akasha.rtc

Interface RTCRtpSynchronizationSource

All Superinterfaces:

RTCRtpContributingSource

@Generated(value="org.realityforge.webtack")
 @JsType(isNative=true,
        namespace="<global>",
        name="Object")
public interface RTCRtpSynchronizationSource
extends RTCRtpContributingSource

The RTCRtpSynchronizationSource dictionary of the WebRTC API is used by getSynchronizationSources() to describe a particular synchronization source (SSRC).

See Also:

RTCRtpSynchronizationSource - MDN, RTCRtpSynchronizationSource - WebRTC 1.0: Real-time Communication Between Browsers

Method Summary

Modifier and Type	Method and Description
default RTCRtpSynchronizationSource	audioLevel(double audioLevel) The read-only audioLevel property of the RTCRtpContributingSource interface indicates the audio level contained in the last RTP packet played from the described source.
static RTCRtpSynchronizationSource	create(int rtpTimestamp, int source, double timestamp)
default RTCRtpSynchronizationSource	rtpTimestamp(int rtpTimestamp) The read-only rtpTimestamp property of the RTCRtpContributingSource interface returns a DOMHighResTimeStamp indicating the source-generated time at which the media contained int he packet was first sampled or obtained.
default RTCRtpSynchronizationSource	source(int source) The read-only source property of the RTCRtpContributingSource interface returns the source identifier of a particular stream of RTP packets.
default RTCRtpSynchronizationSource	timestamp(double timestamp) The read-only timestamp property of the RTCRtpContributingSource interface returns a DOMHighResTimeStamp indicating the most recent time of playout of an RTP packet from the source.

Methods inherited from interface akasha.rtc.RTCRtpContributingSource

audioLevel, rtpTimestamp, setAudioLevel, setRtpTimestamp, setSource, setTimestamp, source, timestamp

Method Detail

create

@JsOverlay
 @Nonnull
static RTCRtpSynchronizationSource create(int rtpTimestamp,
                                                               int source,
                                                               double timestamp)

Specified by:

create in interface RTCRtpContributingSource

audioLevel

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource audioLevel(double audioLevel)

The read-only audioLevel property of the RTCRtpContributingSource interface indicates the audio level contained in the last RTP packet played from the described source.

Specified by:

audioLevel in interface RTCRtpContributingSource

See Also:

RTCRtpContributingSource.audioLevel - MDN, audioLevel - WebRTC 1.0: Real-time Communication Between Browsers

rtpTimestamp

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource rtpTimestamp(int rtpTimestamp)

The read-only rtpTimestamp property of the RTCRtpContributingSource interface returns a DOMHighResTimeStamp indicating the source-generated time at which the media contained int he packet was first sampled or obtained.

Specified by:

rtpTimestamp in interface RTCRtpContributingSource

See Also:

RTCRtpContributingSource.rtpTimestamp - MDN, rtpTimestamp - WebRTC 1.0: Real-time Communication Between Browsers

source

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource source(int source)

The read-only source property of the RTCRtpContributingSource interface returns the source identifier of a particular stream of RTP packets.

Specified by:

source in interface RTCRtpContributingSource

See Also:

RTCRtpContributingSource.source - MDN, source - WebRTC 1.0: Real-time Communication Between Browsers

timestamp

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource timestamp(double timestamp)

The read-only timestamp property of the RTCRtpContributingSource interface returns a DOMHighResTimeStamp indicating the most recent time of playout of an RTP packet from the source.

Specified by:

timestamp in interface RTCRtpContributingSource

See Also:

RTCRtpContributingSource.timestamp - MDN, timestamp - WebRTC 1.0: Real-time Communication Between Browsers