Table of Contents

Class AudioContext

Namespace
CSharpToJavaScript.APIs.JS
Assembly
CSharpToJavaScript.dll

The AudioContext interface represents an audio-processing graph built from audio modules linked together, each represented by an AudioNode.

[Value("AudioContext")]
public class AudioContext : BaseAudioContext
Inheritance
AudioContext
Inherited Members

Remarks

An audio context controls both the creation of the nodes it contains and the execution of the audio processing, or decoding. You need to create an AudioContext before you do anything else, as everything happens inside a context. It's recommended to create one AudioContext and reuse it instead of initializing a new one each time, and it's OK to use a single AudioContext for several different audio sources and pipeline concurrently.

-Using the Web Audio API
-OfflineAudioContext

See also on MDN

Constructors

AudioContext() 

public AudioContext()

AudioContext(AudioContextOptions)

The AudioContext() constructor
creates a new AudioContext object which represents an audio-processing
graph, built from audio modules linked together, each represented by an
AudioNode.

public AudioContext(AudioContextOptions contextOptions = null)

Parameters

contextOptions AudioContextOptions

Remarks

-OfflineAudioContext(OfflineAudioContextOptions) constructor

See also on MDN

Properties

BaseLatency

The baseLatency read-only property of the
AudioContext interface returns a double that represents the number of
seconds of processing latency incurred by the AudioContext passing an audio
buffer from the AudioDestinationNode — i.e., the end of the audio graph —
into the host system's audio subsystem ready for playing.

[Value("baseLatency")]
public Number BaseLatency { get; }

Property Value

Number

A double representing the base latency in seconds.

Remarks

NOTE

You can request a certain latency during
{{domxref("AudioContext.AudioContext()", "construction time", "", "true")}} with the
latencyHint option, but the browser may ignore the option.

-Using the Web Audio API
-Web Audio API

See also on MDN

Onerror 

[Value("onerror")]
public EventHandlerNonNull Onerror { get; set; }

Property Value

EventHandlerNonNull

Onsinkchange 

[Value("onsinkchange")]
public EventHandlerNonNull Onsinkchange { get; set; }

Property Value

EventHandlerNonNull

OutputLatency

The outputLatency read-only property of
the AudioContext Interface provides an estimation of the output latency
of the current audio context.

[Value("outputLatency")]
public Number OutputLatency { get; }

Property Value

Number

A double representing the output latency in seconds.

Remarks

This is the time, in seconds, between the browser passing an audio buffer out of an
audio graph over to the host system's audio subsystem to play, and the time at which the
first sample in the buffer is actually processed by the audio output device.

It varies depending on the platform and the available hardware.

-Using the Web Audio API
-Web Audio API

See also on MDN

RenderCapacity 

[Value("renderCapacity")]
public AudioRenderCapacity RenderCapacity { get; }

Property Value

AudioRenderCapacity

SinkId

NOTE
Experimental
The sinkId read-only property of the
AudioContext interface returns the sink ID of the current output audio device. 
[Value("sinkId")]
public Union180 SinkId { get; }

Property Value

Union180

This property returns one of the following values, depending on how the sink ID was set:

Remarks

-Change the destination output device in Web Audio
-SetSinkId(Union181)
-AudioContextsinkchange

See also on MDN

Methods

Close()

The close() method of the AudioContext Interface closes the audio context, releasing any system audio resources that it uses.

[Value("close")]
public Task<GlobalObject.Undefined> Close()

Returns

Task<GlobalObject.Undefined>

A Promise that resolves with 'undefined'.

Remarks

This function does not automatically release all AudioContext-created objects, unless other references have been released as well; however, it will forcibly release any system audio resources that might prevent additional AudioContexts from being created and used, suspend the progression of audio time in the audio context, and stop processing audio data. The returned Promise resolves when all AudioContext-creation-blocking resources have been released. This method throws an INVALID_STATE_ERR exception if called on an OfflineAudioContext.

-Using the Web Audio API
-Web Audio API

See also on MDN

CreateMediaElementSource(HTMLMediaElement)

The createMediaElementSource() method of the AudioContext Interface is used to create a new MediaElementAudioSourceNode object, given an existing HTML {{htmlelement("audio")}} or {{htmlelement("video")}} element, the audio from which can then be played and manipulated.

[Value("createMediaElementSource")]
public MediaElementAudioSourceNode CreateMediaElementSource(HTMLMediaElement mediaElement)

Parameters

mediaElement HTMLMediaElement

Returns

MediaElementAudioSourceNode

A MediaElementAudioSourceNode.

Remarks

For more details about media element audio source nodes, check out the MediaElementAudioSourceNode reference page.

-Using the Web Audio API

See also on MDN

CreateMediaStreamDestination()

The createMediaStreamDestination() method of the AudioContext Interface is used to create a new MediaStreamAudioDestinationNode object associated with a WebRTC MediaStream representing an audio stream, which may be stored in a local file or sent to another computer.

