RTCRtpSender
方法
setStreams()
associates the sender's
track
采用指定
MediaStream
or array of
MediaStream
对象。
rtcRtpSender.setStreams(mediaStream); rtcRtpSender.setStreams([mediaStream...]);
mediaStream
or
[mediaStream...]
可选
MediaStream
object—or an array of multiple
MediaStream
objects—identifying the streams to which the
RTCRtpSender
's
track
belongs. If this parameter isn't specified, no new streams will be associated with the track.
None.
InvalidStateError
The sender's connection is closed.
setStreams()
is purely additive. It doesn't remove the track from any streams; it simply adds it to new ones. If you specify streams to which the track already belongs, that stream is unaffected.
Once the track has been added to all of the streams, renegotiation of the connection will be triggered by the
negotiationneeded
event being dispatched to the
RTCPeerConnection
to which the sender belongs.
This example adds all of an
RTCPeerConnection
's tracks to the specified stream.
function addTracksToStream(stream) {
let senders = pc.getSenders();
senders.forEach((sender) => {
if (sender.track && (sender.transport.state === connected)) {
sender.setStreams(stream);
}
});
}
After calling the
RTCPeerConnection
方法
getSenders()
to get the list of the connection's senders, the
addTracksToStream()
function iterates over the list. For each sender, if the sender's track is non-null and its transport's state is
connected
, we call
setStreams()
to add the track to the
stream
指定。
| 规范 | 状态 | 注释 |
|---|---|---|
|
WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpSender.setStreams()' in that specification. |
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