RTCRtpReceiver: getContributingSources() method
Baseline
Widely available
*
This feature is well established and works across many devices and browser versions. It’s been available across browsers since March 2019.
* Some parts of this feature may have varying levels of support.
The getContributingSources() method of the RTCRtpReceiver interface returns an array of objects, each corresponding to one CSRC (contributing source) identifier received by the current RTCRtpReceiver in the last ten seconds.
Syntax
getContributingSources()
Parameters
None.
Return value
An array of objects, each describing one of the contributing sources that provided data to the incoming stream in the past ten seconds. These objects contain the following properties:
audioLevelOptional-
A floating-point value between 0.0 and 1.0 specifying the audio level contained in the last RTP packet played from the contributing source.
The value is on a linear scale and is defined in units of dBov, or decibels (overload). This is the amplitude relative to the point at which clipping of the audio begins to occur. A value of 1.0 represents 0 dBov (maximum volume), a value of 0.0 represents silence, and a value of 0.5 represents approximately 6 dB SPL (decibels of sound pressure level) change in the sound pressure level from 0 dBov.
If this property is not present then no volume level was provided by the source.
rtpTimestampOptional-
The RTP timestamp (an integer
DOMHighResTimeStamp) of the media. This source-generated timestamp indicates the time at which the media in this packet, scheduled for play out at the time indicated bytimestamp, was initially sampled or generated. It may be useful for sequencing and synchronization purposes. sourceOptional-
A positive integer value specifying the CSRC identifier of the contributing source. This uniquely identifies the source of the particular stream RTP packets.
timestampOptional-
A
DOMHighResTimeStampindicating the most recent time at which a frame originating from this source was delivered to the receiver'sMediaStreamTrack
Specifications
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| WebRTC: Real-Time Communication in Browsers # dom-rtcrtpreceiver-getcontributingsources |
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getContributingSources
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audioLevel property in returned object
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Audio tracks supported
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rtpTimestamp property in returned object
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source property in returned object
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timestamp property in returned object
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Video tracks supported
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Legend
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- Full support
- Full support
- No support
- No support
- See implementation notes.