akasha.rtc

Interface RTCRtpContributingSource.Builder

All Superinterfaces:

RTCRtpContributingSource

Enclosing interface:

RTCRtpContributingSource

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

The RTCRtpContributingSource dictionary of the WebRTC API is used by getContributingSources() to provide information about a given contributing source (CSRC), including the most recent time a packet that the source contributed was played out.

See Also:

RTCRtpContributingSource - MDN, (WebRTC 1.0) # dom-rtcrtpcontributingsource

Nested Class Summary

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

RTCRtpContributingSource.Builder

Method Summary

Modifier and Type	Method and Description
default RTCRtpContributingSource.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 RTCRtpContributingSource.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 RTCRtpContributingSource.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 RTCRtpContributingSource.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.RTCRtpContributingSource

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

Method Detail

audioLevel

@JsOverlay
 @Nonnull
default RTCRtpContributingSource.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 RTCRtpContributingSource.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 RTCRtpContributingSource.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 RTCRtpContributingSource.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