Returns the AudioMixController for this audio-video controller.
Returns the session configuration for this audio-video controller.
Returns the EventController for this audio-video controller.
Returns the MediaStreamBroker for this audio-video controller.
Returns the RealtimeController for this audio-video controller.
Returns the RTCPeerConnection for this audio-video controller if there is one.
Returns the VideoTileController for this audio-video controller.
Called when the audio input stream is changed.
Remove the promoted attendee from the Primary meeting. This client will stop sharing audio, video, and data messages.
This will revert the end-user to precisely the state they were before a call to promoteToPrimaryMeeting
This will have no effect if there was no previous successful call to promoteToPrimaryMeeting. This
may result in AudioVideoObserver.audioVideoWasDemotedFromPrimaryMeeting but there is no need to wait for that callback
to revert UX, etc.
Dispose of this instance. The instance cannot be used after this method has been called.
Called when simulcast is enabled and simulcast uplink encoding layers get changed.
Iterates through each observer, so that their notification functions may be called.
This API will be deprecated in favor of ClientMetricReport.getRTCStatsReport().
It makes an additional call to the getStats API and therefore may cause slight performance degradation.
Please subscribe to metricsDidReceive(clientMetricReport: ClientMetricReport) callback,
and get the raw RTCStatsReport via clientMetricReport.getRTCStatsReport().
Get all the remote video sending sources.
Handles the meeting session status and returns true if it will restart the session.
Pauses receiving stream on peer connection by streamId
Allows an attendee in a Replica meeting to immediately transition to a Primary meeting attendee without need for reconnection. The promise should always return a session status even upon failure (i.e. it should never reject). See the guide for more information.
The resolved MeetingSessionStatus will contain a MeetingSessionStatusCode of the following:
MeetingSessionStatusCode.OK: The promotion was successful (i.e. session token was valid,
there was room in the Primary meeting, etc.), audio will begin flowing
and the attendee can begin to send data messages, and content/video if the call is not already at limit.MeetingSessionStatusCode.AudioAuthenticationRejected: Credentials provided
were invalid when connection attempted to Primary meeting. There may be an issue
with your mechanism which allocates the Primary meeting attendee for the Replica
meeting proxied promotion. This also may indicate that this API was called in a
non-Replica meeting.MeetingSessionStatusCode.AudioCallAtCapacity: Credentials provided were correct
but there was no room in the Primary meeting. Promotions to Primary meeting attendee take up a slot, just like
regular Primary meeting attendee connections and are limited by the same mechanisms.MeetingSessionStatusCode.SignalingBadRequest or MeetingSessionStatusCode.SignalingInternalServerError:
Other failure, possibly due to disconnect or timeout. These failures are likely retryable.Application code may also receive a callback on AudioVideoObserver.videoAvailabilityDidChange to indicate they
can begin to share video.
chime::DeleteAttendee on the Primary meeting attendee will result in AudioVideoObserver.audioVideoWasDemotedFromPrimaryMeeting
to indicate the attendee is no longer able to share.
Any disconnection will trigger an automatic demotion to avoid unexpected or unwanted promotion state on reconnection.
If the attendee still needs to be an interactive participant in the Primary meeting, promoteToPrimaryMeeting
should be called again with the same credentials.
This function should not be called until the first one resolves.
Promise which resolves to a session status for the request. See possible options above.
Restarts the peer connection and/or the session.
Restarts the local audio. This function assumes the peer connection is established and an active audio stream must be chosen in DeviceController
Replaces the local video device without a WebRTC negotiation on the sender side and
avoids that all video subscribers to re-negotiate at the same time.
This is an optimized way of replacing video stream and requires WebRTC Unified Plan.
Restarts the local video device. The callback is called when the device has been restarted.
Resumes receiving stream on peer connection by streamId
Sets the audio profile to use for audio. The new audio profile takes effect
on the next call to start or if already started, upon the next reconnect.
Set codec preferences for this clients send stream in order of most preferred to least preferred. The controller will fallback for one of two reasons
If there is no overlap between what is passed in and what is supported by the browser, this function may not have any effect, and the default set of codecs for this browser will be used.
Sets the max bandwidth for video publishing
Start the meeting session. By default this will connect and begin sending and receiving audio, depending on the implementation.
This method also allows you to provide options for how connection occurs.
The only supported option is signalingOnly. Specifying this option will
cause the controller to only connect the meeting signaling channel. This
can be performed relatively early in the join lifecycle (e.g., prior to
choosing audio devices), which can improve join latency.
Your code is responsible for calling start again without signalingOnly
to complete connection.
Updates the peer connection when video tiles have changed. If needsRenegotiation
is set explicitly to false, we may skip SDP negotiation if it is not necessary.
Called when the video input stream is changed.
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Returns the ActiveSpeakerDetector for this audio-video controller.