs
is less than 1 or is greater than the number of supported channels.Arguments for the BaseAudioContext.createChannelMerger(numberOfInputs) method. Parameter Type Nullable Optional Description numberOfInputs
unsigned long
✘ ✔ Determines the number of inputs. Values of up to 32 MUST be supported. If not specified, then 6
will be used.Return type:ChannelMergerNode
createChannelSplitter(numberOfOutputs)
-
Factory method for a
ChannelSplitterNode
representing a channel splitter. AnIndexSizeError
exception MUST be thrown ifnumberOfOutputs
is less than 1 or is greater than the number of supported channels.Arguments for the BaseAudioContext.createChannelSplitter(numberOfOutputs) method. Parameter Type Nullable Optional Description numberOfOutputs
unsigned long
✘ ✔ The number of outputs. Values of up to 32 MUST be supported. If not specified, then 6
will be used.Return type:ChannelSplitterNode
createConstantSource()
-
Factory method for a
ConstantSourceNode
.No parameters.Return type:ConstantSourceNode
createConvolver()
-
Factory method for a
ConvolverNode
.No parameters.Return type:ConvolverNode
createDelay(maxDelayTime)
-
Factory method for a
DelayNode
. The initial default delay time will be 0 seconds.Arguments for the BaseAudioContext.createDelay(maxDelayTime) method. Parameter Type Nullable Optional Description maxDelayTime
double
✘ ✔ Specifies the maximum delay time in seconds allowed for the delay line. If specified, this value MUST be greater than zero and less than three minutes or a NotSupportedError
exception MUST be thrown. If not specified, then1
will be used.Return type:DelayNode
createDynamicsCompressor()
-
Factory method for a
DynamicsCompressorNode
.No parameters.Return type:DynamicsCompressorNode
createGain()
-
Factory method for
GainNode
.No parameters.Return type:GainNode
createIIRFilter(feedforward, feedback)
-
Arguments for the BaseAudioContext.createIIRFilter() method. Parameter Type Nullable Optional Description feedforward
sequence<double>
✘ ✘ An array of the feedforward (numerator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20. If all of the values are zero, an InvalidStateError
MUST be thrown. ANotSupportedError
MUST be thrown if the array length is 0 or greater than 20.feedback
sequence<double>
✘ ✘ An array of the feedback (denominator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20. If the first element of the array is 0, an InvalidStateError
MUST be thrown. ANotSupportedError
MUST be thrown if the array length is 0 or greater than 20.Return type:IIRFilterNode
createOscillator()
-
Factory method for an
OscillatorNode
.No parameters.Return type:OscillatorNode
createPanner()
-
Factory method for a
PannerNode
.No parameters.Return type:PannerNode
createPeriodicWave(real, imag, constraints)
-
Factory method to create a
PeriodicWave
.When calling this method, execute these steps:-
If
real
andimag
are not of the same length, anIndexSizeError
MUST be thrown. -
Let o be a new object of type
PeriodicWaveOptions
. -
Respectively set the
real
andimag
parameters passed to this factory method to the attributes of the same name on o. -
Set the
disableNormalization
attribute on o to the value of thedisableNormalization
attribute of theconstraints
attribute passed to the factory method. -
Construct a new
PeriodicWave
p, passing theBaseAudioContext
this factory method has been called on as a first argument, and o. -
Return p.
Arguments for the BaseAudioContext.createPeriodicWave() method. Parameter Type Nullable Optional Description real
sequence<float>
✘ ✘ A sequence of cosine parameters. See its real
constructor argument for a more detailed description.imag
sequence<float>
✘ ✘ A sequence of sine parameters. See its imag
constructor argument for a more detailed description.constraints
PeriodicWaveConstraints
✘ ✔ If not given, the waveform is normalized. Otherwise, the waveform is normalized according the value given by constraints
.Return type:PeriodicWave
-
createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels)
-
Factory method for a
ScriptProcessorNode
. This method is DEPRECATED, as it is intended to be replaced byAudioWorkletNode
. Creates aScriptProcessorNode
for direct audio processing using scripts. AnIndexSizeError
exception MUST be thrown ifbufferSize
ornumberOfInputChannels
ornumberOfOutputChannels
are outside the valid range.It is invalid for both
numberOfInputChannels
andnumberOfOutputChannels
to be zero. In this case anIndexSizeError
MUST be thrown.Arguments for the BaseAudioContext.createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels) method. Parameter Type Nullable Optional Description bufferSize
unsigned long
✘ ✔ The bufferSize
parameter determines the buffer size in units of sample-frames. If it’s not passed in, or if the value is 0, then the implementation will choose the best buffer size for the given environment, which will be constant power of 2 throughout the lifetime of the node. Otherwise if the author explicitly specifies the bufferSize, it MUST be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently theaudioprocess
event is dispatched and how many sample-frames need to be processed each call. Lower values forbufferSize
will result in a lower (better) latency. Higher values will be necessary to avoid audio breakup and glitches. It is recommended for authors to not specify this buffer size and allow the implementation to pick a good buffer size to balance between latency and audio quality. If the value of this parameter is not one of the allowed power-of-2 values listed above, anIndexSizeError
MUST be thrown.numberOfInputChannels
unsigned long
✘ ✔ This parameter determines the number of channels for this node’s input. The default value is 2. Values of up to 32 must be supported. A NotSupportedError
must be thrown if the number of channels is not supported.numberOfOutputChannels
unsigned long
✘ ✔ This parameter determines the number of channels for this node’s output. The default value is 2. Values of up to 32 must be supported. A NotSupportedError
must be thrown if the number of channels is not supported.Return type:ScriptProcessorNode
createStereoPanner()
-
Factory method for a
StereoPannerNode
.No parameters.Return type:StereoPannerNode
createWaveShaper()
-
Factory method for a
WaveShaperNode
representing a non-linear distortion.No parameters.Return type:WaveShaperNode
decodeAudioData(audioData, successCallback, errorCallback)
-
Asynchronously decodes the audio file data contained in the
ArrayBuffer
. TheArrayBuffer
can, for example, be loaded from anXMLHttpRequest
’sresponse
attribute after setting theresponseType
to"arraybuffer"
. Audio file data can be in any of the formats supported by theaudio
element. The buffer passed todecodeAudioData()
has its content-type determined by sniffing, as described in [mimesniff].Although the primary method of interfacing with this function is via its promise return value, the callback parameters are provided for legacy reasons.
Encourage implementation to warn authors in case of a corrupted file. It isn’t possible to throw because this would be a breaking change.
Note: If the compressed audio data byte-stream is corrupted but the decoding can otherwise proceed, implementations are encouraged to warn authors for example via the developer tools.WhendecodeAudioData
is called, the following steps MUST be performed on the control thread:-
If this's relevant global object's associated Document is not fully active then return a promise rejected with "
InvalidStateError
"DOMException
. -
Let promise be a new Promise.
-
If
audioData
is detached, execute the following steps:-
Append promise to
[[pending promises]]
. -
Detach the
audioData
ArrayBuffer
. If this operations throws, jump to the step 3. -
Queue a decoding operation to be performed on another thread.
-
-
Else, execute the following error steps:
-
Let error be a
DataCloneError
. -
Reject promise with error, and remove it from
[[pending promises]]
. -
Queue a media element task to invoke
errorCallback
with error.
-
-
Return promise.
When queuing a decoding operation to be performed on another thread, the following steps MUST happen on a thread that is not the control thread nor the rendering thread, called thedecoding thread
.Note: Multiple
decoding thread
s can run in parallel to service multiple calls todecodeAudioData
.-
Let can decode be a boolean flag, initially set to true.
-
Attempt to determine the MIME type of
audioData
, using MIME Sniffing § 6.2 Matching an audio or video type pattern. If the audio or video type pattern matching algorithm returnsundefined
, set can decode to false. -
If can decode is true, attempt to decode the encoded
audioData
into linear PCM. In case of failure, set can decode to false.If the media byte-stream contains multiple audio tracks, only decode the first track to linear pcm.
Note: Authors who need more control over the decoding process can use [WEBCODECS].
-
If can decode is
false
, queue a media element task to execute the following steps:-
Let error be a
DOMException
whose name isEncodingError
.-
Reject promise with error, and remove it from
[[pending promises]]
.
-
-
If
errorCallback
is not missing, invokeerrorCallback
with error.
-
-
Otherwise:
-
Take the result, representing the decoded linear PCM audio data, and resample it to the sample-rate of the
BaseAudioContext
if it is different from the sample-rate ofaudioData
. -
queue a media element task to execute the following steps:
-
Let buffer be an
AudioBuffer
containing the final result (after possibly performing sample-rate conversion). -
Resolve promise with buffer.
-
If
successCallback
is not missing, invokesuccessCallback
with buffer.
-
-
Arguments for the BaseAudioContext.decodeAudioData() method. Parameter Type Nullable Optional Description audioData
ArrayBuffer
✘ ✘ An ArrayBuffer containing compressed audio data. successCallback
DecodeSuccessCallback?
✔ ✔ 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
DecodeErrorCallback?
✔ ✔ A callback function which will be invoked if there is an error decoding the audio file. Return type:Promise
<AudioBuffer
> -
1.1.3. Callback DecodeSuccessCallback()
Parameters
decodedData
, of typeAudioBuffer
-
The AudioBuffer containing the decoded audio data.
1.1.4. Callback DecodeErrorCallback()
Parameters
error
, of typeDOMException
-
The error that occurred while decoding.
1.1.5. Lifetime
Once created, an AudioContext
will continue to play
sound until it has no more sound to play, or the page goes away.
1.1.6. Lack of Introspection or Serialization Primitives
The Web Audio API takes a fire-and-forget approach to
audio source scheduling. That is, source nodes are created
for each note during the lifetime of the AudioContext
, and
never explicitly removed from the graph. This is incompatible with
a serialization API, since there is no stable set of nodes that
could be serialized.
Moreover, having an introspection API would allow content script to be able to observe garbage collections.
1.1.7. System Resources Associated with BaseAudioContext
Subclasses
The subclasses AudioContext
and OfflineAudioContext
should be considered expensive objects. Creating these objects may
involve creating a high-priority thread, or using a low-latency
system audio stream, both having an impact on energy consumption.
It is usually not necessary to create more than one AudioContext
in a document.
Constructing or resuming a BaseAudioContext
subclass
involves acquiring system resources for
that context. For AudioContext
, this also requires creation
of a system audio stream. These operations return when the context
begins generating output from its associated audio graph.
Additionally, a user-agent can have an implementation-defined
maximum number of AudioContext
s, after which any attempt to
create a new AudioContext
will fail, throwing NotSupportedError
.
suspend
and close
allow authors to release system resources, including threads,
processes and audio streams. Suspending a BaseAudioContext
permits implementations to release some of its resources, and
allows it to continue to operate later by invoking resume
. Closing an AudioContext
permits implementations to release all of its
resources, after which it cannot be used or resumed again.
Note: For example, this can involve waiting for the audio callbacks to fire regularly, or to wait for the hardware to be ready for processing.
1.2. The AudioContext
Interface
This interface represents an audio graph whose AudioDestinationNode
is routed to a real-time
output device that produces a signal directed at the user. In most
use cases, only a single AudioContext
is used per
document.
enum {
AudioContextLatencyCategory "balanced" ,"interactive" ,"playback" };
Enum value | Description |
---|---|
"balanced "
| Balance audio output latency and power consumption. |
"interactive "
| Provide the lowest audio output latency possible without glitching. This is the default. |
"playback "
| Prioritize sustained playback without interruption over audio output latency. Lowest power consumption. |
enum {
AudioSinkType "none" };
Enum Value | Description |
---|---|
"none "
| The audio graph will be processed without being played through an audio output device. |
[Exposed =Window ]interface AudioContext :BaseAudioContext {constructor (optional AudioContextOptions contextOptions = {});readonly attribute double baseLatency ;readonly attribute double outputLatency ; [SecureContext ]readonly attribute (DOMString or AudioSinkInfo )sinkId ; [SecureContext ]readonly attribute AudioRenderCapacity renderCapacity ;attribute EventHandler onsinkchange ;attribute EventHandler onerror ;AudioTimestamp getOutputTimestamp ();Promise <undefined >resume ();Promise <undefined >suspend ();Promise <undefined >close (); [SecureContext ]Promise <undefined >((
setSinkId DOMString or AudioSinkOptions ));
sinkId MediaElementAudioSourceNode createMediaElementSource (HTMLMediaElement );
mediaElement MediaStreamAudioSourceNode createMediaStreamSource (MediaStream );
mediaStream MediaStreamTrackAudioSourceNode createMediaStreamTrackSource (MediaStreamTrack );
mediaStreamTrack MediaStreamAudioDestinationNode createMediaStreamDestination (); };
An AudioContext
is said to be allowed to start if the user agent
allows the context state to transition from "suspended
" to
"running
". A user agent may disallow this initial transition,
and to allow it only when the AudioContext
's relevant global object has sticky activation.
AudioContext
has following internal slots:
[[suspended by user]]
-
A boolean flag representing whether the context is suspended by user code. The initial value is
false
. [[sink ID]]
-
A
DOMString
or anAudioSinkInfo
representing the identifier or the information of the current audio output device respectively. The initial value is""
, which means the default audio output device. [[pending resume promises]]
-
An ordered list to store pending
Promise
s created byresume()
. It is initially empty.
1.2.1. Constructors
AudioContext(contextOptions)
-
If the current settings object's relevant global object's associated Document is NOT fully active, throw an "
When creating anInvalidStateError
" and abort these steps.AudioContext
, execute these steps:-
Let context be a new
AudioContext
object. -
Set a
[[control thread state]]
tosuspended
on context. -
Set a
[[rendering thread state]]
tosuspended
on context. -
Let messageChannel be a new
MessageChannel
. -
Let controlSidePort be the value of messageChannel’s
port1
attribute. -
Let renderingSidePort be the value of messageChannel’s
port2
attribute. -
Let serializedRenderingSidePort be the result of StructuredSerializeWithTransfer(renderingSidePort, « renderingSidePort »).
-
Set this
audioWorklet
'sport
to controlSidePort. -
Queue a control message to set the MessagePort on the AudioContextGlobalScope, with serializedRenderingSidePort.
-
If
contextOptions
is given, perform the following substeps:-
If
sinkId
is specified, let sinkId be the value ofcontextOptions.
and run the following substeps:sinkId
-
If both sinkId and
[[sink ID]]
are a type ofDOMString
, and they are equal to each other, abort these substeps. -
If sinkId is a type of
AudioSinkOptions
and[[sink ID]]
is a type ofAudioSinkInfo
, andtype
in sinkId andtype
in[[sink ID]]
are equal, abort these substeps. -
Let validationResult be the return value of sink identifier validation of sinkId.
-
If validationResult is a type of
DOMException
, throw an exception with validationResult and abort these substeps. -
If sinkId is a type of
DOMString
, set[[sink ID]]
to sinkId and abort these substeps. -
If sinkId is a type of
AudioSinkOptions
, set[[sink ID]]
to a new instance ofAudioSinkInfo
created with the value oftype
of sinkId.
-
-
Set the internal latency of context according to
contextOptions.
, as described inlatencyHint
latencyHint
. -
If
contextOptions.
is specified, set thesampleRate
sampleRate
of context to this value. Otherwise, follow these substeps:-
If sinkId is the empty string or a type of
AudioSinkOptions
, use the sample rate of the default output device. Abort these substeps. -
If sinkId is a
DOMString
, use the sample rate of the output device identified by sinkId. Abort these substeps.
If
contextOptions.
differs from the sample rate of the output device, the user agent MUST resample the audio output to match the sample rate of the output device.sampleRate
Note: If resampling is required, the latency of context may be affected, possibly by a large amount.
-
-
-
If context is allowed to start, send a control message to start processing.
-
Return context.
Sending a control message to start processing means executing the following steps:-
Let document be the current settings object's relevant global object's associated Document.
-
Attempt to acquire system resources to use a following audio output device based on
[[sink ID]]
for rendering:-
The default audio output device for the empty string.
-
A audio output device identified by
[[sink ID]]
.
-
If resource acquisition fails, execute the following steps:
-
If document is not allowed to use the feature identified by
"speaker-selection"
, abort these substeps. -
Queue a media element task to fire an event named
error
at theAudioContext
, and abort the following steps.
-
-
-
Set this
[[rendering thread state]]
torunning
on theAudioContext
. -
Queue a media element task to execute the following steps:
-
Set the
state
attribute of theAudioContext
to "running
". -
fire an event named
statechange
at theAudioContext
.
-
NOTE: In cases where an
AudioContext
is constructed with no arguments and resource acquisition fails, the User-Agent will attempt to silently render the audio graph using a mechanism that emulates an audio output device.Sending a control message to set theMessagePort
on theAudioWorkletGlobalScope
means executing the following steps, on the rendering thread, with serializedRenderingSidePort, that has been transfered to theAudioWorkletGlobalScope
:-
Let deserializedPort be the result of StructuredDeserialize(serializedRenderingSidePort, the current Realm).
-
Set
port
to deserializedPort.
Arguments for the AudioContext.constructor(contextOptions) method. Parameter Type Nullable Optional Description contextOptions
AudioContextOptions
✘ ✔ User-specified options controlling how the AudioContext
should be constructed. -
1.2.2. Attributes
baseLatency
, of type double, readonly-
This represents the number of seconds of processing latency incurred by the
AudioContext
passing the audio from theAudioDestinationNode
to the audio subsystem. It does not include any additional latency that might be caused by any other processing between the output of theAudioDestinationNode
and the audio hardware and specifically does not include any latency incurred the audio graph itself.For example, if the audio context is running at 44.1 kHz with default render quantum size, and the
AudioDestinationNode
implements double buffering internally and can process and output audio each render quantum, then the processing latency is \((2\cdot128)/44100 = 5.805 \mathrm{ ms}\), approximately. outputLatency
, of type double, readonly-
The estimation in seconds of audio output latency, i.e., the interval between the time the UA requests the host system to play a buffer and the time at which the first sample in the buffer is actually processed by the audio output device. For devices such as speakers or headphones that produce an acoustic signal, this latter time refers to the time when a sample’s sound is produced.
An
outputLatency
attribute value depends on the platform and the connected audio output device hardware. TheoutputLatency
attribute value may change while the context is running or the associated audio output device changes. It is useful to query this value frequently when accurate synchronization is required. renderCapacity
, of type AudioRenderCapacity, readonly-
Returns an
AudioRenderCapacity
instance associated with anAudioContext
. sinkId
, of type(DOMString or AudioSinkInfo)
, readonly-
Returns the value of
[[sink ID]]
internal slot. This attribute is cached upon update, and it returns the same object after caching. onsinkchange
, of type EventHandler-
An event handler for
setSinkId()
. The event type of this event handler issinkchange
. This event will be dispatched when changing the output device is completed.NOTE: This is not dispatched for the initial device selection in the construction of
AudioContext
. Thestatechange
event is available to check the readiness of the initial output device. onerror
, of type EventHandler-
An event handler for the
Event
dispatched from anAudioContext
. The event type of this handler iserror
and the user agent can dispatch this event in the following cases:-
When initializing and activating a selected audio device encounters failures.
-
When the audio output device associated with an
AudioContext
is disconnected while the context isrunning
. -
When the operating system reports an audio device malfunction.
-
1.2.3. Methods
close()
-
Closes the
AudioContext
, releasing the system resources being used. This will not automatically release allAudioContext
-created objects, but will suspend the progression of theAudioContext
'scurrentTime
, and stop processing audio data.When close is called, execute these steps:-
If this's relevant global object's associated Document is not fully active then return a promise rejected with "
InvalidStateError
"DOMException
. -
Let promise be a new Promise.
-
If the
[[control thread state]]
flag on theAudioContext
isclosed
reject the promise withInvalidStateError
, abort these steps, returning promise. -
Set the
[[control thread state]]
flag on theAudioContext
toclosed
. -
Queue a control message to close the
AudioContext
. -
Return promise.
Running a control message to close anAudioContext
means running these steps on the rendering thread:-
Attempt to release system resources.
-
Set the
[[rendering thread state]]
tosuspended
.This will stop rendering. -
If this control message is being run in a reaction to the document being unloaded, abort this algorithm.
There is no need to notify the control thread in this case. -
queue a media element task to execute the following steps:
-
Resolve promise.
-
If the
state
attribute of theAudioContext
is not already "closed
":-
Set the
state
attribute of theAudioContext
to "closed
". -
queue a media element task to fire an event named
statechange
at theAudioContext
.
