Constructor
new EasyRTCClient(webrtc, settings)
- Source:
- Implements:
Parameters:
| Name | Type | Description |
|---|---|---|
webrtc |
WebRTC | The WebRTC object |
settings |
WebRTCSettings | The WebRTC Settings object |
Methods
_userIdToEasyRtcId(userId) → {string|null}
- Source:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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 |
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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:
- Implements:
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 |