Constructor Detail

• RemoteAudioTrack

  RemoteAudioTrack(String name, AudioTrack rtcTrack, RtpReceiver receiver, RTCThreadToken rtcThreadToken)

Method Detail

• getRtcTrack

   AudioTrack getRtcTrack()

  The underlying WebRTC audio track.

• getEvents

   final EventListenable<TrackEvent> getEvents()

• getName

   final String getName()

• setName

   final Unit setName(String name)

• getKind

   final Track.Kind getKind()

• setKind

   final Unit setKind(Track.Kind kind)

• getSid

   final String getSid()

• setSid

   final Unit setSid(String sid)

• getStreamState

   final Track.StreamState getStreamState()

• setStreamState

   final Unit setStreamState(Track.StreamState streamState)

• getEnabled

   final Boolean getEnabled()

• setEnabled

   final Unit setEnabled(Boolean enabled)

• getStatsGetter

   final Function1<RTCStatsCollectorCallback, Unit> getStatsGetter()

• setStatsGetter

   final Unit setStatsGetter(Function1<RTCStatsCollectorCallback, Unit> statsGetter)

• getRtcThreadToken

   final RTCThreadToken getRtcThreadToken()

• addSink

   Unit addSink(AudioTrackSink sink)

  Adds a sink that receives the audio bytes and related information for this audio track. Repeated calls using the same sink will only add the sink once.

  Implementations should copy the audio data into a local copy if they wish to use the data after the AudioTrackSink.onData callback returns. Long running processing of the received audio data should be done in a separate thread, as doing so inline may block the audio thread.

• removeSink

   Unit removeSink(AudioTrackSink sink)

  Removes a previously added sink.

• setVolume

   final Unit setVolume(Double volume)

  Sets the volume.

  Parameters:

  volume - a gain value in the range 0 to 10.