-
-
When an
AudioContext
is closed, anyMediaStream
s andHTMLMediaElement
s that were connected to anAudioContext
will have their output ignored. That is, these will no longer cause any output to speakers or other output devices. For more flexibility in behavior, consider usingHTMLMediaElement.captureStream()
.Note: When an
AudioContext
has been closed, implementation can choose to aggressively release more resources than when suspending.No parameters. -
createMediaElementSource(mediaElement)
-
Creates a
MediaElementAudioSourceNode
given anHTMLMediaElement
. As a consequence of calling this method, audio playback from theHTMLMediaElement
will be re-routed into the processing graph of theAudioContext
.Arguments for the AudioContext.createMediaElementSource() method. Parameter Type Nullable Optional Description mediaElement
HTMLMediaElement
✘ ✘ The media element that will be re-routed. Return type:MediaElementAudioSourceNode
createMediaStreamDestination()
-
Creates a
MediaStreamAudioDestinationNode
No parameters.Return type:MediaStreamAudioDestinationNode
createMediaStreamSource(mediaStream)
-
Creates a
MediaStreamAudioSourceNode
.Arguments for the AudioContext.createMediaStreamSource() method. Parameter Type Nullable Optional Description mediaStream
MediaStream
✘ ✘ The media stream that will act as source. Return type:MediaStreamAudioSourceNode
createMediaStreamTrackSource(mediaStreamTrack)
-
Creates a
MediaStreamTrackAudioSourceNode
.Arguments for the AudioContext.createMediaStreamTrackSource() method. Parameter Type Nullable Optional Description mediaStreamTrack
MediaStreamTrack
✘ ✘ The MediaStreamTrack
that will act as source. The value of itskind
attribute must be equal to"audio"
, or anInvalidStateError
exception MUST be thrown.Return type:MediaStreamTrackAudioSourceNode
getOutputTimestamp()
-
Returns a new
AudioTimestamp
instance containing two related audio stream position values for the context: thecontextTime
member contains the time of the sample frame which is currently being rendered by the audio output device (i.e., output audio stream position), in the same units and origin as context’scurrentTime
; theperformanceTime
member contains the time estimating the moment when the sample frame corresponding to the storedcontextTime
value was rendered by the audio output device, in the same units and origin asperformance.now()
(described in [hr-time-3]).If the context’s rendering graph has not yet processed a block of audio, then
getOutputTimestamp
call returns anAudioTimestamp
instance with both members containing zero.After the context’s rendering graph has started processing of blocks of audio, its
currentTime
attribute value always exceeds thecontextTime
value obtained fromgetOutputTimestamp
method call.The value returned fromgetOutputTimestamp
method can be used to get performance time estimation for the slightly later context’s time value:function outputPerformanceTime( contextTime) { const timestamp= context. getOutputTimestamp(); const elapsedTime= contextTime- timestamp. contextTime; return timestamp. performanceTime+ elapsedTime* 1000 ; } In the above example the accuracy of the estimation depends on how close the argument value is to the current output audio stream position: the closer the given
contextTime
is totimestamp.contextTime
, the better the accuracy of the obtained estimation.Note: The difference between the values of the context’s
currentTime
and thecontextTime
obtained fromgetOutputTimestamp
method call cannot be considered as a reliable output latency estimation becausecurrentTime
may be incremented at non-uniform time intervals, sooutputLatency
attribute should be used instead.No parameters.Return type:AudioTimestamp
resume()
-
Resumes the progression of the
AudioContext
'scurrentTime
when it has been suspended.When resume is called, execute these steps:-
If this's relevant global object's associated Document is not fully active then return a promise rejected with "
InvalidStateError
"DOMException
. -
Let promise be a new Promise.
-
If the
[[control thread state]]
on theAudioContext
isclosed
reject the promise withInvalidStateError
, abort these steps, returning promise. -
Set
[[suspended by user]]
tofalse
. -
If the context is not allowed to start, append promise to
[[pending promises]]
and[[pending resume promises]]
and abort these steps, returning promise. -
Set the
[[control thread state]]
on theAudioContext
torunning
. -
Queue a control message to resume the
AudioContext
. -
Return promise.
Running a control message to resume anAudioContext
means running these steps on the rendering thread:-
Attempt to acquire system resources.
-
Set the
[[rendering thread state]]
on theAudioContext
torunning
. -
Start rendering the audio graph.
-
In case of failure, queue a media element task to execute the following steps:
-
Reject all promises from
[[pending resume promises]]
in order, then clear[[pending resume promises]]
. -
Additionally, remove those promises from
[[pending promises]]
.
-
-
queue a media element task to execute the following steps:
-
Resolve all promises from
[[pending resume promises]]
in order. -
Clear
[[pending resume promises]]
. Additionally, remove those promises from[[pending promises]]
. -
Resolve promise.
-
If the
state
attribute of theAudioContext
is not already "running
":-
Set the
state
attribute of theAudioContext
to "running
". -
Queue a media element task to fire an event named
statechange
at theAudioContext
.
-
-
No parameters. -
suspend()
-
Suspends the progression of
AudioContext
'scurrentTime
, allows any current context processing blocks that are already processed to be played to the destination, and then allows the system to release its claim on audio hardware. This is generally useful when the application knows it will not need theAudioContext
for some time, and wishes to temporarily release system resource associated with theAudioContext
. The promise resolves when the frame buffer is empty (has been handed off to the hardware), or immediately (with no other effect) if the context is alreadysuspended
. The promise is rejected if the context has been closed.When suspend is called, execute these steps:-
If this's relevant global object's associated Document is not fully active then return a promise rejected with "
InvalidStateError
"DOMException
. -
Let promise be a new Promise.
-
If the
[[control thread state]]
on theAudioContext
isclosed
reject the promise withInvalidStateError
, abort these steps, returning promise. -
Append promise to
[[pending promises]]
. -
Set
[[suspended by user]]
totrue
. -
Set the
[[control thread state]]
on theAudioContext
tosuspended
. -
Queue a control message to suspend the
AudioContext
. -
Return promise.
Running a control message to suspend anAudioContext
means running these steps on the rendering thread:-
Attempt to release system resources.
-
Set the
[[rendering thread state]]
on theAudioContext
tosuspended
. -
queue a media element task to execute the following steps:
-
Resolve promise.
-
If the
state
attribute of theAudioContext
is not already "suspended
":-
Set the
state
attribute of theAudioContext
to "suspended
". -
Queue a media element task to fire an event named
statechange
at theAudioContext
.
-
-
While an
AudioContext
is suspended,MediaStream
s will have their output ignored; that is, data will be lost by the real time nature of media streams.HTMLMediaElement
s will similarly have their output ignored until the system is resumed.AudioWorkletNode
s andScriptProcessorNode
s will cease to have their processing handlers invoked while suspended, but will resume when the context is resumed. For the purpose ofAnalyserNode
window functions, the data is considered as a continuous stream - i.e. theresume()
/suspend()
does not cause silence to appear in theAnalyserNode
's stream of data. In particular, callingAnalyserNode
functions repeatedly when aAudioContext
is suspended MUST return the same data.No parameters. -
setSinkId((DOMString or AudioSinkOptions) sinkId)
-
Sets the identifier of an output device. When this method is invoked, the user agent MUST run the following steps:
-
Let sinkId be the method’s first argument.
-
If sinkId is equal to
[[sink ID]]
, return a promise, resolve it immediately and abort these steps. -
Let validationResult be the return value of sink identifier validation of sinkId.
-
If validationResult is not
null
, return a promise rejected with validationResult. Abort these steps. -
Let p be a new promise.
-
Send a control message with p and sinkId to start processing.
-
Return p.
Sending a control message to start processing duringsetSinkId()
means executing the following steps:-
Let p be the promise passed into this algorithm.
-
Let sinkId be the sink identifier passed into this algorithm.
-
If both sinkId and
[[sink ID]]
are a type ofDOMString
, and they are equal to each other, queue a media element task to resolve p and abort these steps. -
If sinkId is a type of
AudioSinkOptions
and[[sink ID]]
is a type ofAudioSinkInfo
, andtype
in sinkId andtype
in[[sink ID]]
are equal, queue a media element task to resolve p and abort these steps. -
Let wasRunning be true.
-
Set wasRunning to false if the
[[rendering thread state]]
on theAudioContext
is"suspended"
. -
Pause the renderer after processing the current render quantum.
-
Attempt to release system resources.
-
If wasRunning is true:
-
Set the
[[rendering thread state]]
on theAudioContext
to"suspended"
. -
Queue a media element task to execute the following steps:
-
If the
state
attribute of theAudioContext
is not already "suspended
":-
Set the
state
attribute of theAudioContext
to "suspended
". -
Fire an event named
statechange
at the associatedAudioContext
.
-
-
-
-
Attempt to acquire system resources to use a following audio output device based on
[[sink ID]]
for rendering:-
The default audio output device for the empty string.
-
A audio output device identified by
[[sink ID]]
.
In case of failure, reject p with "
InvalidAccessError
" abort the following steps. -
-
Queue a media element task to execute the following steps:
-
If sinkId is a type of
DOMString
, set[[sink ID]]
to sinkId. Abort these steps. -
If sinkId is a type of
AudioSinkOptions
and[[sink ID]]
is a type ofDOMString
, set[[sink ID]]
to a new instance ofAudioSinkInfo
created with the value oftype
of sinkId. -
If sinkId is a type of
AudioSinkOptions
and[[sink ID]]
is a type ofAudioSinkInfo
, settype
of[[sink ID]]
to thetype
value of sinkId. -
Resolve p.
-
Fire an event named
sinkchange
at the associatedAudioContext
.
-
-
If wasRunning is true:
-
Set the
[[rendering thread state]]
on theAudioContext
to"running"
. -
Queue a media element task to execute the following steps:
-
If the
state
attribute of theAudioContext
is not already "running
":-
Set the
state
attribute of theAudioContext
to "running
". -
Fire an event named
statechange
at the associatedAudioContext
.
-
-
-
-
1.2.4. Validating sinkId
This algorithm is used to validate the information provided to modify sinkId
:
-
Let document be the current settings object’s associated Document.
-
Let sinkIdArg be the value passed in to this algorithm.
-
If document is not allowed to use the feature identified by
"speaker-selection"
, return a newDOMException
whose name is "NotAllowedError
". -
If sinkIdArg is a type of
DOMString
but it is not equal to the empty string or it does not match any audio output device identified by the result that would be provided byenumerateDevices()
, return a newDOMException
whose name is "NotFoundError
". -
Return
null
.
1.2.5. AudioContextOptions
The AudioContextOptions
dictionary is used to
specify user-specified options for an AudioContext
.
dictionary AudioContextOptions { (AudioContextLatencyCategory or double )latencyHint = "interactive";float sampleRate ; (DOMString or AudioSinkOptions )sinkId ; (AudioContextRenderSizeCategory or unsigned long )renderSizeHint = "default"; };
1.2.5.1. Dictionary AudioContextOptions
Members
latencyHint
, of type(AudioContextLatencyCategory or double)
, defaulting to"interactive"
-
Identify the type of playback, which affects tradeoffs between audio output latency and power consumption.
The preferred value of the
latencyHint
is a value fromAudioContextLatencyCategory
. However, a double can also be specified for the number of seconds of latency for finer control to balance latency and power consumption. It is at the browser’s discretion to interpret the number appropriately. The actual latency used is given by AudioContext’sbaseLatency
attribute. sampleRate
, of type float-
Set the
sampleRate
to this value for theAudioContext
that will be created. The supported values are the same as the sample rates for anAudioBuffer
. ANotSupportedError
exception MUST be thrown if the specified sample rate is not supported.If
sampleRate
is not specified, the preferred sample rate of the output device for thisAudioContext
is used. sinkId
, of type(DOMString or AudioSinkOptions)
-
The identifier or associated information of the audio output device. See
sinkId
for more details. renderSizeHint
, of type(AudioContextRenderSizeCategory or unsigned long)
, defaulting to"default"
-
This allows users to ask for a particular render quantum size when an integer is passed, to use the default of 128 frames if nothing or
"default"
is passed, or to ask the User-Agent to pick a good render quantum size if"hardware"
is specified.It is a hint that might not be honored.
1.2.6. AudioSinkOptions
The AudioSinkOptions
dictionary is used to specify options for sinkId
.
dictionary AudioSinkOptions {required AudioSinkType type ; };
1.2.6.1. Dictionary AudioSinkOptions
Members
type
, of type AudioSinkType-
A value of
AudioSinkType
to specify the type of the device.
1.2.7. AudioSinkInfo
The AudioSinkInfo
interface is used to get information on the current
audio output device via sinkId
.
[Exposed =Window ]interface AudioSinkInfo {readonly attribute AudioSinkType type ; };
1.2.7.1. Attributes
type
, of type AudioSinkType, readonly-
A value of
AudioSinkType
that represents the type of the device.
1.2.8. AudioTimestamp
dictionary AudioTimestamp {double contextTime ;DOMHighResTimeStamp performanceTime ; };
1.2.8.1. Dictionary AudioTimestamp
Members
contextTime
, of type double-
Represents a point in the time coordinate system of BaseAudioContext’s
currentTime
. performanceTime
, of type DOMHighResTimeStamp-
Represents a point in the time coordinate system of a
Performance
interface implementation (described in [hr-time-3]).
1.2.9. AudioRenderCapacity
[Exposed =Window ]interface :
AudioRenderCapacity EventTarget {undefined start (optional AudioRenderCapacityOptions = {});
options undefined stop ();attribute EventHandler onupdate ; };
This interface provides rendering performance metrics of an AudioContext
. In order to calculate them, the renderer collects a load value per system-level audio callback.
1.2.9.1. Attributes
onupdate
, of type EventHandler-
The event type of this event handler is
update
. Events dispatched to the event handler will use theAudioRenderCapacityEvent
interface.
1.2.9.2. Methods
start(options)
-
Starts metric collection and analysis. This will repeatedly fire an event named
update
at theAudioRenderCapacity
, usingAudioRenderCapacityEvent
, with the given update interval inAudioRenderCapacityOptions
. stop()
-
Stops metric collection and analysis. It also stops dispatching
update
events.
1.2.10. AudioRenderCapacityOptions
The AudioRenderCapacityOptions
dictionary can be used to provide user
options for an AudioRenderCapacity
.
dictionary {
AudioRenderCapacityOptions double updateInterval = 1; };
1.2.10.1. Dictionary AudioRenderCapacityOptions
Members
updateInterval
, of type double, defaulting to1
-
An update interval (in second) for dispaching
AudioRenderCapacityEvent
s. A load value is calculated per system-level audio callback, and multiple load values will be collected over the specified interval period. For example, if the renderer runs at a 48Khz sample rate and the system-level audio callback’s buffer size is 192 frames, 250 load values will be collected over 1 second interval.If the given value is smaller than the duration of the system-level audio callback,
NotSupportedError
is thrown.
1.2.11. AudioRenderCapacityEvent
[Exposed =Window ]interface :
AudioRenderCapacityEvent Event {(
constructor DOMString ,
type optional AudioRenderCapacityEventInit = {});
eventInitDict readonly attribute double timestamp ;readonly attribute double averageLoad ;readonly attribute double peakLoad ;readonly attribute double underrunRatio ; };dictionary :
AudioRenderCapacityEventInit EventInit {double = 0;
timestamp double = 0;
averageLoad double = 0;
peakLoad double = 0; };
underrunRatio
1.2.11.1. Attributes
timestamp
, of type double, readonly-
The start time of the data collection period in terms of the associated
AudioContext
'scurrentTime
. averageLoad
, of type double, readonly-
An average of collected load values over the given update interval. The precision is limited to 1/100th.
peakLoad
, of type double, readonly-
A maximum value from collected load values over the given update interval. The precision is also limited to 1/100th.
underrunRatio
, of type double, readonly-
A ratio between the number of buffer underruns (when a load value is greater than 1.0) and the total number of system-level audio callbacks over the given update interval.
Where \(u\) is the number of buffer underruns and \(N\) is the number of system-level audio callbacks over the given update interval, the buffer underrun ratio is:
-
0.0 if \(u\) = 0.
-
Otherwise, compute \(u/N\) and take a ceiling value of the nearest 100th.
-
1.3. The OfflineAudioContext
Interface
OfflineAudioContext
is a particular type of BaseAudioContext
for rendering/mixing-down
(potentially) faster than real-time. It does not render to the audio
hardware, but instead renders as quickly as possible, fulfilling the
returned promise with the rendered result as an AudioBuffer
.
[Exposed =Window ]interface OfflineAudioContext :BaseAudioContext {constructor (OfflineAudioContextOptions contextOptions );constructor (unsigned long numberOfChannels ,unsigned long length ,float sampleRate );Promise <AudioBuffer >startRendering ();Promise <undefined >resume ();Promise <undefined >suspend (double );
suspendTime readonly attribute unsigned long length ;attribute EventHandler oncomplete ; };
1.3.1. Constructors
OfflineAudioContext(contextOptions)
-
If the current settings object's relevant global object's associated Document is NOT fully active, throw an
Let c be a newInvalidStateError
and abort these steps.OfflineAudioContext
object. Initialize c as follows:-
Set the
[[control thread state]]
for c to"suspended"
. -
Set the
[[rendering thread state]]
for c to"suspended"
. -
Determine the
[[render quantum size]]
for thisOfflineAudioContext
, based on the value of therenderSizeHint
:-
If it has the default value of
"default"
or"hardware
", set the[[render quantum size]]
private slot to 128. -
Else, if an integer has been passed, the User-Agent can decide to honour this value by setting it to the
[[render quantum size]]
private slot.
-
-
Construct an
AudioDestinationNode
with itschannelCount
set tocontextOptions.numberOfChannels
. -
Let messageChannel be a new
MessageChannel
. -
Let controlSidePort be the value of messageChannel’s
port1
attribute. -
Let renderingSidePort be the value of messageChannel’s
port2
attribute. -
Let serializedRenderingSidePort be the result of StructuredSerializeWithTransfer(renderingSidePort, « renderingSidePort »).
-
Set this
audioWorklet
'sport
to controlSidePort. -
Queue a control message to set the MessagePort on the AudioContextGlobalScope, with serializedRenderingSidePort.
Arguments for the OfflineAudioContext.constructor(contextOptions) method. Parameter Type Nullable Optional Description contextOptions
The initial parameters needed to construct this context. -
OfflineAudioContext(numberOfChannels, length, sampleRate)
-
The
OfflineAudioContext
can be constructed with the same arguments as AudioContext.createBuffer. ANotSupportedError
exception MUST be thrown if any of the arguments is negative, zero, or outside its nominal range.The OfflineAudioContext is constructed as if
new OfflineAudioContext({ numberOfChannels: numberOfChannels, length: length, sampleRate: sampleRate}) were called instead.
Arguments for the OfflineAudioContext.constructor(numberOfChannels, length, sampleRate) method. Parameter Type Nullable Optional Description numberOfChannels
unsigned long
✘ ✘ Determines how many channels the buffer will have. See createBuffer()
for the supported number of channels.length
unsigned long
✘ ✘ Determines the size of the buffer in sample-frames. sampleRate
float
✘ ✘ Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. See createBuffer()
for valid sample rates.
1.3.2. Attributes
length
, of type unsigned long, readonly-
The size of the buffer in sample-frames. This is the same as the value of the
length
parameter for the constructor. oncomplete
, of type EventHandler-
The event type of this event handler is
complete
. The event dispatched to the event handler will use theOfflineAudioCompletionEvent
interface. It is the last event fired on anOfflineAudioContext
.
1.3.3. Methods
startRendering()
-
Given the current connections and scheduled changes, starts rendering audio.
Although the primary method of getting the rendered audio data is via its promise return value, the instance will also fire an event named
complete
for legacy reasons.Let[[rendering started]]
be an internal slot of thisOfflineAudioContext
. Initialize this slot to false.When
startRendering
is called, the following steps MUST be performed on the control thread:- If this's relevant global object's associated Document is not fully active then return a promise rejected with "
InvalidStateError
"DOMException
. - If the
[[rendering started]]
slot on theOfflineAudioContext
is true, return a rejected promise withInvalidStateError
, and abort these steps. - Set the
[[rendering started]]
slot of theOfflineAudioContext
to true. - Let promise be a new promise.
- Create a new
AudioBuffer
, with a number of channels, length and sample rate equal respectively to thenumberOfChannels
,length
andsampleRate
values passed to this instance’s constructor in thecontextOptions
parameter. Assign this buffer to an internal slot[[rendered buffer]]
in theOfflineAudioContext
. - If an exception was thrown during the preceding
AudioBuffer
constructor call, reject promise with this exception. - Otherwise, in the case that the buffer was successfully constructed, begin offline rendering.
- Append promise to
[[pending promises]]
. - Return promise.
To begin offline rendering, the following steps MUST happen on a rendering thread that is created for the occasion.- Given the current connections and scheduled changes, start
rendering
length
sample-frames of audio into[[rendered buffer]]
- For every render quantum, check and
suspend
rendering if necessary. - If a suspended context is resumed, continue to render the buffer.
-
Once the rendering is complete, queue a media element task to execute the following steps:
- Resolve the promise created by
startRendering()
with[[rendered buffer]]
. - Queue a media element task to fire an event named
complete
at theOfflineAudioContext
usingOfflineAudioCompletionEvent
whoserenderedBuffer
property is set to[[rendered buffer]]
.
- Resolve the promise created by
No parameters.Return type:Promise
<AudioBuffer
> - If this's relevant global object's associated Document is not fully active then return a promise rejected with "
resume()
-
Resumes the progression of the
OfflineAudioContext
'scurrentTime
when it has been suspended.When resume is called, execute these steps:-
If this's relevant global object's associated Document is not fully active then return a promise rejected with "
InvalidStateError
"DOMException
. -
Let promise be a new Promise.
-
Abort these steps and reject promise with
InvalidStateError
when any of following conditions is true:-
The
[[control thread state]]
on theOfflineAudioContext
isclosed
. -
The
[[rendering started]]
slot on theOfflineAudioContext
is false.
-
-
Set the
[[control thread state]]
flag on theOfflineAudioContext
torunning
. -
Queue a control message to resume the
OfflineAudioContext
. -
Return promise.
Running a control message to resume anOfflineAudioContext
means running these steps on the rendering thread:-
Set the
[[rendering thread state]]
on theOfflineAudioContext
torunning
. -
Start rendering the audio graph.
-
In case of failure, queue a media element task to reject promise and abort the remaining steps.
-
queue a media element task to execute the following steps:
-
Resolve promise.
-
If the
state
attribute of theOfflineAudioContext
is not already "running
":-
Set the
state
attribute of theOfflineAudioContext
to "running
". -
Queue a media element task to fire an event named
statechange
at theOfflineAudioContext
.
