akasha.rtc

Interface RTCRtpContributingSource

All Known Subinterfaces:

RTCRtpSynchronizationSource

@Generated(value="org.realityforge.webtack")
 @JsType(isNative=true,
        namespace="<global>",
        name="Object")
public interface 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, RTCRtpContributingSource - WebRTC 1.0: Real-time Communication Between Browsers

Method Summary

Modifier and Type	Method and Description
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	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 RTCRtpContributingSource	create(int rtpTimestamp, int source, double timestamp)
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	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.
void	setAudioLevel(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.
void	setRtpTimestamp(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.
void	setSource(int source) The read-only source property of the RTCRtpContributingSource interface returns the source identifier of a particular stream of RTP packets.
void	setTimestamp(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.
int	source() The read-only source property of the RTCRtpContributingSource interface returns the source identifier of a particular stream of RTP packets.
default RTCRtpContributingSource	source(int source) The read-only source property of the RTCRtpContributingSource interface returns the source identifier of a particular stream of RTP packets.
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.
default RTCRtpContributingSource	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.

Method Detail

create

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

audioLevel

@JsProperty(name="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

setAudioLevel

@JsProperty
void setAudioLevel(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

audioLevel

@JsOverlay
 @Nonnull
default RTCRtpContributingSource 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

@JsProperty(name="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

setRtpTimestamp

@JsProperty
void setRtpTimestamp(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

rtpTimestamp

@JsOverlay
 @Nonnull
default RTCRtpContributingSource 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

@JsProperty(name="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

setSource

@JsProperty
void setSource(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

source

@JsOverlay
 @Nonnull
default RTCRtpContributingSource 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

@JsProperty(name="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

setTimestamp

@JsProperty
void setTimestamp(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

timestamp

@JsOverlay
 @Nonnull
default RTCRtpContributingSource 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