[Value("createMediaStreamDestination")]
public MediaStreamAudioDestinationNode CreateMediaStreamDestination()

Returns

MediaStreamAudioDestinationNode

A MediaStreamAudioDestinationNode.

Remarks

The MediaStream is created when the node is created and is accessible via the MediaStreamAudioDestinationNode's stream attribute. This stream can be used in a similar way as a MediaStream obtained via Navigator.GetUserMedia — it can, for example, be sent to a remote peer using the addStream() method of RTCPeerConnection.

For more details about media stream destination nodes, check out the MediaStreamAudioDestinationNode reference page.

-Using the Web Audio API

See also on MDN

CreateMediaStreamSource(MediaStream)

The createMediaStreamSource() method of the AudioContext
Interface is used to create a new MediaStreamAudioSourceNode
object, given a media stream (say, from a GetUserMedia(MediaStreamConstraints)
instance), the audio from which can then be played and manipulated.

[Value("createMediaStreamSource")]
public MediaStreamAudioSourceNode CreateMediaStreamSource(MediaStream mediaStream)

Parameters

mediaStream MediaStream

Returns

MediaStreamAudioSourceNode

A new MediaStreamAudioSourceNode object representing the audio node
whose media is obtained from the specified source stream.

Remarks

For more details about media stream audio source nodes, check out the MediaStreamAudioSourceNode reference page.

-Using the Web Audio API

See also on MDN

CreateMediaStreamTrackSource(MediaStreamTrack)

The createMediaStreamTrackSource() method of the AudioContext interface creates and returns a MediaStreamTrackAudioSourceNode which represents an audio source whose data comes from the specified MediaStreamTrack.

[Value("createMediaStreamTrackSource")]
public MediaStreamTrackAudioSourceNode CreateMediaStreamTrackSource(MediaStreamTrack mediaStreamTrack)

Parameters

mediaStreamTrack MediaStreamTrack

Returns

MediaStreamTrackAudioSourceNode

A MediaStreamTrackAudioSourceNode object which acts as a source for
audio data found in the specified audio track.

Remarks

This differs from CreateMediaStreamSource(MediaStream), which creates a MediaStreamAudioSourceNode whose audio comes from the audio track in a specified MediaStream whose Id is first, lexicographically (alphabetically).

-Web Audio API
-Using the Web Audio API
-MediaStreamTrackAudioSourceNode

See also on MDN

GetOutputTimestamp()

The
getOutputTimestamp() method of the
AudioContext interface returns a new AudioTimestamp object
containing two audio timestamp values relating to the current audio context.

[Value("getOutputTimestamp")]
public AudioTimestamp GetOutputTimestamp()

Returns

AudioTimestamp

An AudioTimestamp object, which has the following properties.

Remarks

The two values are as follows:

See also on MDN

Resume()

The resume() method of the AudioContext
interface resumes the progression of time in an audio context that has previously been
suspended.

[Value("resume")]
public Task<GlobalObject.Undefined> Resume()

Returns

Task<GlobalObject.Undefined>

A Promise that resolves when the context has resumed. The promise is
rejected if the context has already been closed.

Remarks

This method will cause an INVALID_STATE_ERR exception to be thrown if
called on an OfflineAudioContext.

-Using the Web Audio API
-Web Audio API

See also on MDN

SetSinkId(Union181)

NOTE
Experimental
The setSinkId() method of the AudioContext interface sets the output audio device for the AudioContext. If a sink ID is not explicitly set, the default system audio output device will be used. 
[Value("setSinkId")]
public Task<GlobalObject.Undefined> SetSinkId(Union181 sinkId)

Parameters

sinkId Union181

Returns

Task<GlobalObject.Undefined>

A Promise that fulfills with a value of undefined.Attempting to set the sink ID to its existing value (i.e., returned by SinkId), throws no errors, but it aborts the process immediately.

Remarks

To set the audio device to a device different than the default one, the developer needs permission to access to audio devices. If required, the user can be prompted to grant the required permission via a GetUserMedia(MediaStreamConstraints) call.

In addition, this feature may be blocked by a speaker-selection Permissions Policy.

-Change the destination output device in Web Audio
-SinkId
-AudioContextsinkchange

See also on MDN

Suspend()

The suspend() method of the AudioContext Interface suspends the progression of time in the audio context, temporarily halting audio hardware access and reducing CPU/battery usage in the process — this is useful if you want an application to power down the audio hardware when it will not be using an audio context for a while.

[Value("suspend")]
public Task<GlobalObject.Undefined> Suspend()

Returns

Task<GlobalObject.Undefined>

A Promise that resolves with 'undefined'. The promise is rejected if the context has already been closed.

Remarks

This method will cause an INVALID_STATE_ERR exception to be thrown if called on an OfflineAudioContext.

-Using the Web Audio API
-Web Audio API

See also on MDN

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