-
-
No parameters. -
suspend(suspendTime)
-
Schedules a suspension of the time progression in the audio context at the specified time and returns a promise. This is generally useful when manipulating the audio graph synchronously on
OfflineAudioContext
.Note that the maximum precision of suspension is the size of the render quantum and the specified suspension time will be rounded up to the nearest render quantum boundary. For this reason, it is not allowed to schedule multiple suspends at the same quantized frame. Also, scheduling should be done while the context is not running to ensure precise suspension.
Arguments for the OfflineAudioContext.suspend() method. Parameter Type Nullable Optional Description suspendTime
double
✘ ✘ Schedules a suspension of the rendering at the specified time, which is quantized and rounded up to the render quantum size. If the quantized frame number - is negative or
- is less than or equal to the current time or
- is greater than or equal to the total render duration or
- is scheduled by another suspend for the same time,
InvalidStateError
.
1.3.4. OfflineAudioContextOptions
This specifies the options to use in constructing an OfflineAudioContext
.
dictionary OfflineAudioContextOptions {unsigned long numberOfChannels = 1;required unsigned long length ;required float sampleRate ; (AudioContextRenderSizeCategory or unsigned long )renderSizeHint = "default"; };
1.3.4.1. Dictionary OfflineAudioContextOptions
Members
length
, of type unsigned long-
The length of the rendered
AudioBuffer
in sample-frames. numberOfChannels
, of type unsigned long, defaulting to1
-
The number of channels for this
OfflineAudioContext
. sampleRate
, of type float-
The sample rate for this
OfflineAudioContext
. renderSizeHint
, of type(AudioContextRenderSizeCategory or unsigned long)
, defaulting to"default"
-
A hint for the render quantum size of this
OfflineAudioContext
.
1.3.5. The OfflineAudioCompletionEvent
Interface
This is an Event
object which is dispatched to OfflineAudioContext
for legacy reasons.
[Exposed =Window ]interface OfflineAudioCompletionEvent :Event {(
constructor DOMString ,
type OfflineAudioCompletionEventInit );
eventInitDict readonly attribute AudioBuffer renderedBuffer ; };
1.3.5.1. Attributes
renderedBuffer
, of type AudioBuffer, readonly-
An
AudioBuffer
containing the rendered audio data.
1.3.5.2. OfflineAudioCompletionEventInit
dictionary OfflineAudioCompletionEventInit :EventInit {required AudioBuffer renderedBuffer ; };
1.3.5.2.1. Dictionary OfflineAudioCompletionEventInit
Members
renderedBuffer
, of type AudioBuffer-
Value to be assigned to the
renderedBuffer
attribute of the event.
1.4. The AudioBuffer
Interface
This interface represents a memory-resident audio asset. It can contain one or
more channels with each channel appearing to be 32-bit floating-point linear PCM values with a nominal range of \([-1,1]\) but the
values are not limited to this range. 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 AudioContext
s, and can be shared between an OfflineAudioContext
and an AudioContext
.
AudioBuffer
has four internal slots:
[[number of channels]]
-
The number of audio channels for this
AudioBuffer
, which is an unsigned long. [[length]]
-
The length of each channel of this
AudioBuffer
, which is an unsigned long. [[sample rate]]
-
The sample-rate, in Hz, of this
AudioBuffer
, a float. [[internal data]]
-
A data block holding the audio sample data.
[Exposed =Window ]interface AudioBuffer {constructor (AudioBufferOptions );
options readonly attribute float sampleRate ;readonly attribute unsigned long length ;readonly attribute double duration ;readonly attribute unsigned long numberOfChannels ;Float32Array getChannelData (unsigned long );
channel undefined copyFromChannel (Float32Array ,
destination unsigned long ,
channelNumber optional unsigned long = 0);
bufferOffset undefined copyToChannel (Float32Array ,
source unsigned long ,
channelNumber optional unsigned long = 0); };
bufferOffset
1.4.1. Constructors
AudioBuffer(options)
-
-
If any of the values in
options
lie outside its nominal range, throw aNotSupportedError
exception and abort the following steps. -
Let b be a new
AudioBuffer
object. -
Respectively assign the values of the attributes
numberOfChannels
,length
,sampleRate
of theAudioBufferOptions
passed in the constructor to the internal slots[[number of channels]]
,[[length]]
,[[sample rate]]
. -
Set the internal slot
[[internal data]]
of thisAudioBuffer
to the result of callingCreateByteDataBlock
(
.[[length]]
*[[number of channels]]
)Note: This initializes the underlying storage to zero.
-
Return b.
Arguments for the AudioBuffer.constructor() method. Parameter Type Nullable Optional Description options
AudioBufferOptions
✘ ✘ An AudioBufferOptions
that determine the properties for thisAudioBuffer
. -
1.4.2. Attributes
duration
, of type double, readonly-
Duration of the PCM audio data in seconds.
This is computed from the
[[sample rate]]
and the[[length]]
of theAudioBuffer
by performing a division between the[[length]]
and the[[sample rate]]
. length
, of type unsigned long, readonly-
Length of the PCM audio data in sample-frames. This MUST return the value of
[[length]]
. numberOfChannels
, of type unsigned long, readonly-
The number of discrete audio channels. This MUST return the value of
[[number of channels]]
. sampleRate
, of type float, readonly-
The sample-rate for the PCM audio data in samples per second. This MUST return the value of
[[sample rate]]
.
1.4.3. Methods
copyFromChannel(destination, channelNumber, bufferOffset)
-
The
copyFromChannel()
method copies the samples from the specified channel of theAudioBuffer
to thedestination
array.Let
buffer
be theAudioBuffer
with \(N_b\) frames, let \(N_f\) be the number of elements in thedestination
array, and \(k\) be the value ofbufferOffset
. Then the number of frames copied frombuffer
todestination
is \(\max(0, \min(N_b - k, N_f))\). If this is less than \(N_f\), then the remaining elements ofdestination
are not modified.Arguments for the AudioBuffer.copyFromChannel() method. Parameter Type Nullable Optional Description destination
Float32Array
✘ ✘ The array the channel data will be copied to. channelNumber
unsigned long
✘ ✘ The index of the channel to copy the data from. If channelNumber
is greater or equal than the number of channels of theAudioBuffer
, anIndexSizeError
MUST be thrown.bufferOffset
unsigned long
✘ ✔ An optional offset, defaulting to 0. Data from the AudioBuffer
starting at this offset is copied to thedestination
.Return type:undefined
copyToChannel(source, channelNumber, bufferOffset)
-
The
copyToChannel()
method copies the samples to the specified channel of theAudioBuffer
from thesource
array.A
UnknownError
may be thrown ifsource
cannot be copied to the buffer.Let
buffer
be theAudioBuffer
with \(N_b\) frames, let \(N_f\) be the number of elements in thesource
array, and \(k\) be the value ofbufferOffset
. Then the number of frames copied fromsource
to thebuffer
is \(\max(0, \min(N_b - k, N_f))\). If this is less than \(N_f\), then the remaining elements ofbuffer
are not modified.Arguments for the AudioBuffer.copyToChannel() method. Parameter Type Nullable Optional Description source
Float32Array
✘ ✘ The array the channel data will be copied from. channelNumber
unsigned long
✘ ✘ The index of the channel to copy the data to. If channelNumber
is greater or equal than the number of channels of theAudioBuffer
, anIndexSizeError
MUST be thrown.bufferOffset
unsigned long
✘ ✔ An optional offset, defaulting to 0. Data from the source
is copied to theAudioBuffer
starting at this offset.Return type:undefined
getChannelData(channel)
-
According to the rules described in acquire the content either allow writing into or getting a copy of the bytes stored in
[[internal data]]
in a newFloat32Array
A
UnknownError
may be thrown if the[[internal data]]
or the newFloat32Array
cannot be created.Arguments for the AudioBuffer.getChannelData() method. Parameter Type Nullable Optional Description channel
unsigned long
✘ ✘ This 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 [[number of channels]]
or anIndexSizeError
exception MUST be thrown.Return type:Float32Array
Note: The methods copyToChannel()
and copyFromChannel()
can be used to fill part of an array by
passing in a Float32Array
that’s a view onto the larger
array. When reading data from an AudioBuffer
's channels, and
the data can be processed in chunks, copyFromChannel()
should be preferred to calling getChannelData()
and
accessing the resulting array, because it may avoid unnecessary
memory allocation and copying.
An internal operation acquire the
contents of an AudioBuffer is invoked when the
contents of an AudioBuffer
are needed by some API
implementation. This operation returns immutable channel data to the
invoker.
AudioBuffer
, run the following steps:
-
If any of the
AudioBuffer
'sArrayBuffer
s are detached, returntrue
, abort these steps, and return a zero-length channel data buffer to the invoker. -
Detach all
ArrayBuffer
s for arrays previously returned bygetChannelData()
on thisAudioBuffer
.Note: Because
AudioBuffer
can only be created viacreateBuffer()
or via theAudioBuffer
constructor, this cannot throw. -
Retain the underlying
[[internal data]]
from thoseArrayBuffer
s and return references to them to the invoker. -
Attach
ArrayBuffer
s containing copies of the data to theAudioBuffer
, to be returned by the next call togetChannelData()
.
The acquire the contents of an AudioBuffer operation is invoked in the following cases:
-
When
AudioBufferSourceNode.start
is called, it acquires the contents of the node’sbuffer
. If the operation fails, nothing is played. -
When the
buffer
of anAudioBufferSourceNode
is set andAudioBufferSourceNode.start
has been previously called, the setter acquires the content of theAudioBuffer
. If the operation fails, nothing is played. -
When a
ConvolverNode
'sbuffer
is set to anAudioBuffer
it acquires the content of theAudioBuffer
. -
When the dispatch of an
AudioProcessingEvent
completes, it acquires the contents of itsoutputBuffer
.
Note: This means that copyToChannel()
cannot be used to change
the content of an AudioBuffer
currently in use by an AudioNode
that has acquired the content of an AudioBuffer since the AudioNode
will continue to use the data previously
acquired.
1.4.4. AudioBufferOptions
This specifies the options to use in constructing an AudioBuffer
. The length
and sampleRate
members are
required.
dictionary AudioBufferOptions {unsigned long numberOfChannels = 1;required unsigned long length ;required float sampleRate ; };
1.4.4.1. Dictionary AudioBufferOptions
Members
The allowed values for the members of this dictionary are constrained. See createBuffer()
.
length
, of type unsigned long-
The length in sample frames of the buffer. See
length
for constraints. numberOfChannels
, of type unsigned long, defaulting to1
-
The number of channels for the buffer. See
numberOfChannels
for constraints. sampleRate
, of type float-
The sample rate in Hz for the buffer. See
sampleRate
for constraints.
1.5. The AudioNode
Interface
AudioNode
s 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. A source node has no inputs and a single
output. Most processing nodes such as filters will have one input and
one output. Each type of AudioNode
differs in the
details of how it processes or synthesizes audio. But, in general, an AudioNode
will process its inputs (if it has
any), and generate audio for its outputs (if it has any).
Each output has one or more channels. The exact number of channels
depends on the details of the specific AudioNode
.
An output may connect to one or more AudioNode
inputs, thus fan-out is supported. An input initially has no
connections, but may be connected from one or more AudioNode
outputs, thus fan-in is supported. When the connect()
method is called to connect an output of an AudioNode
to an input of an AudioNode
, we call that a connection to the input.
Each AudioNode
input has a specific number of
channels at any given time. This number can change depending on the connection(s) made to the input. If the input has no
connections then it has one channel which is silent.
For each input, an AudioNode
performs a
mixing of all connections to that input.
Please see § 4 Channel Up-Mixing and Down-Mixing for normative requirements and details.
The processing of inputs and the internal operations of an AudioNode
take place continuously with respect to AudioContext
time, regardless of whether the node has
connected outputs, and regardless of whether these outputs ultimately
reach an AudioContext
's AudioDestinationNode
.
[Exposed =Window ]interface AudioNode :EventTarget {AudioNode connect (AudioNode destinationNode ,optional unsigned long output = 0,optional unsigned long input = 0);undefined connect (AudioParam destinationParam ,optional unsigned long output = 0);undefined disconnect ();undefined disconnect (unsigned long output );undefined disconnect (AudioNode destinationNode );undefined disconnect (AudioNode destinationNode ,unsigned long output );undefined disconnect (AudioNode destinationNode ,unsigned long output ,unsigned long input );undefined disconnect (AudioParam destinationParam );undefined disconnect (AudioParam destinationParam ,unsigned long output );readonly attribute BaseAudioContext context ;readonly attribute unsigned long numberOfInputs ;readonly attribute unsigned long numberOfOutputs ;attribute unsigned long channelCount ;attribute ChannelCountMode channelCountMode ;attribute ChannelInterpretation channelInterpretation ; };
1.5.1. AudioNode Creation
AudioNode
s can be created in two ways: by using the
constructor for this particular interface, or by using the factory method on the BaseAudioContext
or AudioContext
.
The BaseAudioContext
passed as first argument of the
constructor of an AudioNode
s is called the associated BaseAudioContext
of the AudioNode
to be created. Similarly, when using the factory
method, the associated BaseAudioContext
of the AudioNode
is the BaseAudioContext
this factory method
is called on.
AudioNode
of a particular type n using its factory method, called on a BaseAudioContext
c, execute these steps:
-
Let node be a new object of type n.
-
Let option be a dictionary of the type associated to the interface associated to this factory method.
-
For each parameter passed to the factory method, set the dictionary member of the same name on option to the value of this parameter.
-
Call the constructor for n on node with c and option as arguments.
-
Return node
AudioNode
means executing the following
steps, given the arguments context and dict passed to
the constructor of this interface.
-
Set o’s associated
BaseAudioContext
to context. -
Set its value for
numberOfInputs
,numberOfOutputs
,channelCount
,channelCountMode
,channelInterpretation
to the default value for this specific interface outlined in the section for eachAudioNode
. -
For each member of dict passed in, execute these steps, with k the key of the member, and v its value. If any exceptions is thrown when executing these steps, abort the iteration and propagate the exception to the caller of the algorithm (constructor or factory method).
-
If k is the name of an
AudioParam
on this interface, set thevalue
attribute of thisAudioParam
to v. -
Else if k is the name of an attribute on this interface, set the object associated with this attribute to v.
-
The associated interface for a factory method is the interface of the objects that are returned from this method. The associated option object for an interface is the option object that can be passed to the constructor for this interface.
AudioNode
s are EventTarget
s, as described in [DOM].
This means that it is possible to dispatch events to AudioNode
s the same way that other EventTarget
s
accept events.
enum {
ChannelCountMode "max" ,"clamped-max" ,"explicit" };
The ChannelCountMode
, in conjuction with the node’s channelCount
and channelInterpretation
values, is used to determine
the computedNumberOfChannels that controls how inputs to a
node are to be mixed. The computedNumberOfChannels is
determined as shown below. See § 4 Channel Up-Mixing and Down-Mixing for more information on how
mixing is to be done.
Enum value | Description |
---|---|
"max "
| computedNumberOfChannels is the maximum of the number of
channels of all connections to an input. In this mode channelCount is ignored.
|
"clamped-max "
| computedNumberOfChannels is determined as for "max "
and then clamped to a maximum value of the given channelCount .
|
"explicit "
| computedNumberOfChannels is the exact value as specified
by the channelCount .
|
enum {
ChannelInterpretation "speakers" ,"discrete" };
Enum value | Description |
---|---|
"speakers "
| use up-mix equations or down-mix equations. In cases where the number of
channels do not match any of these basic speaker layouts, revert
to "discrete ".
|
"discrete "
| Up-mix by filling channels until they run out then zero out remaining channels. Down-mix by filling as many channels as possible, then dropping remaining channels. |
1.5.2. AudioNode Tail-Time
An AudioNode
can have a tail-time. This means that even when the AudioNode
is fed silence, the output can be non-silent.
AudioNode
s have a non-zero tail-time if they have internal processing state
such that input in the past affects the future output. AudioNode
s
may continue to produce non-silent output for the calculated tail-time even
after the input transitions from non-silent to silent.
1.5.3. AudioNode Lifetime
AudioNode
can be actively processing during a render quantum, if any of the following conditions hold.
-
An
AudioScheduledSourceNode
is actively processing if and only if it is playing for at least part of the current rendering quantum. -
A
MediaElementAudioSourceNode
is actively processing if and only if itsmediaElement
is playing for at least part of the current rendering quantum. -
A
MediaStreamAudioSourceNode
or aMediaStreamTrackAudioSourceNode
are actively processing when the associatedMediaStreamTrack
object has areadyState
attribute equal to"live"
, amuted
attribute equal tofalse
and anenabled
attribute equal totrue
. -
A
DelayNode
in a cycle is actively processing only when the absolute value of any output sample for the current render quantum is greater than or equal to \( 2^{-126} \). -
A
ScriptProcessorNode
is actively processing when its input or output is connected. -
An
AudioWorkletNode
is actively processing when itsAudioWorkletProcessor
's[[callable process]]
returnstrue
and either its active source flag istrue
or anyAudioNode
connected to one of its inputs is actively processing. -
All other
AudioNode
s start actively processing when anyAudioNode
connected to one of its inputs is actively processing, and stops actively processing when the input that was received from other actively processingAudioNode
no longer affects the output.
Note: This takes into account AudioNode
s that have a tail-time.
AudioNode
s that are not actively processing output a single channel of
silence.
1.5.4. Attributes
channelCount
, of type unsigned long-
channelCount
is the number of channels used when up-mixing and down-mixing connections to any inputs to the node. The default value is 2 except for specific nodes where its value is specially determined. This attribute has no effect for nodes with no inputs. If this value is set to zero or to a value greater than the implementation’s maximum number of channels the implementation MUST throw aNotSupportedError
exception.In addition, some nodes have additional channelCount constraints on the possible values for the channel count:
AudioDestinationNode
-
The behavior depends on whether the destination node is the destination of an
AudioContext
orOfflineAudioContext
:AudioContext
-
The channel count MUST be between 1 and
maxChannelCount
. AnIndexSizeError
exception MUST be thrown for any attempt to set the count outside this range. OfflineAudioContext
-
The channel count cannot be changed. An
InvalidStateError
exception MUST be thrown for any attempt to change the value.
AudioWorkletNode
-
See § 1.32.4.3.2 Configuring Channels with AudioWorkletNodeOptions Configuring Channels with AudioWorkletNodeOptions.
ChannelMergerNode
-
The channel count cannot be changed, and an
InvalidStateError
exception MUST be thrown for any attempt to change the value. ChannelSplitterNode
-
The channel count cannot be changed, and an
InvalidStateError
exception MUST be thrown for any attempt to change the value. ConvolverNode
-
The channel count cannot be greater than two, and a
NotSupportedError
exception MUST be thrown for any attempt to change it to a value greater than two. DynamicsCompressorNode
-
The channel count cannot be greater than two, and a
NotSupportedError
exception MUST be thrown for any attempt to change it to a value greater than two. PannerNode
-
The channel count cannot be greater than two, and a
NotSupportedError
exception MUST be thrown for any attempt to change it to a value greater than two. ScriptProcessorNode
-
The channel count cannot be changed, and an
NotSupportedError
exception MUST be thrown for any attempt to change the value. StereoPannerNode
-
The channel count cannot be greater than two, and a
NotSupportedError
exception MUST be thrown for any attempt to change it to a value greater than two.
See § 4 Channel Up-Mixing and Down-Mixing for more information on this attribute.
channelCountMode
, of type ChannelCountMode-
channelCountMode
determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node. The default value is "max
". This attribute has no effect for nodes with no inputs.In addition, some nodes have additional channelCountMode constraints on the possible values for the channel count mode:
AudioDestinationNode
-
If the
AudioDestinationNode
is thedestination
node of anOfflineAudioContext
, then the channel count mode cannot be changed. AnInvalidStateError
exception MUST be thrown for any attempt to change the value. ChannelMergerNode
-
The channel count mode cannot be changed from "
explicit
" and anInvalidStateError
exception MUST be thrown for any attempt to change the value. ChannelSplitterNode
-
The channel count mode cannot be changed from "
explicit
" and anInvalidStateError
exception MUST be thrown for any attempt to change the value. ConvolverNode
-
The channel count mode cannot be set to "
max
", and aNotSupportedError
exception MUST be thrown for any attempt to set it to "max
". DynamicsCompressorNode
-
The channel count mode cannot be set to "
max
", and aNotSupportedError
exception MUST be thrown for any attempt to set it to "max
". PannerNode
-
The channel count mode cannot be set to "
max
", and aNotSupportedError
exception MUST be thrown for any attempt to set it to "max
". ScriptProcessorNode
-
The channel count mode cannot be changed from "
explicit
" and anNotSupportedError
exception MUST be thrown for any attempt to change the value. StereoPannerNode
-
The channel count mode cannot be set to "
max
", and aNotSupportedError
exception MUST be thrown for any attempt to set it to "max
".
See the § 4 Channel Up-Mixing and Down-Mixing section for more information on this attribute.
channelInterpretation
, of type ChannelInterpretation-
channelInterpretation
determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node. The default value is "speakers
". This attribute has no effect for nodes with no inputs.In addition, some nodes have additional channelInterpretation constraints on the possible values for the channel interpretation:
ChannelSplitterNode
-
The channel intepretation can not be changed from "
discrete
" and aInvalidStateError
exception MUST be thrown for any attempt to change the value.
