akasha.rtc

Interface RTCRtpContributingSource

All Known Subinterfaces:

RTCRtpContributingSource.Builder, RTCRtpSynchronizationSource, RTCRtpSynchronizationSource.Builder

@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, # dom-rtcrtpcontributingsource

Nested Class Summary

Nested Classes

Modifier and Type	Interface and Description
static interface	RTCRtpContributingSource.Builder 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.
static interface	RTCRtpContributingSource.Step1
static interface	RTCRtpContributingSource.Step2

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.
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.
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.
static RTCRtpContributingSource.Step1	timestamp(double timestamp)

Method Detail

timestamp

@JsOverlay
 @Nonnull
static RTCRtpContributingSource.Step1 timestamp(double timestamp)

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

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

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

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