Class: View

Constructor

new View(streamName, tokenGenerator, mediaElementopt, autoReconnectopt)

Parameters:

Name	Type	Attributes	Default	Description
`streamName`	String			Millicast existing Stream Name where you want to connect.
`tokenGenerator`	tokenGeneratorCallback			Callback function executed when a new token is needed.
`mediaElement`	HTMLMediaElement	<optional>	null	Target HTML media element to mount stream.
`autoReconnect`	Boolean	<optional>	true	Enable auto reconnect to stream.

Source: View.js, line 16

Extends

BaseWebRTC

Methods

(async) addRemoteTrack(media, streams) → {Promise.<RTCRtpTransceiver>}

Add remote receving track.

Parameters:

Name	Type	Description
`media`	String	Media kind ('audio' \| 'video').
`streams`	Array.<MediaStream>	Streams the track will belong to.

Source: View.js, line 135

Returns:

Promise that will be resolved when the RTCRtpTransceiver is assigned an mid value.

Type:: Promise.<RTCRtpTransceiver>

(async) connect(optionsopt) → {Promise.<void>}

Connects to an active stream as subscriber.

In the example, addStreamToYourVideoTag and getYourSubscriberConnectionPath is your own implementation.

Parameters:

Name Type Attributes Description

options

Object

General subscriber options.

Properties

Name Type Attributes Default Description

dtx

Boolean

false

True to modify SDP for supporting dtx in opus. Otherwise False.

absCaptureTime

Boolean

false

True to modify SDP for supporting absolute capture time header extension. Otherwise False.

disableVideo

Boolean

false

Disable the opportunity to receive video stream.

disableAudio

Boolean

false

Disable the opportunity to receive audio stream.

multiplexedAudioTracks

Number

Number of audio tracks to recieve VAD multiplexed audio for secondary sources.

pinnedSourceId

String

Id of the main source that will be received by the default MediaStream.

excludedSourceIds

Array.<String>

Do not receive media from the these source ids.

events

Array.<String>

Override which events will be delivered by the server (any of "active" | "inactive" | "vad" | "layers" | "viewercount").*

peerConfig

RTCConfiguration

Options to configure the new RTCPeerConnection.

layer

LayerInfo

Select the simulcast encoding layer and svc layers for the main video track, leave empty for automatic layer selection based on bandwidth estimation.

forcePlayoutDelay

Object

false

Ask the server to use the playout delay header extension.

Properties

Name	Type	Attributes	Description
`min`	Number	<optional>	Set minimum playout delay value.
`max`	Number	<optional>	Set maximum playout delay value.

Source: View.js, line 114

Fires:

Returns:

Promise object which resolves when the connection was successfully established.

Type:: Promise.<void>

Examples

await millicastView.connect(options)

import View from '@millicast/sdk'

// Create media element
const videoElement = document.createElement("video")

//Define callback for generate new token
const tokenGenerator = () => getYourSubscriberInformation(accountId, streamName)

//Create a new instance
const streamName = "Millicast Stream Name where i want to connect"
const millicastView = new View(streamName, tokenGenerator, videoElement)

//Start connection to broadcast
try {
 await millicastView.connect()
} catch (e) {
 console.log('Connection failed, handle error', e)
}

import View from '@millicast/sdk'

//Define callback for generate new token
const tokenGenerator = () => getYourSubscriberInformation(accountId, streamName)

//Create a new instance
const streamName = "Millicast Stream Name where i want to connect"
const millicastView = new View(streamName, tokenGenerator)

//Set track event handler to receive streams from Publisher.
millicastView.on('track', (event) => {
  addStreamToYourVideoTag(event.streams[0])
})

//Start connection to broadcast
try {
 await millicastView.connect()
} catch (e) {
 console.log('Connection failed, handle error', e)
}

getRTCPeerConnection() → {RTCPeerConnection}

Get current RTC peer connection.

Inherited From: BaseWebRTC#getRTCPeerConnection
Source: utils/BaseWebRTC.js, line 65

Returns:

Object which represents the RTCPeerConnection.

Type:: RTCPeerConnection

isActive() → {Boolean}

Get if the current connection is active.

Inherited From: BaseWebRTC#isActive
Source: utils/BaseWebRTC.js, line 85

Returns:

True if connected, false if not.

Type:: Boolean

(async) project(sourceId, mapping)

Start projecting source in selected media ids.

Parameters:

Name Type Description

sourceId

String

Selected source id.

mapping

Array.<Object>

Mapping of the source track ids to the receiver mids

Properties

Name	Type	Attributes	Description
`trackId`	String	<optional>	Track id from the source (received on the "active" event), if not set the media kind will be used instead.
`media`	String	<optional>	Track kind of the source ('audio' \| 'video'), if not set the trackId will be used instead.
`mediaId`	String	<optional>	mid value of the rtp receiver in which the media is going to be projected. If no mediaId is defined, the first track from the main media stream with the same media type as the input source track will be used.
`layer`	LayerInfo	<optional>	Select the simulcast encoding layer and svc layers, only applicable to video tracks.
`promote`	Boolean	<optional>	To remove all existing limitations from the source, such as restricted bitrate or resolution, set this to true.

Source: View.js, line 150

(async) reconnect(dataopt)

Reconnects to last broadcast.

Parameters:

Name	Type	Attributes	Description
`data`	Object	<optional>	This object contains the error property. It may be expanded to contain more information in the future.

Properties

Name	Type	Description
`error`	String	The value sent in the first reconnect event within the error key of the payload

Inherited From: BaseWebRTC#reconnect
Source: utils/BaseWebRTC.js, line 125

Fires:

BaseWebRTC#event:reconnect

(async) select(layer)

Select the simulcast encoding layer and svc layers for the main video track

Parameters:

Name	Type	Description
`layer`	LayerInfo	leave empty for automatic layer selection based on bandwidth estimation.

Source: View.js, line 123

setReconnect()

Sets reconnection if autoReconnect is enabled.

Inherited From: BaseWebRTC#setReconnect
Source: utils/BaseWebRTC.js, line 94

stop()

Stops connection.

Inherited From: BaseWebRTC#stop
Source: utils/BaseWebRTC.js, line 72

(async) unproject(mediaIds)

Stop projecting attached source in selected media ids.

Parameters:

Name	Type	Description
`mediaIds`	Array.<String>	mid value of the receivers that are going to be detached.

Source: View.js, line 172

Events

reconnect

Emits with every reconnection attempt made when an active stream stopped unexpectedly.

Type:

Object

Properties

Name Type Description

Name	Type	Description
`timeout`	Number	Next retry interval in milliseconds.
`error`	Error	Error object with cause of failure. Possible errors are: `Signaling error: wsConnectionError` if there was an error in the Websocket connection. `Connection state change: RTCPeerConnectionState disconnected` if there was an error in the RTCPeerConnection. `Attempting to reconnect` if the reconnect was trigered externally. Or any internal error thrown by either Publish.connect or View.connect methods

timeout

Number

Next retry interval in milliseconds.

error

Error

Error object with cause of failure. Possible errors are:

Signaling error: wsConnectionError if there was an error in the Websocket connection.
Connection state change: RTCPeerConnectionState disconnected if there was an error in the RTCPeerConnection.
Attempting to reconnect if the reconnect was trigered externally.
Or any internal error thrown by either Publish.connect or View.connect methods

Inherited From: BaseWebRTC#event:reconnect
Source: utils/BaseWebRTC.js, line 130