See § 4 Channel Up-Mixing and Down-Mixing for more information on this attribute.
context
, of type BaseAudioContext, readonly-
The
BaseAudioContext
which owns thisAudioNode
. numberOfInputs
, of type unsigned long, readonly-
The number of inputs feeding into the
AudioNode
. For source nodes, this will be 0. This attribute is predetermined for manyAudioNode
types, but someAudioNode
s, like theChannelMergerNode
and theAudioWorkletNode
, have variable number of inputs. numberOfOutputs
, of type unsigned long, readonly-
The number of outputs coming out of the
AudioNode
. This attribute is predetermined for someAudioNode
types, but can be variable, like for theChannelSplitterNode
and theAudioWorkletNode
.
1.5.5. Methods
connect(destinationNode, output, input)
-
There can only be one connection between a given output of one specific node and a given input of another specific node. Multiple connections with the same termini are ignored.
For example:nodeA
. connect( nodeB); nodeA. connect( nodeB); will have the same effect as
nodeA
. connect( nodeB); This method returns
destination
AudioNode
object.Arguments for the AudioNode.connect(destinationNode, output, input) method. Parameter Type Nullable Optional Description destinationNode
The destination
parameter is theAudioNode
to connect to. If thedestination
parameter is anAudioNode
that has been created using anotherAudioContext
, anInvalidAccessError
MUST be thrown. That is,AudioNode
s cannot be shared betweenAudioContext
s. MultipleAudioNode
s can be connected to the sameAudioNode
, this is described in Channel Upmixing and down mixing section.output
unsigned long
✘ ✔ The output
parameter is an index describing which output of theAudioNode
from which to connect. If this parameter is out-of-bounds, anIndexSizeError
exception MUST be thrown. It is possible to connect anAudioNode
output to more than one input with multiple calls to connect(). Thus, "fan-out" is supported.input
The input
parameter is an index describing which input of the destinationAudioNode
to connect to. If this parameter is out-of-bounds, anIndexSizeError
exception MUST be thrown. It is possible to connect anAudioNode
to anotherAudioNode
which creates a cycle: anAudioNode
may connect to anotherAudioNode
, which in turn connects back to the input orAudioParam
of the firstAudioNode
.Return type:AudioNode
connect(destinationParam, output)
-
Connects the
AudioNode
to anAudioParam
, controlling the parameter value with an a-rate signal.It is possible to connect an
AudioNode
output to more than oneAudioParam
with multiple calls to connect(). Thus, "fan-out" is supported.It is possible to connect more than one
AudioNode
output to a singleAudioParam
with multiple calls to connect(). Thus, "fan-in" is supported.An
AudioParam
will take the rendered audio data from anyAudioNode
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 (thevalue
theAudioParam
would normally have without any audio connections), including any timeline changes scheduled for the parameter.The down-mixing to mono is equivalent to the down-mixing for an
AudioNode
withchannelCount
= 1,channelCountMode
= "explicit
", andchannelInterpretation
= "speakers
".There can only be one connection between a given output of one specific node and a specific
AudioParam
. Multiple connections with the same termini are ignored.For example:nodeA
. connect( param); nodeA. connect( param); will have the same effect as
nodeA
. connect( param); Arguments for the AudioNode.connect(destinationParam, output) method. Parameter Type Nullable Optional Description destinationParam
AudioParam
✘ ✘ The destination
parameter is theAudioParam
to connect to. This method does not return thedestination
AudioParam
object. IfdestinationParam
belongs to anAudioNode
that belongs to aBaseAudioContext
that is different from theBaseAudioContext
that has created theAudioNode
on which this method was called, anInvalidAccessError
MUST be thrown.output
unsigned long
✘ ✔ The output
parameter is an index describing which output of theAudioNode
from which to connect. If theparameter
is out-of-bounds, anIndexSizeError
exception MUST be thrown.Return type:undefined
disconnect()
-
Disconnects all outgoing connections from the
AudioNode
.No parameters.Return type:undefined
disconnect(output)
-
Disconnects a single output of the
AudioNode
from any otherAudioNode
orAudioParam
objects to which it is connected.Arguments for the AudioNode.disconnect(output) method. Parameter Type Nullable Optional Description output
unsigned long
✘ ✘ This parameter is an index describing which output of the AudioNode
to disconnect. It disconnects all outgoing connections from the given output. If this parameter is out-of-bounds, anIndexSizeError
exception MUST be thrown.Return type:undefined
disconnect(destinationNode)
-
Disconnects all outputs of the
AudioNode
that go to a specific destinationAudioNode
.Arguments for the AudioNode.disconnect(destinationNode) method. Parameter Type Nullable Optional Description destinationNode
The destinationNode
parameter is theAudioNode
to disconnect. It disconnects all outgoing connections to the givendestinationNode
. If there is no connection to thedestinationNode
, anInvalidAccessError
exception MUST be thrown.Return type:undefined
disconnect(destinationNode, output)
-
Disconnects a specific output of the
AudioNode
from any and all inputs of some destinationAudioNode
.Arguments for the AudioNode.disconnect(destinationNode, output) method. Parameter Type Nullable Optional Description destinationNode
The destinationNode
parameter is theAudioNode
to disconnect. If there is no connection to thedestinationNode
from the given output, anInvalidAccessError
exception MUST be thrown.output
unsigned long
✘ ✘ The output
parameter is an index describing which output of theAudioNode
from which to disconnect. If this parameter is out-of-bounds, anIndexSizeError
exception MUST be thrown.Return type:undefined
disconnect(destinationNode, output, input)
-
Disconnects a specific output of the
AudioNode
from a specific input of some destinationAudioNode
.Arguments for the AudioNode.disconnect(destinationNode, output, input) method. Parameter Type Nullable Optional Description destinationNode
The destinationNode
parameter is theAudioNode
to disconnect. If there is no connection to thedestinationNode
from the given output to the given input, anInvalidAccessError
exception MUST be thrown.output
unsigned long
✘ ✘ The output
parameter is an index describing which output of theAudioNode
from which to disconnect. If this parameter is out-of-bounds, anIndexSizeError
exception MUST be thrown.input
The input
parameter is an index describing which input of the destinationAudioNode
to disconnect. If this parameter is out-of-bounds, anIndexSizeError
exception MUST be thrown.Return type:undefined
disconnect(destinationParam)
-
Disconnects all outputs of the
AudioNode
that go to a specific destinationAudioParam
. The contribution of thisAudioNode
to the computed parameter value goes to 0 when this operation takes effect. The intrinsic parameter value is not affected by this operation.Arguments for the AudioNode.disconnect(destinationParam) method. Parameter Type Nullable Optional Description destinationParam
AudioParam
✘ ✘ The destinationParam
parameter is theAudioParam
to disconnect. If there is no connection to thedestinationParam
, anInvalidAccessError
exception MUST be thrown.Return type:undefined
disconnect(destinationParam, output)
-
Disconnects a specific output of the
AudioNode
from a specific destinationAudioParam
. The contribution of thisAudioNode
to the computed parameter value goes to 0 when this operation takes effect. The intrinsic parameter value is not affected by this operation.Arguments for the AudioNode.disconnect(destinationParam, output) method. Parameter Type Nullable Optional Description destinationParam
AudioParam
✘ ✘ The destinationParam
parameter is theAudioParam
to disconnect. If there is no connection to thedestinationParam
, anInvalidAccessError
exception MUST be thrown.output
unsigned long
✘ ✘ The output
parameter is an index describing which output of theAudioNode
from which to disconnect. If theparameter
is out-of-bounds, anIndexSizeError
exception MUST be thrown.Return type:undefined
1.5.6. AudioNodeOptions
This specifies the options that can be used in constructing all AudioNode
s. All members are optional. However, the specific
values used for each node depends on the actual node.
dictionary AudioNodeOptions {unsigned long channelCount ;ChannelCountMode channelCountMode ;ChannelInterpretation channelInterpretation ; };
1.5.6.1. Dictionary AudioNodeOptions
Members
channelCount
, of type unsigned long-
Desired number of channels for the
channelCount
attribute. channelCountMode
, of type ChannelCountMode-
Desired mode for the
channelCountMode
attribute. channelInterpretation
, of type ChannelInterpretation-
Desired mode for the
channelInterpretation
attribute.
1.6. The AudioParam
Interface
AudioParam
controls an individual aspect of an AudioNode
's functionality, such as volume. The
parameter can be set immediately to a particular value using the value
attribute. Or, value changes can be scheduled to
happen at very precise times (in the coordinate system of AudioContext
's currentTime
attribute), for envelopes, volume
fades, LFOs, filter sweeps, grain windows, etc. In this way,
arbitrary timeline-based automation curves can be set on any AudioParam
. Additionally, audio signals from the
outputs of AudioNode
s can be connected to an AudioParam
, summing with the intrinsic parameter value.
Some synthesis and processing AudioNode
s have AudioParam
s as attributes whose values MUST be taken
into account on a per-audio-sample basis. For other AudioParam
s, 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.
Implementations MUST use block processing, with each AudioNode
processing one render quantum.
For each render quantum,
the value of a k-rate parameter MUST be sampled at the time of the
very first sample-frame, and that value MUST be used for the entire
block. a-rate parameters MUST be sampled for
each sample-frame of the block.
Depending on the AudioParam
, its rate can be controlled by setting
the automationRate
attribute to either
"a-rate
" or "k-rate
". See the
description of the individual AudioParam
s for further details.
Each AudioParam
includes minValue
and maxValue
attributes that together form
the simple nominal range for the parameter. In effect,
value of the parameter is clamped to the range \([\mathrm{minValue},
\mathrm{maxValue}]\). See § 1.6.3 Computation of Value for full details.
For many AudioParam
s the minValue
and maxValue
is intended to be set to the maximum
possible range. In this case, maxValue
should be set to the most-positive-single-float value, which is 3.4028235e38.
(However, in JavaScript which only supports IEEE-754 double precision
float values, this must be written as 3.4028234663852886e38.)
Similarly, minValue
should be set
to the most-negative-single-float value, which is the
negative of the most-positive-single-float: -3.4028235e38.
(Similarly, this must be written in JavaScript as
-3.4028234663852886e38.)
An AudioParam
maintains a list of zero or more automation events. Each automation event
specifies changes to the parameter’s value over a specific time
range, in relation to its automation event time in the
time coordinate system of the AudioContext
's currentTime
attribute. The
list of automation events is maintained in ascending order of
automation event time.
The behavior of a given automation event is a function of the AudioContext
's current time, as well as the automation event
times of this event and of adjacent events in the list. The following automation methods change the
event list by adding a new event to the event list, of a type
specific to the method:
-
setValueAtTime()
-SetValue
-
linearRampToValueAtTime()
-LinearRampToValue
-
exponentialRampToValueAtTime()
-ExponentialRampToValue
-
setTargetAtTime()
-SetTarget
-
setValueCurveAtTime()
-SetValueCurve
The following rules will apply when calling these methods:
-
Automation event times are not quantized with respect to the prevailing sample rate. Formulas for determining curves and ramps are applied to the exact numerical times given when scheduling events.
-
If one of these events is added at a time where there is already one or more events, then it will be placed in the list after them, but before events whose times are after the event.
-
If setValueCurveAtTime() is called for time \(T\) and duration \(D\) and there are any events having a time strictly greater than \(T\), but strictly less than \(T + D\), then a
NotSupportedError
exception MUST be thrown. In other words, it’s not ok to schedule a value curve during a time period containing other events, but it’s ok to schedule a value curve exactly at the time of another event. -
Similarly a
NotSupportedError
exception MUST be thrown if any automation method is called at a time which is contained in \([T, T+D)\), \(T\) being the time of the curve and \(D\) its duration.
Note: AudioParam
attributes are read only, with the exception
of the value
attribute.
The automation rate of an AudioParam
can be selected setting the automationRate
attribute with one of the following
values. However, some AudioParam
s have constraints on whether the
automation rate can be changed.
enum {
AutomationRate "a-rate" ,"k-rate" };
Enum value | Description |
---|---|
"a-rate "
| This AudioParam is set for a-rate processing.
|
"k-rate "
| This AudioParam is set for k-rate processing.
|
Each AudioParam
has an internal slot [[current value]]
,
initially set to the AudioParam
's defaultValue
.
[Exposed =Window ]interface AudioParam {attribute float value ;attribute AutomationRate automationRate ;readonly attribute float defaultValue ;readonly attribute float minValue ;readonly attribute float maxValue ;AudioParam setValueAtTime (float ,
value double );
startTime AudioParam linearRampToValueAtTime (float ,
value double );
endTime AudioParam exponentialRampToValueAtTime (float ,
value double );
endTime AudioParam setTargetAtTime (float ,
target double ,
startTime float );
timeConstant AudioParam setValueCurveAtTime (sequence <float >,
values double ,
startTime double );
duration AudioParam cancelScheduledValues (double );
cancelTime AudioParam cancelAndHoldAtTime (double ); };
cancelTime
1.6.1. Attributes
automationRate
, of type AutomationRate-
The automation rate for the
AudioParam
. The default value depends on the actualAudioParam
; see the description of each individualAudioParam
for the default value.Some nodes have additional automation rate constraints as follows:
AudioBufferSourceNode
-
The
AudioParam
splaybackRate
anddetune
MUST be "k-rate
". AnInvalidStateError
must be thrown if the rate is changed to "a-rate
". DynamicsCompressorNode
-
The
AudioParam
sthreshold
,knee
,ratio
,attack
, andrelease
MUST be "k-rate
". AnInvalidStateError
must be thrown if the rate is changed to "a-rate
". PannerNode
-
If the
panningModel
is "HRTF
", the setting of theautomationRate
for anyAudioParam
of thePannerNode
is ignored. Likewise, the setting of theautomationRate
for anyAudioParam
of theAudioListener
is ignored. In this case, theAudioParam
behaves as if theautomationRate
were set to "k-rate
".
defaultValue
, of type float, readonly-
Initial value for the
value
attribute. maxValue
, of type float, readonly-
The nominal maximum value that the parameter can take. Together with
minValue
, this forms the nominal range for this parameter. minValue
, of type float, readonly-
The nominal minimum value that the parameter can take. Together with
maxValue
, this forms the nominal range for this parameter. value
, of type float-
The parameter’s floating-point value. This attribute is initialized to the
defaultValue
.Getting this attribute returns the contents of the
[[current value]]
slot. See § 1.6.3 Computation of Value for the algorithm for the value that is returned.Setting this attribute has the effect of assigning the requested value to the
[[current value]]
slot, and calling the setValueAtTime() method with the currentAudioContext
'scurrentTime
and[[current value]]
. Any exceptions that would be thrown bysetValueAtTime()
will also be thrown by setting this attribute.
1.6.2. Methods
cancelAndHoldAtTime(cancelTime)
-
This is similar to
cancelScheduledValues()
in that it cancels all scheduled parameter changes with times greater than or equal tocancelTime
. However, in addition, the automation value that would have happened atcancelTime
is then proprogated for all future time until other automation events are introduced.The behavior of the timeline in the face of
cancelAndHoldAtTime()
when automations are running and can be introduced at any time after callingcancelAndHoldAtTime()
and beforecancelTime
is reached is quite complicated. The behavior ofcancelAndHoldAtTime()
is therefore specified in the following algorithm.Let \(t_c\) be the value ofcancelTime
. Then-
Let \(E_1\) be the event (if any) at time \(t_1\) where \(t_1\) is the largest number satisfying \(t_1 \le t_c\).
-
Let \(E_2\) be the event (if any) at time \(t_2\) where \(t_2\) is the smallest number satisfying \(t_c \lt t_2\).
-
If \(E_2\) exists:
-
If \(E_2\) is a linear or exponential ramp,
-
Effectively rewrite \(E_2\) to be the same kind of ramp ending at time \(t_c\) with an end value that would be the value of the original ramp at time \(t_c\).
-
Go to step 5.
-
-
Otherwise, go to step 4.
-
-
If \(E_1\) exists:
-
If \(E_1\) is a
setTarget
event,-
Implicitly insert a
setValueAtTime
event at time \(t_c\) with the value that thesetTarget
would have at time \(t_c\). -
Go to step 5.
-
-
If \(E_1\) is a
setValueCurve
with a start time of \(t_3\) and a duration of \(d\)-
If \(t_c \gt t_3 + d\), go to step 5.
-
Otherwise,
-
Effectively replace this event with a
setValueCurve
event with a start time of \(t_3\) and a new duration of \(t_c-t_3\). However, this is not a true replacement; this automation MUST take care to produce the same output as the original, and not one computed using a different duration. (That would cause sampling of the value curve in a slightly different way, producing different results.) -
Go to step 5.
-
-
-
-
Remove all events with time greater than \(t_c\).
If no events are added, then the automation value after
cancelAndHoldAtTime()
is the constant value that the original timeline would have had at time \(t_c\).Arguments for the AudioParam.cancelAndHoldAtTime() method. Parameter Type Nullable Optional Description cancelTime
double
✘ ✘ The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the AudioContext
'scurrentTime
attribute. ARangeError
exception MUST be thrown ifcancelTime
is negative. IfcancelTime
is less thancurrentTime
, it is clamped tocurrentTime
.Return type:AudioParam
-
cancelScheduledValues(cancelTime)
-
Cancels all scheduled parameter changes with times greater than or equal to
cancelTime
. Cancelling a scheduled parameter change means removing the scheduled event from the event list. Any active automations whose automation event time is less thancancelTime
are also cancelled, and such cancellations may cause discontinuities because the original value (from before such automation) is restored immediately. Any hold values scheduled bycancelAndHoldAtTime()
are also removed if the hold time occurs aftercancelTime
.For a
setValueCurveAtTime()
, let \(T_0\) and \(T_D\) be the correspondingstartTime
andduration
, respectively of this event. Then ifcancelTime
is in the range \([T_0, T_0 + T_D]\), the event is removed from the timeline.Arguments for the AudioParam.cancelScheduledValues() method. Parameter Type Nullable Optional Description cancelTime
double
✘ ✘ The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the AudioContext
'scurrentTime
attribute. ARangeError
exception MUST be thrown ifcancelTime
is negative. IfcancelTime
is less thancurrentTime
, it is clamped tocurrentTime
.Return type:AudioParam
exponentialRampToValueAtTime(value, endTime)
-
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 during the time interval \(T_0 \leq t < T_1\) (where \(T_0\) is the time of the previous event and \(T_1\) is the
endTime
parameter passed into this method) will be calculated as:$$ v(t) = V_0 \left(\frac{V_1}{V_0}\right)^\frac{t - T_0}{T_1 - T_0} $$
where \(V_0\) is the value at the time \(T_0\) and \(V_1\) is the
value
parameter passed into this method. If \(V_0\) and \(V_1\) have opposite signs or if \(V_0\) is zero, then \(v(t) = V_0\) for \(T_0 \le t \lt T_1\).This also implies an exponential ramp to 0 is not possible. A good approximation can be achieved using
setTargetAtTime()
with an appropriately chosen time constant.If there are no more events after this ExponentialRampToValue event then for \(t \geq T_1\), \(v(t) = V_1\).
If there is no event preceding this event, the exponential ramp behaves as if
setValueAtTime(value, currentTime)
were called wherevalue
is the current value of the attribute andcurrentTime
is the contextcurrentTime
at the timeexponentialRampToValueAtTime()
is called.If the preceding event is a
SetTarget
event, \(T_0\) and \(V_0\) are chosen from the current time and value ofSetTarget
automation. That is, if theSetTarget
event has not started, \(T_0\) is the start time of the event, and \(V_0\) is the value just before theSetTarget
event starts. In this case, theExponentialRampToValue
event effectively replaces theSetTarget
event. If theSetTarget
event has already started, \(T_0\) is the current context time, and \(V_0\) is the currentSetTarget
automation value at time \(T_0\). In both cases, the automation curve is continuous.Arguments for the AudioParam.exponentialRampToValueAtTime() method. Parameter Type Nullable Optional Description value
float
✘ ✘ The value the parameter will exponentially ramp to at the given time. A RangeError
exception MUST be thrown if this value is equal to 0.endTime
double
✘ ✘ The time in the same time coordinate system as the AudioContext
'scurrentTime
attribute where the exponential ramp ends. ARangeError
exception MUST be thrown ifendTime
is negative or is not a finite number. If endTime is less thancurrentTime
, it is clamped tocurrentTime
.Return type:AudioParam
linearRampToValueAtTime(value, endTime)
-
Schedules a linear continuous change in parameter value from the previous scheduled parameter value to the given value.
The value during the time interval \(T_0 \leq t < T_1\) (where \(T_0\) is the time of the previous event and \(T_1\) is the
endTime
parameter passed into this method) will be calculated as:$$ v(t) = V_0 + (V_1 - V_0) \frac{t - T_0}{T_1 - T_0} $$
where \(V_0\) is the value at the time \(T_0\) and \(V_1\) is the
value
parameter passed into this method.If there are no more events after this LinearRampToValue event then for \(t \geq T_1\), \(v(t) = V_1\).
If there is no event preceding this event, the linear ramp behaves as if
setValueAtTime(value, currentTime)
were called wherevalue
is the current value of the attribute andcurrentTime
is the contextcurrentTime
at the timelinearRampToValueAtTime()
is called.If the preceding event is a
SetTarget
event, \(T_0\) and \(V_0\) are chosen from the current time and value ofSetTarget
automation. That is, if theSetTarget
event has not started, \(T_0\) is the start time of the event, and \(V_0\) is the value just before theSetTarget
event starts. In this case, theLinearRampToValue
event effectively replaces theSetTarget
event. If theSetTarget
event has already started, \(T_0\) is the current context time, and \(V_0\) is the currentSetTarget
automation value at time \(T_0\). In both cases, the automation curve is continuous.Arguments for the AudioParam.linearRampToValueAtTime() method. Parameter Type Nullable Optional Description value
float
✘ ✘ The value the parameter will linearly ramp to at the given time. endTime
double
✘ ✘ The time in the same time coordinate system as the AudioContext
'scurrentTime
attribute at which the automation ends. ARangeError
exception MUST be thrown ifendTime
is negative or is not a finite number. If endTime is less thancurrentTime
, it is clamped tocurrentTime
.Return type:AudioParam
setTargetAtTime(target, startTime, timeConstant)
-
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.
