akasha.rtc

Interface RTCRtpSynchronizationSource.Builder

All Superinterfaces:

RTCRtpContributingSource, RTCRtpSynchronizationSource

Enclosing interface:

RTCRtpSynchronizationSource

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

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

Nested Class Summary

Nested classes/interfaces inherited from interface akasha.rtc.RTCRtpSynchronizationSource

RTCRtpSynchronizationSource.Builder

Method Summary

Modifier and Type	Method and Description
default RTCRtpSynchronizationSource.Builder	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.
default RTCRtpSynchronizationSource.Builder	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.Builder	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.Builder	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.RTCRtpSynchronizationSource

create

Methods inherited from interface akasha.rtc.RTCRtpContributingSource

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

Method Detail

audioLevel

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource.Builder 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.

See Also:

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

rtpTimestamp

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource.Builder 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.

See Also:

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

source

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource.Builder source(int source)

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

See Also:

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

timestamp

@JsOverlay
 @Nonnull
default RTCRtpSynchronizationSource.Builder 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.

See Also:

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