Class LocalAudioTrack

Audio track sending to the remote peer audio frames originating from a local track source (local microphone or other audio recording device).

Inheritance
Object
MediaTrack
LocalMediaTrack
LocalAudioTrack
Implements
IDisposable
Inherited Members
Namespace: Microsoft.MixedReality.WebRTC
Assembly: Microsoft.MixedReality.WebRTC.dll
Syntax
public class LocalAudioTrack : LocalMediaTrack, IDisposable, IAudioSource

Properties

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Enabled

Enabled status of the track. If enabled, send local audio frames to the remote peer as expected. If disabled, send only black frames instead.

Declaration
public bool Enabled { get; set; }
Property Value
Type Description
Boolean
Remarks

Reading the value of this property after the track has been disposed is valid, and returns false. Writing to this property after the track has been disposed throws an exception.

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Source

Audio track source this track is pulling its audio frames from.

Declaration
public AudioTrackSource Source { get; }
Property Value
Type Description
AudioTrackSource

Methods

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CreateFromSource(AudioTrackSource, LocalAudioTrackInitConfig)

Create an audio track from an existing audio track source.

This does not add the track to any peer connection. Instead, the track must be added manually to an audio transceiver to be attached to a peer connection and transmitted to a remote peer.

Declaration
public static LocalAudioTrack CreateFromSource(AudioTrackSource source, LocalAudioTrackInitConfig initConfig)
Parameters
Type Name Description
AudioTrackSource source

The track source which provides the raw audio frames to the newly created track.
LocalAudioTrackInitConfig initConfig

Configuration to initialize the track being created.
Returns
Type Description
LocalAudioTrack

Asynchronous task completed once the track is created.
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CreateReadBuffer()

Starts buffering the audio frames from in an AudioTrackReadBuffer.

Declaration
public AudioTrackReadBuffer CreateReadBuffer()
Returns
Type Description
AudioTrackReadBuffer
Remarks

WebRTC audio tracks produce an audio frame every 10 ms. If you want the audio frames to be buffered (and optionally resampled) automatically, and you want the application to control when new audio data is read, create an AudioTrackReadBuffer using CreateReadBuffer(). If you want to process the audio frames as soon as they are received, without conversions, subscribe to AudioFrameReady instead.

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Dispose()

Remove the track from the associated Transceiver (if there is one) and release the corresponding resources.

Declaration
public override void Dispose()
Overrides
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ToString()

Declaration
public override string ToString()
Returns
Type Description
String
Overrides

Events

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AudioFrameReady

Event that occurs when a new audio frame is available from the source, either because the source produced it locally (AudioTrackSource, LocalAudioTrack) or because it received it from the remote peer (RemoteAudioTrack).

Declaration
public event AudioFrameDelegate AudioFrameReady
Event Type
Type Description
AudioFrameDelegate
Remarks

WebRTC audio tracks produce an audio frame every 10 ms. If you want to process the audio frames as soon as they are received, without conversions, subscribe to AudioFrameReady. If you want the audio frames to be buffered (and optionally resampled) automatically, and you want the application to control when new audio data is read, create an AudioTrackReadBuffer using CreateReadBuffer().

Implements

IDisposable