During the time interval: \(T_0 \leq t\), where \(T_0\) is the
startTime
parameter:$$ v(t) = V_1 + (V_0 - V_1)\, e^{-\left(\frac{t - T_0}{\tau}\right)} $$
where \(V_0\) is the initial value (the
[[current value]]
attribute) at \(T_0\) (thestartTime
parameter), \(V_1\) is equal to thetarget
parameter, and \(\tau\) is thetimeConstant
parameter.If a
LinearRampToValue
orExponentialRampToValue
event follows this event, the behavior is described inlinearRampToValueAtTime()
orexponentialRampToValueAtTime()
, respectively. For all other events, theSetTarget
event ends at the time of the next event.Arguments for the AudioParam.setTargetAtTime() method. Parameter Type Nullable Optional Description target
float
✘ ✘ The value the parameter will start changing to at the given time. startTime
double
✘ ✘ The time at which the exponential approach will begin, in the same time coordinate system as the AudioContext
'scurrentTime
attribute. ARangeError
exception MUST be thrown ifstart
is negative or is not a finite number. If startTime is less thancurrentTime
, it is clamped tocurrentTime
.timeConstant
float
✘ ✘ 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. The value MUST be non-negative or a RangeError
exception MUST be thrown. IftimeConstant
is zero, the output value jumps immediately to the final value. 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).Return type:AudioParam
setValueAtTime(value, startTime)
-
Schedules a parameter value change at the given time.
If there are no more events after this
SetValue
event, then for \(t \geq T_0\), \(v(t) = V\), where \(T_0\) is thestartTime
parameter and \(V\) is thevalue
parameter. In other words, the value will remain constant.If the next event (having time \(T_1\)) after this
SetValue
event is not of typeLinearRampToValue
orExponentialRampToValue
, then, for \(T_0 \leq t < T_1\):$$ v(t) = V $$
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 typeLinearRampToValue
orExponentialRampToValue
then please seelinearRampToValueAtTime()
orexponentialRampToValueAtTime()
, respectively.Arguments for the AudioParam.setValueAtTime() method. Parameter Type Nullable Optional Description value
float
✘ ✘ The value the parameter will change to at the given time. startTime
double
✘ ✘ The time in the same time coordinate system as the BaseAudioContext
'scurrentTime
attribute at which the parameter changes to the given value. ARangeError
exception MUST be thrown ifstartTime
is negative or is not a finite number. If startTime is less thancurrentTime
, it is clamped tocurrentTime
.Return type:AudioParam
setValueCurveAtTime(values, startTime, duration)
-
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.
Let \(T_0\) be
startTime
, \(T_D\) beduration
, \(V\) be thevalues
array, and \(N\) be the length of thevalues
array. Then, during the time interval: \(T_0 \le t < T_0 + T_D\), let$$ \begin{align*} k &= \left\lfloor \frac{N - 1}{T_D}(t-T_0) \right\rfloor \\ \end{align*} $$
Then \(v(t)\) is computed by linearly interpolating between \(V[k]\) and \(V[k+1]\),
After the end of the curve time interval (\(t \ge T_0 + T_D\)), the value will remain constant at the final curve value, until there is another automation event (if any).
An implicit call to
setValueAtTime()
is made at time \(T_0 + T_D\) with value \(V[N-1]\) so that following automations will start from the end of thesetValueCurveAtTime()
event.Arguments for the AudioParam.setValueCurveAtTime() method. Parameter Type Nullable Optional Description values
sequence<float>
✘ ✘ A sequence of float values representing a parameter value curve. These values will apply starting at the given time and lasting for the given duration. When this method is called, an internal copy of the curve is created for automation purposes. Subsequent modifications of the contents of the passed-in array therefore have no effect on the AudioParam
. AnInvalidStateError
MUST be thrown if this attribute is asequence<float>
object that has a length less than 2.startTime
double
✘ ✘ The start time in the same time coordinate system as the AudioContext
'scurrentTime
attribute at which the value curve will be applied. ARangeError
exception MUST be thrown ifstartTime
is negative or is not a finite number. If startTime is less thancurrentTime
, it is clamped tocurrentTime
.duration
double
✘ ✘ The amount of time in seconds (after the startTime
parameter) where values will be calculated according to thevalues
parameter. ARangeError
exception MUST be thrown ifduration
is not strictly positive or is not a finite number.Return type:AudioParam
1.6.3. Computation of Value
There are two different kind of AudioParam
s, simple
parameters and compound parameters. Simple parameters (the default) are used
on their own to compute the final audio output of an AudioNode
. Compound
parameters are AudioParam
s that are used with other AudioParam
s to compute a value that is then used as an input
to compute the output of an AudioNode
.
The computedValue is the final value controlling the audio DSP and is computed by the audio rendering thread during each rendering time quantum.
AudioParam
consists of two parts:
-
the paramIntrinsicValue value that is computed from the
value
attribute and any automation events. -
the paramComputedValue that is the final value controlling the audio DSP and is computed by the audio rendering thread during each render quantum.
These values MUST be computed as follows:
-
paramIntrinsicValue will be calculated at each time, which is either the value set directly to the
value
attribute, or, if there are any automation events with times before or at this time, the value as calculated from these events. If automation events are removed from a given time range, then the paramIntrinsicValue value will remain unchanged and stay at its previous value until either thevalue
attribute is directly set, or automation events are added for the time range. -
Set
[[current value]]
to the value of paramIntrinsicValue at the beginning of this render quantum. -
paramComputedValue is the sum of the paramIntrinsicValue value and the value of the input AudioParam buffer. If the sum is
NaN
, replace the sum with thedefaultValue
. -
If this
AudioParam
is a compound parameter, compute its final value with otherAudioParam
s. -
Set computedValue to paramComputedValue.
The nominal range for a computedValue are the
lower and higher values this parameter can effectively have. For simple parameters, the computedValue is clamped to
the simple nominal range for this parameter. Compound
parameters have their final value clamped to their nominal
range after having been computed from the different AudioParam
values they are composed of.
When automation methods are used, clamping is still applied. However, the automation is run as if there were no clamping at all. Only when the automation values are to be applied to the output is the clamping done as specified above.
N. p. setValueAtTime( 0 , 0 ); N. p. linearRampToValueAtTime( 4 , 1 ); N. p. linearRampToValueAtTime( 0 , 2 );
The initial slope of the curve is 4, until it reaches the maximum value of 1, at which time, the output is held constant. Finally, near time 2, the slope of the curve is -4. This is illustrated in the graph below where the dashed line indicates what would have happened without clipping, and the solid line indicates the actual expected behavior of the audioparam due to clipping to the nominal range.
1.6.4. AudioParam
Automation Example
const curveLength= 44100 ; const curve= new Float32Array( curveLength); for ( const i= 0 ; i< curveLength; ++ i) curve[ i] = Math. sin( Math. PI* i/ curveLength); const t0= 0 ; const t1= 0.1 ; const t2= 0.2 ; const t3= 0.3 ; const t4= 0.325 ; const t5= 0.5 ; const t6= 0.6 ; const t7= 0.7 ; const t8= 1.0 ; const timeConstant= 0.1 ; param. setValueAtTime( 0.2 , t0); param. setValueAtTime( 0.3 , t1); param. setValueAtTime( 0.4 , t2); param. linearRampToValueAtTime( 1 , t3); param. linearRampToValueAtTime( 0.8 , t4); param. setTargetAtTime( .5 , t4, timeConstant); // Compute where the setTargetAtTime will be at time t5 so we can make // the following exponential start at the right point so there’s no // jump discontinuity. From the spec, we have // v(t) = 0.5 + (0.8 - 0.5)*exp(-(t-t4)/timeConstant) // Thus v(t5) = 0.5 + (0.8 - 0.5)*exp(-(t5-t4)/timeConstant) param. setValueAtTime( 0.5 + ( 0.8 - 0.5 ) * Math. exp( - ( t5- t4) / timeConstant), t5); param. exponentialRampToValueAtTime( 0.75 , t6); param. exponentialRampToValueAtTime( 0.05 , t7); param. setValueCurveAtTime( curve, t7, t8- t7);
1.7. The AudioScheduledSourceNode
Interface
The interface represents the common features of source nodes such
as AudioBufferSourceNode
, ConstantSourceNode
, and OscillatorNode
.
Before a source is started (by calling start()
, the source node
MUST output silence (0). After a source has been stopped (by calling stop()
),
the source MUST then output silence (0).
AudioScheduledSourceNode
cannot be instantiated directly, but
is instead extended by the concrete interfaces for the source nodes.
An AudioScheduledSourceNode
is said to be playing when
its associated BaseAudioContext
's currentTime
is
greater or equal to the time the AudioScheduledSourceNode
is set to start,
and less than the time it’s set to stop.
AudioScheduledSourceNode
s are created with an internal boolean
slot [[source started]]
, initially
set to false.
[Exposed =Window ]interface AudioScheduledSourceNode :AudioNode {attribute EventHandler onended ;undefined start (optional double when = 0);undefined stop (optional double when = 0); };
1.7.1. Attributes
onended
, of type EventHandler-
A property used to set an event handler for the
ended
event type that is dispatched toAudioScheduledSourceNode
node types. When the source node has stopped playing (as determined by the concrete node), an event that uses theEvent
interface will be dispatched to the event handler.For all
AudioScheduledSourceNode
s, theended
event is dispatched when the stop time determined bystop()
is reached. For anAudioBufferSourceNode
, the event is also dispatched because theduration
has been reached or if the entirebuffer
has been played.
1.7.2. Methods
start(when)
-
Schedules a sound to playback at an exact time.
When this method is called, execute these steps:-
If this
AudioScheduledSourceNode
internal slot[[source started]]
is true, anInvalidStateError
exception MUST be thrown. -
Check for any errors that must be thrown due to parameter constraints described below. If any exception is thrown during this step, abort those steps.
-
Set the internal slot
[[source started]]
on thisAudioScheduledSourceNode
totrue
. -
Queue a control message to start the
AudioScheduledSourceNode
, including the parameter values in the message. -
Send a control message to the associated
AudioContext
to start running its rendering thread only when all the following conditions are met:-
The context’s
[[control thread state]]
is "suspended
". -
The context is allowed to start.
-
[[suspended by user]]
flag isfalse
.
NOTE: This can allow
start()
to start anAudioContext
that is currently allowed to start, but has previously been prevented from starting. -
Arguments for the AudioScheduledSourceNode.start(when) method. Parameter Type Nullable Optional Description when
double
✘ ✔ The when
parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as theAudioContext
'scurrentTime
attribute. When the signal emitted by theAudioScheduledSourceNode
depends on the sound’s start time, the exact value ofwhen
is always used without rounding to the nearest sample frame. If 0 is passed in for this value or if the value is less thancurrentTime
, then the sound will start playing immediately. ARangeError
exception MUST be thrown ifwhen
is negative.Return type:undefined
-
stop(when)
-
Schedules a sound to stop playback at an exact time. If
stop
is called again after already having been called, the last invocation will be the only one applied; stop times set by previous calls will not be applied, unless the buffer has already stopped prior to any subsequent calls. If the buffer has already stopped, further calls tostop
will have no effect. If a stop time is reached prior to the scheduled start time, the sound will not play.When this method is called, execute these steps:-
If this
AudioScheduledSourceNode
internal slot[[source started]]
is nottrue
, anInvalidStateError
exception MUST be thrown. -
Check for any errors that must be thrown due to parameter constraints described below.
-
Queue a control message to stop the
AudioScheduledSourceNode
, including the parameter values in the message.
If the node is anAudioBufferSourceNode
, running a control message to stop theAudioBufferSourceNode
means invoking thehandleStop()
function in the playback algorithm.Arguments for the AudioScheduledSourceNode.stop(when) method. Parameter Type Nullable Optional Description when
double
✘ ✔ The when
parameter describes at what time (in seconds) the source should stop playing. It is in the same time coordinate system as theAudioContext
'scurrentTime
attribute. If 0 is passed in for this value or if the value is less thancurrentTime
, then the sound will stop playing immediately. ARangeError
exception MUST be thrown ifwhen
is negative.Return type:undefined
-
1.8. The AnalyserNode
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.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | This output may be left unconnected. |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | No |
[Exposed =Window ]interface AnalyserNode :AudioNode {constructor (BaseAudioContext ,
context optional AnalyserOptions = {});
options undefined getFloatFrequencyData (Float32Array );
array undefined getByteFrequencyData (Uint8Array );
array undefined getFloatTimeDomainData (Float32Array );
array undefined getByteTimeDomainData (Uint8Array );
array attribute unsigned long fftSize ;readonly attribute unsigned long frequencyBinCount ;attribute double minDecibels ;attribute double maxDecibels ;attribute double smoothingTimeConstant ; };
1.8.1. Constructors
AnalyserNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the AnalyserNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newAnalyserNode
will be associated with.options
AnalyserOptions
✘ ✔ Optional initial parameter value for this AnalyserNode
.
1.8.2. Attributes
fftSize
, of type unsigned long-
The size of the FFT used for frequency-domain analysis (in sample-frames). This MUST be a power of two in the range 32 to 32768, otherwise an
IndexSizeError
exception MUST be thrown. The default value is 2048. Note that large FFT sizes can be costly to compute.If the
fftSize
is changed to a different value, then all state associated with smoothing of the frequency data (forgetByteFrequencyData()
andgetFloatFrequencyData()
) is reset. That is the previous block, \(\hat{X}_{-1}[k]\), used for smoothing over time is set to 0 for all \(k\).Note that increasing
fftSize
does mean that the current time-domain data must be expanded to include past frames that it previously did not. This means that theAnalyserNode
effectively MUST keep around the last 32768 sample-frames and the current time-domain data is the most recentfftSize
sample-frames out of that. frequencyBinCount
, of type unsigned long, readonly-
Half the FFT size.
maxDecibels
, of type double-
maxDecibels
is the maximum power value in the scaling range for the FFT analysis data for conversion to unsigned byte values. The default value is -30. If the value of this attribute is set to a value less than or equal tominDecibels
, anIndexSizeError
exception MUST be thrown. minDecibels
, of type double-
minDecibels
is the minimum power value in the scaling range for the FFT analysis data for conversion to unsigned byte values. The default value is -100. If the value of this attribute is set to a value more than or equal tomaxDecibels
, anIndexSizeError
exception MUST be thrown. smoothingTimeConstant
, of type double-
A value from 0 -> 1 where 0 represents no time averaging with the last analysis frame. The default value is 0.8. If the value of this attribute is set to a value less than 0 or more than 1, an
IndexSizeError
exception MUST be thrown.
1.8.3. Methods
getByteFrequencyData(array)
-
Write the current frequency data into array. If array’s byte length is less than
frequencyBinCount
, the excess elements will be dropped. If array’s byte length is greater than thefrequencyBinCount
, the excess elements will be ignored. The most recentfftSize
frames are used in computing the frequency data.If another call to
getByteFrequencyData()
orgetFloatFrequencyData()
occurs within the same render quantum as a previous call, the current frequency data is not updated with the same data. Instead, the previously computed data is returned.The values stored in the unsigned byte array are computed in the following way. Let \(Y[k]\) be the current frequency data as described in FFT windowing and smoothing. Then the byte value, \(b[k]\), is
$$ b[k] = \left\lfloor \frac{255}{\mbox{dB}_{max} - \mbox{dB}_{min}} \left(Y[k] - \mbox{dB}_{min}\right) \right\rfloor $$
where \(\mbox{dB}_{min}\) is
minDecibels
and \(\mbox{dB}_{max}\) is
. If \(b[k]\) lies outside the range of 0 to 255, \(b[k]\) is clipped to lie in that range.maxDecibels
Arguments for the AnalyserNode.getByteFrequencyData() method. Parameter Type Nullable Optional Description array
Uint8Array
✘ ✘ This parameter is where the frequency-domain analysis data will be copied. Return type:undefined
getByteTimeDomainData(array)
-
Write the current time-domain data (waveform data) into array. If array’s byte length is less than
fftSize
, the excess elements will be dropped. If array’s byte length is greater than thefftSize
, the excess elements will be ignored. The most recentfftSize
frames are used in computing the byte data.The values stored in the unsigned byte array are computed in the following way. Let \(x[k]\) be the time-domain data. Then the byte value, \(b[k]\), is
$$ b[k] = \left\lfloor 128(1 + x[k]) \right\rfloor. $$
If \(b[k]\) lies outside the range 0 to 255, \(b[k]\) is clipped to lie in that range.
Arguments for the AnalyserNode.getByteTimeDomainData() method. Parameter Type Nullable Optional Description array
Uint8Array
✘ ✘ This parameter is where the time-domain sample data will be copied. Return type:undefined
getFloatFrequencyData(array)
-
Write the current frequency data into array. If array has fewer elements than the
frequencyBinCount
, the excess elements will be dropped. If array has more elements than thefrequencyBinCount
, the excess elements will be ignored. The most recentfftSize
frames are used in computing the frequency data.If another call to
getFloatFrequencyData()
orgetByteFrequencyData()
occurs within the same render quantum as a previous call, the current frequency data is not updated with the same data. Instead, the previously computed data is returned.The frequency data are in dB units.
Arguments for the AnalyserNode.getFloatFrequencyData() method. Parameter Type Nullable Optional Description array
Float32Array
✘ ✘ This parameter is where the frequency-domain analysis data will be copied. Return type:undefined
getFloatTimeDomainData(array)
-
Write the current time-domain data (waveform data) into array. If array has fewer elements than the value of
fftSize
, the excess elements will be dropped. If array has more elements than the value offftSize
, the excess elements will be ignored. The most recentfftSize
frames are written (after downmixing).Arguments for the AnalyserNode.getFloatTimeDomainData() method. Parameter Type Nullable Optional Description array
Float32Array
✘ ✘ This parameter is where the time-domain sample data will be copied. Return type:undefined
1.8.4. AnalyserOptions
This specifies the options to be used when constructing an AnalyserNode
. All members are optional; if not
specified, the normal default values are used to construct the
node.
dictionary AnalyserOptions :AudioNodeOptions {unsigned long fftSize = 2048;double maxDecibels = -30;double minDecibels = -100;double smoothingTimeConstant = 0.8; };
1.8.4.1. Dictionary AnalyserOptions
Members
fftSize
, of type unsigned long, defaulting to2048
-
The desired initial size of the FFT for frequency-domain analysis.
maxDecibels
, of type double, defaulting to-30
-
The desired initial maximum power in dB for FFT analysis.
minDecibels
, of type double, defaulting to-100
-
The desired initial minimum power in dB for FFT analysis.
smoothingTimeConstant
, of type double, defaulting to0.8
-
The desired initial smoothing constant for the FFT analysis.
1.8.5. Time-Domain Down-Mixing
When the current time-domain data are computed, the
input signal must be down-mixed to mono as if channelCount
is 1, channelCountMode
is
"max
" and channelInterpretation
is "speakers
". This is independent of the
settings for the AnalyserNode
itself. The most recent fftSize
frames are used for the
down-mixing operation.
1.8.6. FFT Windowing and Smoothing over Time
When the current frequency data are computed, the following operations are to be performed:
-
Compute the current time-domain data.
-
Apply a Blackman window to the time domain input data.
-
Apply a Fourier transform to the windowed time domain input data to get real and imaginary frequency data.
-
Smooth over time the frequency domain data.
In the following, let \(N\) be the value of the fftSize
attribute of this AnalyserNode
.
$$ \begin{align*} \alpha &= \mbox{0.16} \\ a_0 &= \frac{1-\alpha}{2} \\ a_1 &= \frac{1}{2} \\ a_2 &= \frac{\alpha}{2} \\ w[n] &= a_0 - a_1 \cos\frac{2\pi n}{N} + a_2 \cos\frac{4\pi n}{N}, \mbox{ for } n = 0, \ldots, N - 1 \end{align*} $$
The windowed signal \(\hat{x}[n]\) is
$$ \hat{x}[n] = x[n] w[n], \mbox{ for } n = 0, \ldots, N - 1 $$
$$ X[k] = \frac{1}{N} \sum_{n = 0}^{N - 1} \hat{x}[n]\, W^{-kn}_{N} $$
for \(k = 0, \dots, N/2-1\) where \(W_N = e^{2\pi i/N}\).
-
Let \(\hat{X}_{-1}[k]\) be the result of this operation on the previous block. The previous block is defined as being the buffer computed by the previous smoothing over time operation, or an array of \(N\) zeros if this is the first time we are smoothing over time.
-
Let \(\tau\) be the value of the
smoothingTimeConstant
attribute for thisAnalyserNode
. -
Let \(X[k]\) be the result of applying a Fourier transform of the current block.
Then the smoothed value, \(\hat{X}[k]\), is computed by
$$ \hat{X}[k] = \tau\, \hat{X}_{-1}[k] + (1 - \tau)\, \left|X[k]\right| $$
-
If \(\hat{X}[k]\) is
NaN
, positive infinity or negative infinity, set \(\hat{X}[k]\) = 0.
for \(k = 0, \ldots, N - 1\).
$$ Y[k] = 20\log_{10}\hat{X}[k] $$
for \(k = 0, \ldots, N-1\).
