EasyRTCClient - Foundry Virtual Tabletop Developer API

EasyRTCClient

WebRTC Client using the EasyRTC framework for its implementation. Whenever a new remote stream is received by this implementation, a call to this.webrtc.onUserStreamChange() will be made to notify of the new or deleted stream.

Constructor

new EasyRTCClient(webrtc, settings)

Source:

foundry.js, line 44824

Implements:

WebRTCInterface

Parameters:

Name	Type	Description
`webrtc`	WebRTC	The WebRTC object
`settings`	WebRTCSettings	The WebRTC Settings object

Methods

_userIdToEasyRtcId(userId) → {string|null}

Source:

foundry.js, line 45248

Returns the EasyRtcId of a user There should only be one remote peer per user

Parameters:

Name	Type	Description
`userId`	string	The ID of the user

Returns:

The EasyRtcId of the peer

Type: string | null

assignStreamToVideo(stream, video)

Source:

foundry.js, line 45157

Implements:

WebRTCInterface#assignStreamToVideo

Assigns a stream to a video element

Parameters:

Name	Type	Description
`stream`	MediaStream	The stream to assign
`video`	HTMLVideoElement	The video element to configure

(async) closeLocalStream(temporary) → {Promise}

Source:

foundry.js, line 45137

Implements:

WebRTCInterface#closeLocalStream

Closes a local media stream. If the master stream is closed, any subsequent WebRTC calls will not have any streams sent to the peer.

If @temporary is false (default), the master stream will be destroyed and all local streams removed from any existing calls. If @temporary is true, closes the temporary stream

Parameters:

Name	Type	Default	Description
`temporary`	boolean	`false`	Whether to create a temporary stream or the master stream

Returns:

Type: Promise

(async) connect(host, room, username, password) → {Promise.boolean}

Source:

foundry.js, line 44902

Implements:

WebRTCInterface#connect

Connect to the signaling server. Any existing connections will be dropped and any existing calls hung up. Once a connection to the server is established and the user authenticated, join a room and automatically establish a call with other users in that same room. If a local master media stream has been created, add a copy of that stream to every peer. Whether to include audio or video in that cloned stream, or whether to enable or disable the audio and video of the remote stream from that peer will automatically be determined based on the saved settings. This will be called again in case a server configuration changed.

In case of using FVTT server, we can't use the same socket as game.socket because the server side can close the socket in case of errors (such as disconnection/reconnection attempt where it would say that we are 'already connected' and disconnect us forcibly) and thus it could destroy the actual game's socket, so we use a different path for it instead with easyrtc.useThisSocketConnection(game.socket)

Parameters:

Name	Type	Description
`host`	string	Server host address. Set to `null` to use current FVTT server
`room`	string	Name of the room to join on the signaling server
`username`	string	Username to authenticate to the server (if needed)
`password`	string	Password to authenticate the user (if needed)

Returns:

Returns success/failure to connect

Type: Promise.boolean

(async) disconnect() → {Promise.boolean}

Source:

foundry.js, line 44995

Implements:

WebRTCInterface#disconnect

Disconnect from the signaling server, any existing calls will be terminated. This is also called whenever the server configuration is changed.

Returns:

Returns success/failure to connect

Type: Promise.boolean

(async) getAudioSinks() → {Promise.Object}

Source:

foundry.js, line 45339

Implements:

WebRTCInterface#getAudioSinks

Get the list of available audio output devices The expected result is an object with the device id as key and its human-readable label as value

Note: This feature is not supported by Firefox by default as it depends on the enumerateDevices API which doesn't list output devices on Firefox 63+ unless the media.setsinkid.enabled settings is enabled. See https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/enumerateDevices

Returns:

Type: Promise.Object

(async) getAudioSources() → {Promise.Object}

Source:

foundry.js, line 45316

Implements:

WebRTCInterface#getAudioSources

Get the list of available audio sources. The expected result is an object with the device id as key and its human-readable label as value.

Returns:

Type: Promise.Object

getConnectedStreams() → {Array.Object}

Source:

foundry.js, line 45282

Implements:

WebRTCInterface#getConnectedStreams

Get the list of connected streams The result would be an array of objects in the form of {id, pc, local, remote} where id is the user's ID, pc is the RTCPeerConnection object associated to the peer, local is the local stream added to the call and remote is the remote user's stream.

Returns:

Type: Array.Object

getStreamForUser(userId) → {MediaStream}

Source:

foundry.js, line 45264

Implements:

WebRTCInterface#getStreamForUser

Retrieve the stream for a user. Calling this with game.user.id is the proper way of retrieving the local master stream without re-initializing it. Any other user will return the remote stream of that user or null if no stream could be found.

Parameters:

Name	Type	Description
`userId`	string	ID of the user

Returns:

The remote stream of the user

Type: MediaStream

(async) getVideoSources() → {Promise.Object}

Source:

foundry.js, line 45299

Implements:

WebRTCInterface#getVideoSources

Get the list of available video sources. The expected result is an object with the device id as key and its human-readable label as value.

Returns:

Type: Promise.Object

(async) initialize() → {Promise.boolean}

Source:

foundry.js, line 44861

Implements:

WebRTCInterface#initialize

Initialize the WebRTC implementation. This Will only be called once by the main setupGame() initialization function.

Returns:

Type: Promise.boolean

(async) initLocalStream(audioSrc, videoSrc, temporary) → {Promise.MediaStream}

Source:

foundry.js, line 45026

Implements:

WebRTCInterface#initLocalStream

Initialize a local media stream Capture the local audio and video and returns the stream associated with them.

If @temporary is false (default), then this will initialize the master stream, not the actual streams being sent to individual users. However, if a master stream was already created, it will automatically get closed and every individual streams derived from it that are being sent to connected users will be removed from the calls. Each established or subsequent calls will receive a copy of the created stream (A/V depending on user permissions)

If @temporary is true then this only applies to a temporary stream and does not affect the master stream or any streams in existing calls. Note that this assumes only one temporary stream can be created at a time.

Parameters:

Name	Type	Default	Description
`audioSrc`	string \| null		ID of the audio source to capture from or null to disable Audio
`videoSrc`	string \| null		ID of the video source to capture from or null to disable Video
`temporary`	boolean	`false`	Whether to create a temporary stream or the master stream

Returns:

Returns the local stream or null if none could be created

Type: Promise.MediaStream

onSettingsChanged(changed)

Source:

foundry.js, line 45561

Implements:

WebRTCInterface#onSettingsChanged

Notify of settings changes This can be used to act according

Parameters:

Name	Type	Description
`changed`	Object	Object consisting of the changed settings in the form {key: value}

setAudioOutput(video, audioSinkId)

Source:

foundry.js, line 45173

Implements:

WebRTCInterface#setAudioOutput

Sets the audio output on a video element

Note: This feature is not supported by Firefox by default as it depends on the setSinkId API which is available behind the media.setsinkid.enabled settings in Firefox See https://developer.mozilla.org/en-US/docs/Web/API/HTMLMediaElement/setSinkId

Parameters:

Name	Type	Description
`video`	HTMLVideoElement	The video element to configure
`audioSinkId`	String	ID of the audio output device