This array, \(Y[k]\), is copied to the output array for getFloatFrequencyData()
. For getByteFrequencyData()
, the \(Y[k]\) is clipped to lie
between minDecibels
and
and then scaled to fit in an
unsigned byte such that maxDecibels
minDecibels
is
represented by the value 0 and
is
represented by the value 255.maxDecibels
1.9. The AudioBufferSourceNode
Interface
This interface represents an audio source from an in-memory audio
asset in an AudioBuffer
. It is useful for playing audio
assets which require a high degree of scheduling flexibility and
accuracy. If sample-accurate playback of network- or disk-backed
assets is required, an implementer should use AudioWorkletNode
to implement playback.
The start()
method is used to
schedule when sound playback will happen. The start()
method may not be
issued multiple times. The playback will stop automatically when the
buffer’s audio data has been completely played (if the loop
attribute is false
), or when the stop()
method has been
called and the specified time has been reached. Please see more
details in the start()
and stop()
descriptions.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 0 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | No |
The number of channels of the output equals the number of channels of the
AudioBuffer assigned to the buffer
attribute,
or is one channel of silence if buffer
is null
.
In addition, if the buffer has more than one channel,
then the AudioBufferSourceNode
output must change to a single channel
of silence at the beginning of a render quantum after the time at which any one
of the following conditions holds:
-
the end of the
buffer
has been reached; -
the
duration
has been reached; -
the
stop
time has been reached.
A playhead position for an AudioBufferSourceNode
is
defined as any quantity representing a time offset in seconds,
relative to the time coordinate of the first sample frame in the
buffer. Such values are to be considered independently from the
node’s playbackRate
and detune
parameters.
In general, playhead positions may be subsample-accurate and need not
refer to exact sample frame positions. They may assume valid values
between 0 and the duration of the buffer.
The playbackRate
and detune
attributes form a compound parameter. They are used together to determine a computedPlaybackRate value:
computedPlaybackRate(t) = playbackRate(t) * pow(2, detune(t) / 1200)
The nominal range for this compound parameter is \((-\infty, \infty)\).
AudioBufferSourceNode
s are created with an internal boolean
slot [[buffer set]]
, initially set to false.
[Exposed =Window ]interface AudioBufferSourceNode :AudioScheduledSourceNode {constructor (BaseAudioContext ,
context optional AudioBufferSourceOptions = {});
options attribute AudioBuffer ?buffer ;readonly attribute AudioParam playbackRate ;readonly attribute AudioParam detune ;attribute boolean loop ;attribute double loopStart ;attribute double loopEnd ;undefined start (optional double when = 0,optional double offset ,optional double duration ); };
1.9.1. Constructors
AudioBufferSourceNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the AudioBufferSourceNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newAudioBufferSourceNode
will be associated with.options
AudioBufferSourceOptions
✘ ✔ Optional initial parameter value for this AudioBufferSourceNode
.
1.9.2. Attributes
buffer
, of type AudioBuffer, nullable-
Represents the audio asset to be played.
To set thebuffer
attribute, execute these steps:-
Let new buffer be the
AudioBuffer
ornull
value to be assigned tobuffer
. -
If new buffer is not
null
and[[buffer set]]
is true, throw anInvalidStateError
and abort these steps. -
If new buffer is not
null
, set[[buffer set]]
to true. -
Assign new buffer to the
buffer
attribute. -
If
start()
has previously been called on this node, perform the operation acquire the content onbuffer
.
-
detune
, of type AudioParam, readonly-
An additional parameter, in cents, to modulate the speed at which is rendered the audio stream. This parameter is a compound parameter with
playbackRate
to form a computedPlaybackRate.Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" k-rate
"Has automation rate constraints loop
, of type boolean-
Indicates if the region of audio data designated by
loopStart
andloopEnd
should be played continuously in a loop. The default value isfalse
. loopEnd
, of type double-
An optional playhead position where looping should end if the
loop
attribute is true. Its value is exclusive of the content of the loop. Its defaultvalue
is 0, and it may usefully be set to any value between 0 and the duration of the buffer. IfloopEnd
is less than or equal to 0, or ifloopEnd
is greater than the duration of the buffer, looping will end at the end of the buffer. loopStart
, of type double-
An optional playhead position where looping should begin if the
loop
attribute is true. Its defaultvalue
is 0, and it may usefully be set to any value between 0 and the duration of the buffer. IfloopStart
is less than 0, looping will begin at 0. IfloopStart
is greater than the duration of the buffer, looping will begin at the end of the buffer. playbackRate
, of type AudioParam, readonly-
The speed at which to render the audio stream. This is a compound parameter with
detune
to form a computedPlaybackRate.Parameter Value Notes defaultValue
1 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" k-rate
"Has automation rate constraints
1.9.3. Methods
start(when, offset, duration)
-
Schedules a sound to playback at an exact time.
When this method is called, execute these steps:-
If this
AudioBufferSourceNode
internal slot[[source started]]
istrue
, anInvalidStateError
exception MUST be thrown. -
Check for any errors that must be thrown due to parameter constraints described below. If any exception is thrown during this step, abort those steps.
-
Set the internal slot
[[source started]]
on thisAudioBufferSourceNode
totrue
. -
Queue a control message to start the
AudioBufferSourceNode
, including the parameter values in the message. -
Acquire the contents of the
buffer
if thebuffer
has been set. -
Send a control message to the associated
AudioContext
to start running its rendering thread only when all the following conditions are met:-
The context’s
[[control thread state]]
issuspended
. -
The context is allowed to start.
-
[[suspended by user]]
flag isfalse
.
NOTE: This can allow
start()
to start anAudioContext
that is currently allowed to start, but has previously been prevented from starting. -
Running a control message to start theAudioBufferSourceNode
means invoking thehandleStart()
function in the playback algorithm which follows.Arguments for the AudioBufferSourceNode.start(when, offset, duration) method. Parameter Type Nullable Optional Description when
double
✘ ✔ The when
parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as theAudioContext
'scurrentTime
attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. ARangeError
exception MUST be thrown ifwhen
is negative.offset
double
✘ ✔ The offset
parameter supplies a playhead position where playback will begin. If 0 is passed in for this value, then playback will start from the beginning of the buffer. ARangeError
exception MUST be thrown ifoffset
is negative. Ifoffset
is greater thanloopEnd
,playbackRate
is positive or zero, andloop
istrue
, playback will begin atloopEnd
. Ifoffset
is greater thanloopStart
,playbackRate
is negative, andloop
istrue
, playback will begin atloopStart
.offset
is silently clamped to [0,duration
], whenstartTime
is reached, whereduration
is the value of theduration
attribute of theAudioBuffer
set to thebuffer
attribute of thisAudioBufferSourceNode
.duration
double
✘ ✔ The duration
parameter describes the duration of sound to be played, expressed as seconds of total buffer content to be output, including any whole or partial loop iterations. The units ofduration
are independent of the effects ofplaybackRate
. For example, aduration
of 5 seconds with a playback rate of 0.5 will output 5 seconds of buffer content at half speed, producing 10 seconds of audible output. ARangeError
exception MUST be thrown ifduration
is negative.Return type:undefined
-
1.9.4. AudioBufferSourceOptions
This specifies options for constructing a AudioBufferSourceNode
. All members are
optional; if not specified, the normal default is used in
constructing the node.
dictionary AudioBufferSourceOptions {AudioBuffer ?buffer ;float detune = 0;boolean loop =false ;double loopEnd = 0;double loopStart = 0;float playbackRate = 1; };
1.9.4.1. Dictionary AudioBufferSourceOptions
Members
buffer
, of type AudioBuffer, nullable-
The audio asset to be played. This is equivalent to assigning
buffer
to thebuffer
attribute of theAudioBufferSourceNode
. detune
, of type float, defaulting to0
-
The initial value for the
detune
AudioParam. loop
, of type boolean, defaulting tofalse
-
The initial value for the
loop
attribute. loopEnd
, of type double, defaulting to0
-
The initial value for the
loopEnd
attribute. loopStart
, of type double, defaulting to0
-
The initial value for the
loopStart
attribute. playbackRate
, of type float, defaulting to1
-
The initial value for the
playbackRate
AudioParam.
1.9.5. Looping
This section is non-normative. Please see the playback algorithm for normative requirements.
Setting the loop
attribute to true causes playback of the region of the buffer
defined by the endpoints loopStart
and loopEnd
to continue indefinitely, once
any part of the looped region has been played. While loop
remains true,
looped playback will continue until one of the following occurs:
-
stop()
is called, -
the scheduled stop time has been reached,
-
the
duration
has been exceeded, ifstart()
was called with aduration
value.
The body of the loop is considered to occupy a region from loopStart
up to, but
not including, loopEnd
. The direction of playback of
the looped region respects the sign of the node’s playback rate.
For positive playback rates, looping occurs from loopStart
to loopEnd
; for negative rates, looping
occurs from loopEnd
to loopStart
.
Looping does not affect the interpretation of the offset
argument of start()
. Playback always
starts at the requested offset, and looping only begins once the
body of the loop is encountered during playback.
The effective loop start and end points are required to lie within
the range of zero and the buffer duration, as specified in the
algorithm below. loopEnd
is further constrained to be at
or after loopStart
. If
any of these constraints are violated, the loop is considered to
include the entire buffer contents.
Loop endpoints have subsample accuracy. When endpoints do not fall on exact sample frame offsets, or when the playback rate is not equal to 1, playback of the loop is interpolated to splice the beginning and end of the loop together just as if the looped audio occurred in sequential, non-looped regions of the buffer.
Loop-related properties may be varied during playback of the buffer, and in general take effect on the next rendering quantum. The exact results are defined by the normative playback algorithm which follows.
The default values of the loopStart
and loopEnd
attributes are both 0. Since a loopEnd
value of zero
is equivalent to the length of the buffer, the default endpoints
cause the entire buffer to be included in the loop.
Note that the values of the loop endpoints are expressed as time
offsets in terms of the sample rate of the buffer, meaning that
these values are independent of the node’s playbackRate
parameter which can vary
dynamically during the course of playback.
1.9.6. Playback of AudioBuffer Contents
This normative section specifies the playback of the contents of the buffer, accounting for the fact that playback is influenced by the following factors working in combination:
-
A starting offset, which can be expressed with sub-sample precision.
-
Loop points, which can be expressed with sub-sample precision and can vary dynamically during playback.
-
Playback rate and detuning parameters, which combine to yield a single computedPlaybackRate that can assume finite values which may be positive or negative.
The algorithm to be followed internally to generate output from an AudioBufferSourceNode
conforms to the following principles:
-
Resampling of the buffer may be performed arbitrarily by the UA at any desired point to increase the efficiency or quality of the output.
-
Sub-sample start offsets or loop points may require additional interpolation between sample frames.
-
The playback of a looped buffer should behave identically to an unlooped buffer containing consecutive occurrences of the looped audio content, excluding any effects from interpolation.
The description of the algorithm is as follows:
let buffer; // AudioBuffer employed by this node let context; // AudioContext employed by this node // The following variables capture attribute and AudioParam values for the node. // They are updated on a k-rate basis, prior to each invocation of process(). let loop; let detune; let loopStart; let loopEnd; let playbackRate; // Variables for the node's playback parameters let start= 0 , offset= 0 , duration= Infinity ; // Set by start() let stop= Infinity ; // Set by stop() // Variables for tracking node's playback state let bufferTime= 0 , started= false , enteredLoop= false ; let bufferTimeElapsed= 0 ; let dt= 1 / context. sampleRate; // Handle invocation of start method call function handleStart( when, pos, dur) { if ( arguments. length>= 1 ) { start= when; } offset= pos; if ( arguments. length>= 3 ) { duration= dur; } } // Handle invocation of stop method call function handleStop( when) { if ( arguments. length>= 1 ) { stop= when; } else { stop= context. currentTime; } } // Interpolate a multi-channel signal value for some sample frame. // Returns an array of signal values. function playbackSignal( position) { /* This function provides the playback signal function for buffer, which is a function that maps from a playhead position to a set of output signal values, one for each output channel. If |position| corresponds to the location of an exact sample frame in the buffer, this function returns that frame. Otherwise, its return value is determined by a UA-supplied algorithm that interpolates sample frames in the neighborhood of |position|. If |position| is greater than or equal to |loopEnd| and there is no subsequent sample frame in buffer, then interpolation should be based on the sequence of subsequent frames beginning at |loopStart|. */ ... } // Generate a single render quantum of audio to be placed // in the channel arrays defined by output. Returns an array // of |numberOfFrames| sample frames to be output. function process( numberOfFrames) { let currentTime= context. currentTime; // context time of next rendered frame const output= []; // accumulates rendered sample frames // Combine the two k-rate parameters affecting playback rate const computedPlaybackRate= playbackRate* Math. pow( 2 , detune/ 1200 ); // Determine loop endpoints as applicable let actualLoopStart, actualLoopEnd; if ( loop&& buffer!= null ) { if ( loopStart>= 0 && loopEnd> 0 && loopStart< loopEnd) { actualLoopStart= loopStart; actualLoopEnd= Math. min( loopEnd, buffer. duration); } else { actualLoopStart= 0 ; actualLoopEnd= buffer. duration; } } else { // If the loop flag is false, remove any record of the loop having been entered enteredLoop= false ; } // Handle null buffer case if ( buffer== null ) { stop= currentTime; // force zero output for all time } // Render each sample frame in the quantum for ( let index= 0 ; index< numberOfFrames; index++ ) { // Check that currentTime and bufferTimeElapsed are // within allowable range for playback if ( currentTime< start|| currentTime>= stop|| bufferTimeElapsed>= duration) { output. push( 0 ); // this sample frame is silent currentTime+= dt; continue ; } if ( ! started) { // Take note that buffer has started playing and get initial // playhead position. if ( loop&& computedPlaybackRate>= 0 && offset>= actualLoopEnd) { offset= actualLoopEnd; } if ( computedPlaybackRate< 0 && loop&& offset< actualLoopStart) { offset= actualLoopStart; } bufferTime= offset; started= true ; } // Handle loop-related calculations if ( loop) { // Determine if looped portion has been entered for the first time if ( ! enteredLoop) { if ( offset< actualLoopEnd&& bufferTime>= actualLoopStart) { // playback began before or within loop, and playhead is // now past loop start enteredLoop= true ; } if ( offset>= actualLoopEnd&& bufferTime< actualLoopEnd) { // playback began after loop, and playhead is now prior // to the loop end enteredLoop= true ; } } // Wrap loop iterations as needed. Note that enteredLoop // may become true inside the preceding conditional. if ( enteredLoop) { while ( bufferTime>= actualLoopEnd) { bufferTime-= actualLoopEnd- actualLoopStart; } while ( bufferTime< actualLoopStart) { bufferTime+= actualLoopEnd- actualLoopStart; } } } if ( bufferTime>= 0 && bufferTime< buffer. duration) { output. push( playbackSignal( bufferTime)); } else { output. push( 0 ); // past end of buffer, so output silent frame } bufferTime+= dt* computedPlaybackRate; bufferTimeElapsed+= dt* computedPlaybackRate; currentTime+= dt; } // End of render quantum loop if ( currentTime>= stop) { // End playback state of this node. No further invocations of process() // will occur. Schedule a change to set the number of output channels to 1. } return output; }
The following non-normative figures illustrate the behavior of the algorithm in assorted key scenarios. Dynamic resampling of the buffer is not considered, but as long as the times of loop positions are not changed this does not materially affect the resulting playback. In all figures, the following conventions apply:
-
context sample rate is 1000 Hz
-
AudioBuffer
content is shown with the first sample frame at the x origin. -
output signals are shown with the sample frame located at time
start
at the x origin. -
linear interpolation is depicted throughout, although a UA could employ other interpolation techniques.
-
the
duration
values noted in the figures refer to thebuffer
, not arguments tostart()
This figure illustrates basic playback of a buffer, with a simple loop that ends after the last sample frame in the buffer:
This figure illustrates playbackRate
interpolation,
showing half-speed playback of buffer contents in which every other
output sample frame is interpolated. Of particular note is the last
sample frame in the looped output, which is interpolated using the
loop start point:
This figure illustrates sample rate interpolation, showing playback of a buffer whose sample rate is 50% of the context sample rate, resulting in a computed playback rate of 0.5 that corrects for the difference in sample rate between the buffer and the context. The resulting output is the same as the preceding example, but for different reasons.
This figure illustrates subsample offset playback, in which the offset within the buffer begins at exactly half a sample frame. Consequently, every output frame is interpolated:
This figure illustrates subsample loop playback, showing how fractional frame offsets in the loop endpoints map to interpolated data points in the buffer that respect these offsets as if they were references to exact sample frames:
1.10. The AudioDestinationNode
Interface
This is an AudioNode
representing the final audio
destination and is what the user will ultimately hear. It can often
be considered as an audio output device which is connected to
speakers. All rendered audio to be heard will be routed to this node,
a "terminal" node in the AudioContext
's routing
graph. There is only a single AudioDestinationNode per AudioContext
, provided through the destination
attribute of AudioContext
.
The output of a AudioDestinationNode
is produced
by summing its input, allowing to
capture the output of an AudioContext
into, for
example, a MediaStreamAudioDestinationNode
, or a MediaRecorder
(described in [mediastream-recording]).
The AudioDestinationNode
can be either the destination of an AudioContext
or OfflineAudioContext
, and the channel
properties depend on what the context is.
For an AudioContext
, the defaults are
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "explicit "
| |
channelInterpretation
| "speakers "
| |
tail-time | No |
The channelCount
can be set to any
value less than or equal to maxChannelCount
. An IndexSizeError
exception MUST 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 channelCount
to 8, and render 8
channels of output.
For an OfflineAudioContext
, the defaults are
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| numberOfChannels | |
channelCountMode
| "explicit "
| |
channelInterpretation
| "speakers "
| |
tail-time | No |
where numberOfChannels
is the number of channels
specified when constructing the OfflineAudioContext
. This
value may not be changed; a NotSupportedError
exception MUST be thrown if channelCount
is changed to a
different value.
[Exposed =Window ]interface AudioDestinationNode :AudioNode {readonly attribute unsigned long maxChannelCount ; };
1.10.1. Attributes
maxChannelCount
, of type unsigned long, readonly-
The maximum number of channels that the
channelCount
attribute can be set to. AnAudioDestinationNode
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.maxChannelCount
is the maximum number of channels that this hardware is capable of supporting.
1.11. The AudioListener
Interface
This interface represents the position and orientation of the person
listening to the audio scene. All PannerNode
objects spatialize in relation to the BaseAudioContext
's listener
. See § 6 Spatialization/Panning for more details about spatialization.
The positionX
, positionY
, and positionZ
parameters represent
the location of the listener in 3D Cartesian coordinate space. PannerNode
objects use this position relative to
individual audio sources for spatialization.
The forwardX
, forwardY
, and forwardZ
parameters represent a
direction vector in 3D space. Both a forward
vector and
an up
vector are used to determine the orientation of
the listener. In simple human terms, the forward
vector
represents which direction the person’s nose is pointing. The up
vector represents the direction the top of a person’s
head is pointing. These two vectors are expected to be linearly
independent. For normative requirements of how these values are to be
interpreted, see the § 6 Spatialization/Panning section.
[Exposed =Window ]interface AudioListener {readonly attribute AudioParam positionX ;readonly attribute AudioParam positionY ;readonly attribute AudioParam positionZ ;readonly attribute AudioParam forwardX ;readonly attribute AudioParam forwardY ;readonly attribute AudioParam forwardZ ;readonly attribute AudioParam upX ;readonly attribute AudioParam upY ;readonly attribute AudioParam upZ ;undefined setPosition (float ,
x float ,
y float );
z undefined setOrientation (float ,
x float ,
y float ,
z float ,
xUp float ,
yUp float ); };
zUp
1.11.1. Attributes
forwardX
, of type AudioParam, readonly-
Sets the x coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" forwardY
, of type AudioParam, readonly-
Sets the y coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" forwardZ
, of type AudioParam, readonly-
Sets the z coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
-1 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" positionX
, of type AudioParam, readonly-
Sets the x coordinate position of the audio listener in a 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" positionY
, of type AudioParam, readonly-
Sets the y coordinate position of the audio listener in a 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" positionZ
, of type AudioParam, readonly-
Sets the z coordinate position of the audio listener in a 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" upX
, of type AudioParam, readonly-
Sets the x coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" upY
, of type AudioParam, readonly-
Sets the y coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
1 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
" upZ
, of type AudioParam, readonly-
Sets the z coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space.
Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
"
1.11.2. Methods
setOrientation(x, y, z, xUp, yUp, zUp)
-
This method is DEPRECATED. It is equivalent to setting
forwardX
.value
,forwardY
.value
,forwardZ
.value
,upX
.value
,upY
.value
, andupZ
.value
directly with the givenx
,y
,z
,xUp
,yUp
, andzUp
values, respectively.Consequently, if any of the
forwardX
,forwardY
,forwardZ
,upX
,upY
andupZ
AudioParam
s have an automation curve set usingsetValueCurveAtTime()
at the time this method is called, aNotSupportedError
MUST be thrown.setOrientation()
describes which direction the listener is pointing in the 3D cartesian coordinate space. Both a forward vector and an up vector are provided. In simple human terms, the forward vector represents which direction the person’s nose is pointing. The up vector represents the direction the top of a person’s head is pointing. These two vectors are expected to be linearly independent. For normative requirements of how these values are to be interpreted, see the § 6 Spatialization/Panning.The
x
,y
, andz
parameters represent a forward direction vector in 3D space, with the default value being (0,0,-1).The
xUp
,yUp
, andzUp
parameters represent an up direction vector in 3D space, with the default value being (0,1,0).Arguments for the AudioListener.setOrientation() method. Parameter Type Nullable Optional Description x
float
✘ ✘ forward x direction fo the AudioListener
y
float
✘ ✘ forward y direction fo the AudioListener
z
float
✘ ✘ forward z direction fo the AudioListener
xUp
float
✘ ✘ up x direction fo the AudioListener
yUp
float
✘ ✘ up y direction fo the AudioListener
zUp
float
✘ ✘ up z direction fo the AudioListener
Return type:undefined
setPosition(x, y, z)
-
This method is DEPRECATED. It is equivalent to setting
positionX
.value
,positionY
.value
, andpositionZ
.value
directly with the givenx
,y
, andz
values, respectively.Consequently, any of the
positionX
,positionY
, andpositionZ
AudioParam
s for thisAudioListener
have an automation curve set usingsetValueCurveAtTime()
at the time this method is called, aNotSupportedError
MUST be thrown.setPosition()
sets the position of the listener in a 3D cartesian coordinate space.PannerNode
objects use this position relative to individual audio sources for spatialization.The
x
,y
, andz
parameters represent the coordinates in 3D space.The default value is (0,0,0).
Arguments for the AudioListener.setPosition() method. Parameter Type Nullable Optional Description x
float
✘ ✘ x-coordinate of the position of the AudioListener
y
float
✘ ✘ y-coordinate of the position of the AudioListener
z
float
✘ ✘ z-coordinate of the position of the AudioListener
1.11.3. Processing
Because AudioListener
's parameters can be connected with AudioNode
s and
they can also affect the output of PannerNode
s in the same graph, the node
ordering algorithm should take the AudioListener
into consideration when
computing the order of processing. For this reason, all the PannerNode
s in
the graph have the AudioListener
as input.
1.12. The AudioProcessingEvent
Interface - DEPRECATED
This is an Event
object which is dispatched to ScriptProcessorNode
nodes. It will be removed
when the ScriptProcessorNode is removed, as the replacement AudioWorkletNode
uses a different approach.
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
.
[Exposed =Window ]interface AudioProcessingEvent :Event {(
constructor DOMString ,
type AudioProcessingEventInit );
eventInitDict readonly attribute double playbackTime ;readonly attribute AudioBuffer inputBuffer ;readonly attribute AudioBuffer outputBuffer ; };
1.12.1. Attributes
inputBuffer
, of type AudioBuffer, readonly-
An AudioBuffer containing the input audio data. It will have a number of channels equal to the
numberOfInputChannels
parameter of the createScriptProcessor() method. This AudioBuffer is only valid while in the scope of theaudioprocess
event handler functions. Its values will be meaningless outside of this scope. outputBuffer
, of type AudioBuffer, readonly-
An AudioBuffer where the output audio data MUST be written. It will have a number of channels equal to the
numberOfOutputChannels
parameter of the createScriptProcessor() method. Script code within the scope of theaudioprocess
event handler functions are expected to modify theFloat32Array
arrays representing channel data in this AudioBuffer. Any script modifications to this AudioBuffer outside of this scope will not produce any audible effects. playbackTime
, of type double, readonly-
The time when the audio will be played in the same time coordinate system as the
AudioContext
'scurrentTime
.
1.12.2. AudioProcessingEventInit
dictionary AudioProcessingEventInit :EventInit {required double playbackTime ;required AudioBuffer inputBuffer ;required AudioBuffer outputBuffer ; };
1.12.2.1. Dictionary AudioProcessingEventInit
Members
inputBuffer
, of type AudioBuffer-
Value to be assigned to the
inputBuffer
attribute of the event. outputBuffer
, of type AudioBuffer-
Value to be assigned to the
outputBuffer
attribute of the event. playbackTime
, of type double-
Value to be assigned to the
playbackTime
attribute of the event.
1.13. 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"
.
Both frequency
and detune
form
a compound parameter and are both a-rate. They are used
together to determine a computedFrequency value:
computedFrequency(t) = frequency(t) * pow(2, detune(t) / 1200)
The nominal range for this compound parameter is [0, Nyquist frequency].
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | Yes | Continues to output non-silent audio with zero input. Since this is an IIR filter, the filter produces non-zero input forever, but in practice, this can be limited after some finite time where the output is sufficiently close to zero. The actual time depends on the filter coefficients. |
The number of channels of the output always equals the number of channels of the input.
enum {
BiquadFilterType "lowpass" ,"highpass" ,"bandpass" ,"lowshelf" ,"highshelf" ,"peaking" ,"notch" ,"allpass" };
Enum value | Description |
---|---|
"lowpass "
|
A lowpass
filter allows frequencies below the cutoff frequency to
pass through and attenuates frequencies above the cutoff. It
implements a standard second-order resonant lowpass filter with
12dB/octave rolloff.
|
"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. It implements a standard
second-order resonant highpass filter with 12dB/octave rolloff.
|
"bandpass "
|
A bandpass
filter allows a range of frequencies to pass through and
attenuates the frequencies below and above this frequency
range. It implements a second-order bandpass filter.
|
"lowshelf "
|
The lowshelf filter allows all frequencies through, but adds a
boost (or attenuation) to the lower frequencies. It implements
a second-order lowshelf filter.
|
"highshelf "
|
The highshelf filter is the opposite of the lowshelf filter and
allows all frequencies through, but adds a boost to the higher
frequencies. It implements a second-order highshelf filter
|
"peaking "
|
The peaking filter allows all frequencies through, but adds a
boost (or attenuation) to a range of frequencies.
|
"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.
|
"allpass "
|
An allpass filter allows all frequencies through, but changes
the phase relationship between the various frequencies. It
implements a second-order allpass filter
|
All attributes of the BiquadFilterNode
are a-rate AudioParam
s.
[Exposed =Window ]interface BiquadFilterNode :AudioNode {(
constructor BaseAudioContext ,
context optional BiquadFilterOptions = {});
options attribute BiquadFilterType type ;readonly attribute AudioParam frequency ;readonly attribute AudioParam detune ;readonly attribute AudioParam Q ;readonly attribute AudioParam gain ;undefined getFrequencyResponse (Float32Array ,
frequencyHz Float32Array ,
magResponse Float32Array ); };
phaseResponse
1.13.1. Constructors
BiquadFilterNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the BiquadFilterNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newBiquadFilterNode
will be associated with.options
BiquadFilterOptions
✘ ✔ Optional initial parameter value for this BiquadFilterNode
.
1.13.2. Attributes
Q
, of type AudioParam, readonly-
The Q factor of the filter.
For
lowpass
andhighpass
filters theQ
value is interpreted to be in dB. For these filters the nominal range is \([-Q_{lim}, Q_{lim}]\) where \(Q_{lim}\) is the largest value for which \(10^{Q/20}\) does not overflow. This is approximately \(770.63678\).For the
bandpass
,notch
,allpass
, andpeaking
filters, this value is a linear value. The value is related to the bandwidth of the filter and hence should be a positive value. The nominal range is \([0, 3.4028235e38]\), the upper limit being the most-positive-single-float.This is not used for the
lowshelf
andhighshelf
filters.Parameter Value Notes defaultValue
1 minValue
most-negative-single-float Approximately -3.4028235e38, but see above for the actual limits for different filters maxValue
most-positive-single-float Approximately 3.4028235e38, but see above for the actual limits for different filters automationRate
" a-rate
" detune
, of type AudioParam, readonly-
A detune value, in cents, for the frequency. It forms a compound parameter with
frequency
to form the computedFrequency.Parameter Value Notes defaultValue
0 minValue
\(\approx -153600\) maxValue
\(\approx 153600\) This value is approximately \(1200\ \log_2 \mathrm{FLT\_MAX}\) where FLT_MAX is the largest float
value.automationRate
" a-rate
" frequency
, of type AudioParam, readonly-
The frequency at which the
BiquadFilterNode
will operate, in Hz. It forms a compound parameter withdetune
to form the computedFrequency.Parameter Value Notes defaultValue
350 minValue
0 maxValue
Nyquist frequency automationRate
" a-rate
" gain
, of type AudioParam, readonly-
The gain of the filter. Its value is in dB units. The gain is only used for
lowshelf
,highshelf
, andpeaking
filters.Parameter Value Notes defaultValue
0 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
\(\approx 1541\) This value is approximately \(40\ \log_{10} \mathrm{FLT\_MAX}\) where FLT_MAX is the largest float
value.automationRate
" a-rate
" type
, of type BiquadFilterType-
The type of this
BiquadFilterNode
. Its default value is "lowpass
". The exact meaning of the other parameters depend on the value of thetype
attribute.
1.13.3. Methods
getFrequencyResponse(frequencyHz, magResponse, phaseResponse)
-
Given the
[[current value]]
from each of the filter parameters, synchronously calculates the frequency response for the specified frequencies. The three parameters MUST beFloat32Array
s of the same length, or anInvalidAccessError
MUST be thrown.The frequency response returned MUST be computed with the
AudioParam
sampled for the current processing block.Arguments for the BiquadFilterNode.getFrequencyResponse() method. Parameter Type Nullable Optional Description frequencyHz
Float32Array
✘ ✘ This parameter specifies an array of frequencies, in Hz, at which the response values will be calculated. magResponse
Float32Array
✘ ✘ This parameter specifies an output array receiving the linear magnitude response values. If a value in the frequencyHz
parameter is not within [0, sampleRate/2], wheresampleRate
is the value of thesampleRate
property of theAudioContext
, the corresponding value at the same index of themagResponse
array MUST beNaN
.phaseResponse
Float32Array
✘ ✘ This parameter specifies an output array receiving the phase response values in radians. If a value in the frequencyHz
parameter is not within [0; sampleRate/2], wheresampleRate
is the value of thesampleRate
property of theAudioContext
, the corresponding value at the same index of thephaseResponse
array MUST beNaN
.Return type:undefined
1.13.4. BiquadFilterOptions
This specifies the options to be used when constructing a BiquadFilterNode
. All members are optional; if
not specified, the normal default values are used to construct the
node.
dictionary BiquadFilterOptions :AudioNodeOptions {BiquadFilterType type = "lowpass";float Q = 1;float detune = 0;float frequency = 350;float gain = 0; };
1.13.4.1. Dictionary BiquadFilterOptions
Members
Q
, of type float, defaulting to1
-
The desired initial value for
Q
. detune
, of type float, defaulting to0
-
The desired initial value for
detune
. frequency
, of type float, defaulting to350
-
The desired initial value for
frequency
. gain
, of type float, defaulting to0
-
The desired initial value for
gain
. type
, of type BiquadFilterType, defaulting to"lowpass"
-
The desired initial type of the filter.
1.13.5. Filters Characteristics
There are multiple ways of implementing the type of filters
available through the BiquadFilterNode
each
having very different characteristics. The formulas in this section
describe the filters that a conforming implementation MUST
implement, as they determine the characteristics of the different
filter types. They are inspired by formulas found in the Audio EQ Cookbook.
The BiquadFilterNode
processes audio with a transfer function of
$$ H(z) = \frac{\frac{b_0}{a_0} + \frac{b_1}{a_0}z^{-1} + \frac{b_2}{a_0}z^{-2}} {1+\frac{a_1}{a_0}z^{-1}+\frac{a_2}{a_0}z^{-2}} $$
which is equivalent to a time-domain equation of:
$$ a_0 y(n) + a_1 y(n-1) + a_2 y(n-2) = b_0 x(n) + b_1 x(n-1) + b_2 x(n-2) $$
The initial filter state is 0.
Note: While fixed filters are stable, it is possible to create
unstable biquad filters using automations of AudioParam
s. It is
the developers responsibility to manage this.
Note: The UA may produce a warning to notify the user that NaN values have occurred in the filter state. This is usually indicative of an unstable filter.
The coefficients in the transfer function above are different for
each node type. The following intermediate variables are necessary for
their computation, based on the computedValue of the AudioParam
s of the BiquadFilterNode
.
-
Let \(F_s\) be the value of the
sampleRate
attribute for thisAudioContext
. -
Let \(f_0\) be the value of the computedFrequency.
-
Let \(G\) be the value of the
gain
AudioParam
. -
Let \(Q\) be the value of the
Q
AudioParam
. -
Finally let
$$ \begin{align*} A &= 10^{\frac{G}{40}} \\ \omega_0 &= 2\pi\frac{f_0}{F_s} \\ \alpha_Q &= \frac{\sin\omega_0}{2Q} \\ \alpha_{Q_{dB}} &= \frac{\sin\omega_0}{2 \cdot 10^{Q/20}} \\ S &= 1 \\ \alpha_S &= \frac{\sin\omega_0}{2}\sqrt{\left(A+\frac{1}{A}\right)\left(\frac{1}{S}-1\right)+2} \end{align*} $$
The six coefficients (\(b_0, b_1, b_2, a_0, a_1, a_2\)) for each filter type, are:
- "
lowpass
" -
$$ \begin{align*} b_0 &= \frac{1 - \cos\omega_0}{2} \\ b_1 &= 1 - \cos\omega_0 \\ b_2 &= \frac{1 - \cos\omega_0}{2} \\ a_0 &= 1 + \alpha_{Q_{dB}} \\ a_1 &= -2 \cos\omega_0 \\ a_2 &= 1 - \alpha_{Q_{dB}} \end{align*} $$
- "
highpass
" -
$$ \begin{align*} b_0 &= \frac{1 + \cos\omega_0}{2} \\ b_1 &= -(1 + \cos\omega_0) \\ b_2 &= \frac{1 + \cos\omega_0}{2} \\ a_0 &= 1 + \alpha_{Q_{dB}} \\ a_1 &= -2 \cos\omega_0 \\ a_2 &= 1 - \alpha_{Q_{dB}} \end{align*} $$
- "
bandpass
" -
$$ \begin{align*} b_0 &= \alpha_Q \\ b_1 &= 0 \\ b_2 &= -\alpha_Q \\ a_0 &= 1 + \alpha_Q \\ a_1 &= -2 \cos\omega_0 \\ a_2 &= 1 - \alpha_Q \end{align*} $$
- "
notch
" -
$$ \begin{align*} b_0 &= 1 \\ b_1 &= -2\cos\omega_0 \\ b_2 &= 1 \\ a_0 &= 1 + \alpha_Q \\ a_1 &= -2 \cos\omega_0 \\ a_2 &= 1 - \alpha_Q \end{align*} $$
- "
allpass
" -
$$ \begin{align*} b_0 &= 1 - \alpha_Q \\ b_1 &= -2\cos\omega_0 \\ b_2 &= 1 + \alpha_Q \\ a_0 &= 1 + \alpha_Q \\ a_1 &= -2 \cos\omega_0 \\ a_2 &= 1 - \alpha_Q \end{align*} $$
- "
peaking
" -
$$ \begin{align*} b_0 &= 1 + \alpha_Q\, A \\ b_1 &= -2\cos\omega_0 \\ b_2 &= 1 - \alpha_Q\,A \\ a_0 &= 1 + \frac{\alpha_Q}{A} \\ a_1 &= -2 \cos\omega_0 \\ a_2 &= 1 - \frac{\alpha_Q}{A} \end{align*} $$
- "
lowshelf
" -
$$ \begin{align*} b_0 &= A \left[ (A+1) - (A-1) \cos\omega_0 + 2 \alpha_S \sqrt{A})\right] \\ b_1 &= 2 A \left[ (A-1) - (A+1) \cos\omega_0 )\right] \\ b_2 &= A \left[ (A+1) - (A-1) \cos\omega_0 - 2 \alpha_S \sqrt{A}) \right] \\ a_0 &= (A+1) + (A-1) \cos\omega_0 + 2 \alpha_S \sqrt{A} \\ a_1 &= -2 \left[ (A-1) + (A+1) \cos\omega_0\right] \\ a_2 &= (A+1) + (A-1) \cos\omega_0 - 2 \alpha_S \sqrt{A}) \end{align*} $$
- "
highshelf
" -
$$ \begin{align*} b_0 &= A\left[ (A+1) + (A-1)\cos\omega_0 + 2\alpha_S\sqrt{A} )\right] \\ b_1 &= -2A\left[ (A-1) + (A+1)\cos\omega_0 )\right] \\ b_2 &= A\left[ (A+1) + (A-1)\cos\omega_0 - 2\alpha_S\sqrt{A} )\right] \\ a_0 &= (A+1) - (A-1)\cos\omega_0 + 2\alpha_S\sqrt{A} \\ a_1 &= 2\left[ (A-1) - (A+1)\cos\omega_0\right] \\ a_2 &= (A+1) - (A-1)\cos\omega_0 - 2\alpha_S\sqrt{A} \end{align*} $$
1.14. The ChannelMergerNode
Interface
The ChannelMergerNode
is for use in more advanced
applications and would often be used in conjunction with ChannelSplitterNode
.
Property | Value | Notes |
---|---|---|
numberOfInputs
| see notes | Defaults to 6, but is determined by ChannelMergerOptions ,numberOfInputs or the value specified by createChannelMerger .
|
numberOfOutputs
| 1 | |
channelCount
| 1 | Has channelCount constraints |
channelCountMode
| "explicit "
| Has channelCountMode constraints |
channelInterpretation
| "speakers "
| |
tail-time | No |
This interface represents an AudioNode
for
combining channels from multiple audio streams into a single audio
stream. It has a variable number of inputs (defaulting to 6), but not
all of them need be connected. There is a single output whose audio
stream has a number of channels equal to the number of inputs when any
of the inputs is actively processing. If none of the inputs are actively processing, then output is a single channel of silence.
To merge multiple inputs into one stream, each input gets downmixed into one channel (mono) based on the specified mixing rule. An unconnected input still counts as one silent channel in the output. Changing input streams does not affect the order of output channels.
ChannelMergerNode
has
two connected stereo inputs, the first and second input will be
downmixed to mono respectively before merging. The output will be a
6-channel stream whose first two channels are be filled with the
first two (downmixed) inputs and the rest of channels will be silent.
Also the ChannelMergerNode
can be used to arrange
multiple audio streams in a certain order for the multi-channel
speaker array such as 5.1 surround set up. The merger does not
interpret the channel identities (such as left, right, etc.), but
simply combines channels in the order that they are input.
[Exposed =Window ]interface ChannelMergerNode :AudioNode {constructor (BaseAudioContext ,
context optional ChannelMergerOptions = {}); };
options
1.14.1. Constructors
ChannelMergerNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the ChannelMergerNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newChannelMergerNode
will be associated with.options
ChannelMergerOptions
✘ ✔ Optional initial parameter value for this ChannelMergerNode
.
1.14.2. ChannelMergerOptions
dictionary ChannelMergerOptions :AudioNodeOptions {unsigned long numberOfInputs = 6; };
1.14.2.1. Dictionary ChannelMergerOptions
Members
numberOfInputs
, of type unsigned long, defaulting to6
-
The number inputs for the
ChannelMergerNode
. SeecreateChannelMerger()
for constraints on this value.
1.15. The ChannelSplitterNode
Interface
The ChannelSplitterNode
is for use in more advanced
applications and would often be used in conjunction with ChannelMergerNode
.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| see notes | This defaults to 6, but is otherwise determined from ChannelSplitterOptions.numberOfOutputs or the value specified by createChannelSplitter or the numberOfOutputs member of the ChannelSplitterOptions dictionary for the constructor .
|
channelCount
| numberOfOutputs
| Has channelCount constraints |
channelCountMode
| "explicit "
| Has channelCountMode constraints |
channelInterpretation
| "discrete "
| Has channelInterpretation constraints |
tail-time | No |
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 ChannelSplitterNode
then the number of active
outputs will be two (one from the left channel and one from the
right). There are always a total number of N outputs (determined by
the numberOfOutputs
parameter to the AudioContext
method createChannelSplitter()
), The
default number is 6 if this value is not provided. Any outputs which
are not "active" will output silence and would typically not be
connected to anything.
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 ChannelSplitterNode
is for doing
"matrix mixing" where individual gain control of each channel is
desired.
[Exposed =Window ]interface ChannelSplitterNode :AudioNode {constructor (BaseAudioContext ,
context optional ChannelSplitterOptions = {}); };
options
1.15.1. Constructors
ChannelSplitterNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the ChannelSplitterNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newChannelSplitterNode
will be associated with.options
ChannelSplitterOptions
✘ ✔ Optional initial parameter value for this ChannelSplitterNode
.
1.15.2. ChannelSplitterOptions
dictionary ChannelSplitterOptions :AudioNodeOptions {unsigned long numberOfOutputs = 6; };
1.15.2.1. Dictionary ChannelSplitterOptions
Members
numberOfOutputs
, of type unsigned long, defaulting to6
-
The number outputs for the
ChannelSplitterNode
. SeecreateChannelSplitter()
for constraints on this value.
1.16. The ConstantSourceNode
Interface
This interface represents a constant audio source whose output is
nominally a constant value. It is useful as a constant source node in
general and can be used as if it were a constructible AudioParam
by automating its offset
or connecting another node to it.
The single output of this node consists of one channel (mono).
Property | Value | Notes |
---|---|---|
numberOfInputs
| 0 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | No |
[Exposed =Window ]interface ConstantSourceNode :AudioScheduledSourceNode {constructor (BaseAudioContext ,
context optional ConstantSourceOptions = {});
options readonly attribute AudioParam offset ; };
1.16.1. Constructors
ConstantSourceNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the ConstantSourceNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newConstantSourceNode
will be associated with.options
ConstantSourceOptions
✘ ✔ Optional initial parameter value for this ConstantSourceNode
.
1.16.2. Attributes
offset
, of type AudioParam, readonly-
The constant value of the source.
Parameter Value Notes defaultValue
1 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
"
1.16.3. ConstantSourceOptions
This specifies options for constructing a ConstantSourceNode
. All members are optional;
if not specified, the normal defaults are used for constructing the
node.
dictionary ConstantSourceOptions {float offset = 1; };
1.16.3.1. Dictionary ConstantSourceOptions
Members
offset
, of type float, defaulting to1
-
The initial value for the
offset
AudioParam of this node.
1.17. The ConvolverNode
Interface
This interface represents a processing node which applies a linear convolution effect given an impulse response.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | Has channelCount constraints |
channelCountMode
| "clamped-max "
| Has channelCountMode constraints |
channelInterpretation
| "speakers "
| |
tail-time | Yes | Continues to output non-silent audio with zero input for the length of the buffer .
|
The input of this node is either mono (1 channel) or stereo (2 channels) and cannot be increased. Connections from nodes with more channels will be down-mixed appropriately.
There are channelCount constraints and channelCountMode constraints for this node. These constraints ensure that the input to the node is either mono or stereo.
[Exposed =Window ]interface ConvolverNode :AudioNode {constructor (BaseAudioContext ,
context optional ConvolverOptions = {});
options attribute AudioBuffer ?buffer ;attribute boolean normalize ; };
1.17.1. Constructors
ConvolverNode(context, options)
-
When the constructor is called with a
BaseAudioContext
context and an option object options, execute these steps:-
Set the attributes
normalize
to the inverse of the value ofdisableNormalization
. -
If
buffer
exists, set thebuffer
attribute to its value.Note: This means that the buffer will be normalized according to the value of the
normalize
attribute. -
Let o be new
AudioNodeOptions
dictionary. -
If
channelCount
exists in options, setchannelCount
on o with the same value. -
If
channelCountMode
exists in options, setchannelCountMode
on o with the same value. -
If
channelInterpretation
exists in options, setchannelInterpretation
on o with the same value. -
Initialize the AudioNode this, with c and o as argument.
Arguments for the ConvolverNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newConvolverNode
will be associated with.options
ConvolverOptions
✘ ✔ Optional initial parameter value for this ConvolverNode
. -
1.17.2. Attributes
buffer
, of type AudioBuffer, nullable-
At the time when this attribute is set, the
buffer
and the state of thenormalize
attribute will be used to configure theConvolverNode
with this impulse response having the given normalization. The initial value of this attribute is null.When setting thebuffer attribute
, execute the following steps synchronously:-
If the buffer
number of channels
is not 1, 2, 4, or if thesample-rate
of the buffer is not the same as thesample-rate
of its associatedBaseAudioContext
, aNotSupportedError
MUST be thrown. -
Acquire the content of the
AudioBuffer
.
Note: If the
buffer
is set to an new buffer, audio may glitch. If this is undesirable, it is recommended to create a newConvolverNode
to replace the old, possibly cross-fading between the two.Note: The
ConvolverNode
produces a mono output only in the single case where there is a single input channel and a single-channelbuffer
. In all other cases, the output is stereo. In particular, when thebuffer
has four channels and there are two input channels, theConvolverNode
performs matrix "true" stereo convolution. For normative information please see the channel configuration diagrams -
normalize
, of type boolean-
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 istrue
in order to achieve a more uniform output level from the convolver when loaded with diverse impulse responses. Ifnormalize
is set tofalse
, 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 thebuffer
attribute is set.If the
normalize
attribute is false when thebuffer
attribute is set then theConvolverNode
will perform a linear convolution given the exact impulse response contained within thebuffer
.Otherwise, if the
normalize
attribute is true when thebuffer
attribute is set then theConvolverNode
will first perform a scaled RMS-power analysis of the audio data contained withinbuffer
to calculate a normalizationScale given this algorithm:function calculateNormalizationScale( buffer) { const GainCalibration= 0.00125 ; const GainCalibrationSampleRate= 44100 ; const MinPower= 0.000125 ; // Normalize by RMS power. const numberOfChannels= buffer. numberOfChannels; const length= buffer. length; let power= 0 ; for ( let i= 0 ; i< numberOfChannels; i++ ) { let channelPower= 0 ; const channelData= buffer. getChannelData( i); for ( let j= 0 ; j< length; j++ ) { const sample= channelData[ j]; channelPower+= sample* sample; } power+= channelPower; } power= Math. sqrt( power/ ( numberOfChannels* length)); // Protect against accidental overload. if ( ! isFinite( power) || isNaN( power) || power< MinPower) power= MinPower; let 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 ( 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.
1.17.3. ConvolverOptions
The specifies options for constructing a ConvolverNode
. All members are optional; if not
specified, the node is contructing using the normal defaults.
dictionary ConvolverOptions :AudioNodeOptions {AudioBuffer ?buffer ;boolean disableNormalization =false ; };
1.17.3.1. Dictionary ConvolverOptions
Members
buffer
, of type AudioBuffer, nullable-
The desired buffer for the
ConvolverNode
. This buffer will be normalized according to the value ofdisableNormalization
. disableNormalization
, of type boolean, defaulting tofalse
-
The opposite of the desired initial value for the
normalize
attribute of theConvolverNode
.
1.17.4. Channel Configurations for Input, Impulse Response and Output
Implementations MUST support the following allowable configurations
of impulse response channels in a ConvolverNode
to achieve various reverb effects with 1 or 2 channels of input.
As shown in the diagram below, single channel convolution operates on a mono audio input, using a
mono impulse response, and generating a mono output. The remaining
images in the diagram illustrate the supported cases for mono and
stereo playback where the number of channels of the input is 1 or 2, and the number of channels in the buffer
is 1, 2, or 4.
Developers desiring more complex and arbitrary matrixing can use a ChannelSplitterNode
, multiple single-channel ConvolverNode
s and a ChannelMergerNode
.
If this node is not actively processing, the output is a single channel of silence.
Note: The diagrams below show the outputs when actively processing.
1.18. 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.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | Yes | Continues to output non-silent audio with zero input up to the maxDelayTime of the node.
|
The number of channels of the output always equals the number of channels of the input.
It delays the incoming audio signal by a certain amount.
Specifically, at each time t, input signal input(t), delay time delayTime(t) and output signal output(t), the output will be output(t) = input(t -
delayTime(t)). The default delayTime
is 0 seconds
(no delay).
When the number of channels in a DelayNode
's input changes
(thus changing the output channel count also), there may be delayed
audio samples which have not yet been output by the node and are part
of its internal state. If these samples were received earlier with a
different channel count, they MUST be upmixed or downmixed before
being combined with newly received input so that all internal
delay-line mixing takes place using the single prevailing channel
layout.
Note: By definition, a DelayNode
introduces an audio processing
latency equal to the amount of the delay.
[Exposed =Window ]interface DelayNode :AudioNode {constructor (BaseAudioContext ,
context optional DelayOptions = {});
options readonly attribute AudioParam delayTime ; };
1.18.1. Constructors
DelayNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the DelayNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newDelayNode
will be associated with.options
DelayOptions
✘ ✔ Optional initial parameter value for this DelayNode
.
1.18.2. Attributes
delayTime
, of type AudioParam, readonly-
An
AudioParam
object representing the amount of delay (in seconds) to apply. Its defaultvalue
is 0 (no delay). The minimum value is 0 and the maximum value is determined by themaxDelayTime
argument to theAudioContext
methodcreateDelay()
or themaxDelayTime
member of theDelayOptions
dictionary for theconstructor
.If
DelayNode
is part of a cycle, then the value of thedelayTime
attribute is clamped to a minimum of one render quantum.Parameter Value Notes defaultValue
0 minValue
0 maxValue
maxDelayTime
automationRate
" a-rate
"
1.18.3. DelayOptions
This specifies options for constructing a DelayNode
. All members are optional; if not
given, the node is constructed using the normal defaults.
dictionary DelayOptions :AudioNodeOptions {double maxDelayTime = 1;double delayTime = 0; };
1.18.3.1. Dictionary DelayOptions
Members
delayTime
, of type double, defaulting to0
-
The initial delay time for the node.
maxDelayTime
, of type double, defaulting to1
-
The maximum delay time for the node. See
createDelay(maxDelayTime)
for constraints.
1.18.4. Processing
A DelayNode
has an internal buffer that holds delayTime
seconds of audio.
The processing of a DelayNode
is broken down in two parts: writing to the
delay line, and reading from the delay line. This is done via two internal AudioNode
s (that are not available to authors and exist only to ease
the description of the inner workings of the node). Both are created from a DelayNode
.
Creating a DelayWriter for a DelayNode
means creating an object
that has the same interface as an AudioNode
, and that writes the input audio
into the internal buffer of the DelayNode
. It has the same input connections
as the DelayNode
it was created from.
Creating a DelayReader for a DelayNode
means creating an object
that has the same interface as an AudioNode
, and that can read the audio
data from the internal buffer of the DelayNode
. It is connected to the same AudioNode
s as the DelayNode
it was created from. A DelayReader is
a source node.
When processing an input buffer, a DelayWriter MUST write the audio to
the internal buffer of the DelayNode
.
When producing an output buffer, a DelayReader MUST yield exactly the
audio that was written to the corresponding DelayWriter delayTime
seconds ago.
Note: This means that channel count changes are reflected after the delay time has passed.
1.19. 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.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | Has channelCount constraints |
channelCountMode
| "clamped-max "
| Has channelCountMode constraints |
channelInterpretation
| "speakers "
| |
tail-time | Yes | This node has a tail-time such that this node continues to output non-silent audio with zero input due to the look-ahead delay. |
[Exposed =Window ]interface DynamicsCompressorNode :AudioNode {constructor (BaseAudioContext ,
context optional DynamicsCompressorOptions = {});
options readonly attribute AudioParam threshold ;readonly attribute AudioParam knee ;readonly attribute AudioParam ratio ;readonly attribute float reduction ;readonly attribute AudioParam attack ;readonly attribute AudioParam release ; };
1.19.1. Constructors
DynamicsCompressorNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Let
[[internal reduction]]
be a private slot on this, that holds a floating point number, in decibels. Set[[internal reduction]]
to 0.0.Arguments for the DynamicsCompressorNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newDynamicsCompressorNode
will be associated with.options
DynamicsCompressorOptions
✘ ✔ Optional initial parameter value for this DynamicsCompressorNode
.
1.19.2. Attributes
attack
, of type AudioParam, readonly-
The amount of time (in seconds) to reduce the gain by 10dB.
Parameter Value Notes defaultValue
.003 minValue
0 maxValue
1 automationRate
" k-rate
"Has automation rate constraints knee
, of type AudioParam, readonly-
A decibel value representing the range above the threshold where the curve smoothly transitions to the "ratio" portion.
Parameter Value Notes defaultValue
30 minValue
0 maxValue
40 automationRate
" k-rate
"Has automation rate constraints ratio
, of type AudioParam, readonly-
The amount of dB change in input for a 1 dB change in output.
Parameter Value Notes defaultValue
12 minValue
1 maxValue
20 automationRate
" k-rate
"Has automation rate constraints reduction
, of type float, readonly-
A read-only decibel value for metering purposes, representing the current amount of gain reduction that the compressor is applying to the signal. If fed no signal the value will be 0 (no gain reduction). When this attribute is read, return the value of the private slot
[[internal reduction]]
. release
, of type AudioParam, readonly-
The amount of time (in seconds) to increase the gain by 10dB.
Parameter Value Notes defaultValue
.25 minValue
0 maxValue
1 automationRate
" k-rate
"Has automation rate constraints threshold
, of type AudioParam, readonly-
The decibel value above which the compression will start taking effect.
Parameter Value Notes defaultValue
-24 minValue
-100 maxValue
0 automationRate
" k-rate
"Has automation rate constraints
1.19.3. DynamicsCompressorOptions
This specifies the options to use in constructing a DynamicsCompressorNode
. All members are
optional; if not specified the normal defaults are used in
constructing the node.
dictionary DynamicsCompressorOptions :AudioNodeOptions {float attack = 0.003;float knee = 30;float ratio = 12;float release = 0.25;float threshold = -24; };
1.19.3.1. Dictionary DynamicsCompressorOptions
Members
attack
, of type float, defaulting to0.003
-
The initial value for the
attack
AudioParam. knee
, of type float, defaulting to30
-
The initial value for the
knee
AudioParam. ratio
, of type float, defaulting to12
-
The initial value for the
ratio
AudioParam. release
, of type float, defaulting to0.25
-
The initial value for the
release
AudioParam. threshold
, of type float, defaulting to-24
-
The initial value for the
threshold
AudioParam.
1.19.4. Processing
Dynamics compression can be implemented in a variety of ways. The DynamicsCompressorNode
implements a dynamics processor that
has the following characteristics:
-
Fixed look-ahead (this means that an
DynamicsCompressorNode
adds a fixed latency to the signal chain). -
Configurable attack speed, release speed, threshold, knee hardness and ratio.
-
Side-chaining is not supported.
-
The gain reduction is reported via the
reduction
property on theDynamicsCompressorNode
. -
The compression curve has three parts:
-
The first part is the identity: \(f(x) = x\).
-
The second part is the soft-knee portion, which MUST be a monotonically increasing function.
-
The third part is a linear function: \(f(x) = \frac{1}{ratio} \cdot x \).
This curve MUST be continuous and piece-wise differentiable, and corresponds to a target output level, based on the input level.
-
Graphically, such a curve would look something like this:
Internally, the DynamicsCompressorNode
is described with a
combination of other AudioNode
s, as well as a special
algorithm, to compute the gain reduction value.
The following AudioNode
graph is used internally, input
and output
respectively being the
input and output AudioNode
, context
the BaseAudioContext
for this DynamicsCompressorNode
, and
a new class, EnvelopeFollower, that instantiates a
special object that behaves like an AudioNode
, described
below:
const delay = new DelayNode(context, {delayTime: 0.006}); const gain = new GainNode(context); const compression = new EnvelopeFollower(); input.connect(delay).connect(gain).connect(output); input.connect(compression).connect(gain.gain);
Note: This implements the pre-delay and the application of the reduction gain.
The following algorithm describes the processing performed by an EnvelopeFollower object, to be applied to the input signal to produce the gain reduction value. An EnvelopeFollower has two slots holding floating point values. Those values persist accros invocation of this algorithm.
-
Let
[[detector average]]
be a floating point number, initialized to 0.0. -
Let
[[compressor gain]]
be a floating point number, initialized to 1.0.
-
Let attack and release have the values of
attack
andrelease
, respectively, sampled at the time of processing (those are k-rate parameters), mutiplied by the sample-rate of theBaseAudioContext
thisDynamicsCompressorNode
is associated with. -
Let detector average be the value of the slot
[[detector average]]
. -
Let compressor gain be the value of the slot
[[compressor gain]]
. -
For each sample input of the render quantum to be processed, execute the following steps:
-
If the absolute value of input is less than 0.0001, let attenuation be 1.0. Else, let shaped input be the value of applying the compression curve to the absolute value of input. Let attenuation be shaped input divided by the absolute value of input.
-
Let releasing be
true
if attenuation is greater than compressor gain,false
otherwise. -
Let detector rate be the result of applying the detector curve to attenuation.
-
Subtract detector average from attenuation, and multiply the result by detector rate. Add this new result to detector average.
-
Clamp detector average to a maximum of 1.0.
-
Let envelope rate be the result of computing the envelope rate based on values of attack and release.
-
If releasing is
true
, set compressor gain to be the product of compressor gain and envelope rate, clamped to a maximum of 1.0. -
Else, if releasing is
false
, let gain increment to be detector average minus compressor gain. Multiply gain increment by envelope rate, and add the result to compressor gain. -
Compute reduction gain to be compressor gain multiplied by the return value of computing the makeup gain.
-
Compute metering gain to be reduction gain, converted to decibel.
-
-
Set
[[compressor gain]]
to compressor gain. -
Set
[[detector average]]
to detector average. -
Atomically set the internal slot
[[internal reduction]]
to the value of metering gain.Note: This step makes the metering gain update once per block, at the end of the block processing.
The makeup gain is a fixed gain stage that only depends on ratio, knee and threshold parameter of the compressor, and not on the input signal. The intent here is to increase the output level of the compressor so it is comparable to the input level.
-
Let full range gain be the value returned by applying the compression curve to the value 1.0.
-
Let full range makeup gain be the inverse of full range gain.
-
Return the result of taking the 0.6 power of full range makeup gain.
-
The envelope rate MUST be the calculated from the ratio of the compressor gain and the detector average.
Note: When attacking, this number less than or equal to 1, when releasing, this number is strictly greater than 1.
-
The attack curve MUST be a continuous, monotonically increasing function in the range \([0, 1]\). The shape of this curve MAY be controlled by
attack
. -
The release curve MUST be a continuous, monotonically decreasing function that is always greater than 1. The shape of this curve MAY be controlled by
release
.
This operation returns the value computed by applying this function to the ratio of compressor gain and detector average.
Applying the detector curve to the change rate when attacking or releasing allow implementing adaptive release. It is a function that MUST respect the following constraints:
-
The output of the function MUST be in \([0,1]\).
-
The function MUST be monotonically increasing, continuous.
Note: It is allowed, for example, to have a compressor that performs an adaptive release, that is, releasing faster the harder the compression, or to have curves for attack and release that are not of the same shape.
-
Let threshold and knee have the values of
threshold
andknee
, respectively, converted to linear units and sampled at the time of processing of this block (as k-rate parameters). -
Calculate the sum of
threshold
plusknee
also sampled at the time of processing of this block (as k-rate parameters). -
Let knee end threshold have the value of this sum converted to linear units.
-
Let ratio have the value of the
ratio
, sampled at the time of processing of this block (as a k-rate parameter). -
This function is the identity up to the value of the linear threshold (i.e., \(f(x) = x\)).
-
From the threshold up to the knee end threshold, User-Agents can choose the curve shape. The whole function MUST be monotonically increasing and continuous.
Note: If the knee is 0, the
DynamicsCompressorNode
is called a hard-knee compressor. -
This function is linear, based on the ratio, after the threshold and the soft knee (i.e., \(f(x) = \frac{1}{ratio} \cdot x \)).
-
If \(v\) is equal to zero, return -1000.
-
Else, return \( 20 \, \log_{10}{v} \).
1.20. The GainNode
Interface
Changing the gain of an audio signal is a fundamental operation in
audio applications. This
interface is an AudioNode
with a single input and
single output:
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | No |
Each sample of each channel of the input data of the GainNode
MUST be multiplied by the computedValue of the gain
AudioParam
.
[Exposed =Window ]interface GainNode :AudioNode {constructor (BaseAudioContext ,
context optional GainOptions = {});
options readonly attribute AudioParam gain ; };
1.20.1. Constructors
GainNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the GainNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newGainNode
will be associated with.options
GainOptions
✘ ✔ Optional initial parameter values for this GainNode
.
1.20.2. Attributes
gain
, of type AudioParam, readonly-
Represents the amount of gain to apply.
Parameter Value Notes defaultValue
1 minValue
most-negative-single-float Approximately -3.4028235e38 maxValue
most-positive-single-float Approximately 3.4028235e38 automationRate
" a-rate
"
1.20.3. GainOptions
This specifies options to use in constructing a GainNode
. All members are optional; if not
specified, the normal defaults are used in constructing the node.
dictionary GainOptions :AudioNodeOptions {float gain = 1.0; };
1.20.3.1. Dictionary GainOptions
Members
gain
, of type float, defaulting to1.0
-
The initial gain value for the
gain
AudioParam.
1.21. The IIRFilterNode
Interface
IIRFilterNode
is an AudioNode
processor implementing a general IIR Filter. In general, it is best
to use BiquadFilterNode
's to implement
higher-order filters for the following reasons:
-
Generally less sensitive to numeric issues
-
Filter parameters can be automated
-
Can be used to create all even-ordered IIR filters
However, odd-ordered filters cannot be created, so if such filters are needed or automation is not needed, then IIR filters may be appropriate.
Once created, the coefficients of the IIR filter cannot be changed.
Property | Value | Notes |
---|---|---|
numberOfInputs
| 1 | |
numberOfOutputs
| 1 | |
channelCount
| 2 | |
channelCountMode
| "max "
| |
channelInterpretation
| "speakers "
| |
tail-time | Yes | Continues to output non-silent audio with zero input. Since this is an IIR filter, the filter produces non-zero input forever, but in practice, this can be limited after some finite time where the output is sufficiently close to zero. The actual time depends on the filter coefficients. |
The number of channels of the output always equals the number of channels of the input.
[Exposed =Window ]interface IIRFilterNode :AudioNode {constructor (BaseAudioContext ,
context IIRFilterOptions );
options undefined getFrequencyResponse (Float32Array ,
frequencyHz Float32Array ,
magResponse Float32Array ); };
phaseResponse
1.21.1. Constructors
IIRFilterNode(context, options)
-
When the constructor is called with a
BaseAudioContext
c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.Arguments for the IIRFilterNode.constructor() method. Parameter Type Nullable Optional Description context
BaseAudioContext
✘ ✘ The BaseAudioContext
this newIIRFilterNode
will be associated with.options
IIRFilterOptions
✘ ✘ Initial parameter value for this IIRFilterNode
.
1.21.2. Methods
getFrequencyResponse(frequencyHz, magResponse, phaseResponse)
-
Given the current filter parameter settings, synchronously calculates the frequency response for the specified frequencies. The three parameters MUST be
Float32Array
s of the same length, or anInvalidAccessError
MUST be thrown.Arguments for the IIRFilterNode.getFrequencyResponse() method. Parameter Type Nullable Optional Description frequencyHz
Float32Array
✘ ✘ This parameter specifies an array of frequencies, in Hz, at which the response values will be calculated.