mirrors/gst-plugins-rs
mirrors/gst-plugins-rs
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git synced 2024-11-25 04:51:26 +00:00
Code Issues Projects Releases Wiki Activity

Implement GstWebRTCAPI as class instead of global instance

Browse source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1373>
This commit is contained in:
Yorick Smilda 2024-02-07 11:32:21 +01:00 committed by GStreamer Marge Bot
parent 594400a7f5
commit 563eff1193
10 changed files with 391 additions and 424 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

33
net/webrtc/gstwebrtc-api/index.html
View file

@ -220,13 +220,7 @@
}
</style>
<script>
const signalingProtocol = window.location.protocol.startsWith("https") ? "wss" : "ws";
window.gstWebRTCConfig = {
meta: { name: `WebClient-${Date.now()}` },
signalingServerUrl: `${signalingProtocol}://${window.location.host}/webrtc`
};
function initCapture() {
function initCapture(api) {
const captureSection = document.getElementById("capture");
const clientIdElement = captureSection.querySelector(".client-id");
const videoElement = captureSection.getElementsByTagName("video")[0];
@ -235,7 +229,7 @@
connected: function(clientId) { clientIdElement.textContent = clientId; },
disconnected: function() { clientIdElement.textContent = "none"; }
};
gstWebRTCAPI.registerConnectionListener(listener);
api.registerConnectionListener(listener);
document.getElementById("capture-button").addEventListener("click", (event) => {
event.preventDefault();
@ -249,7 +243,7 @@
video: { width: 1280, height: 720 }
};
navigator.mediaDevices.getUserMedia(constraints).then((stream) => {
const session = gstWebRTCAPI.createProducerSession(stream);
const session = api.createProducerSession(stream);
if (session) {
captureSection._producerSession = session;
@ -270,7 +264,7 @@
session.addEventListener("stateChanged", (event) => {
if ((captureSection._producerSession === session) &&
(event.target.state === gstWebRTCAPI.SessionState.streaming)) {
(event.target.state === GstWebRTCAPI.SessionState.streaming)) {
videoElement.srcObject = stream;
videoElement.play().catch(() => {});
captureSection.classList.remove("starting");
@ -306,7 +300,7 @@
});
}
function initRemoteStreams() {
function initRemoteStreams(api) {
const remoteStreamsElement = document.getElementById("remote-streams");
const listener = {
@ -347,7 +341,7 @@
if (entryElement._consumerSession) {
entryElement._consumerSession.close();
} else {
const session = gstWebRTCAPI.createConsumerSession(producerId);
const session = api.createConsumerSession(producerId);
if (session) {
entryElement._consumerSession = session;
@ -408,8 +402,8 @@
}
};
gstWebRTCAPI.registerProducersListener(listener);
for (const producer of gstWebRTCAPI.getAvailableProducers()) {
api.registerProducersListener(listener);
for (const producer of api.getAvailableProducers()) {
listener.producerAdded(producer);
}
}
@ -422,8 +416,15 @@
}
});
initCapture();
initRemoteStreams();
const signalingProtocol = window.location.protocol.startsWith("https") ? "wss" : "ws";
const gstWebRTCConfig = {
meta: { name: `WebClient-${Date.now()}` },
signalingServerUrl: `${signalingProtocol}://${window.location.host}/webrtc`,
};
const api = new GstWebRTCAPI(gstWebRTCConfig);
initCapture(api);
initRemoteStreams(api);
});
</script>
</head>

2
net/webrtc/gstwebrtc-api/package.json
View file

@ -1,6 +1,6 @@
{
"name": "gstwebrtc-api",
"version": "1.0.1",
"version": "2.0.0",
"description": "Javascript API to integrate GStreamer WebRTC streams (webrtcsrc/webrtcsink) in a web browser",
"keywords": [
"webrtc",

9
net/webrtc/gstwebrtc-api/src/config.js
View file

@ -11,17 +11,16 @@
/**
* GStreamer WebRTC configuration.
* <p>You can override default values by defining configuration before receiving the <i>DOMContentLoaded</i> event.<br>
* Once the <i>DOMContentLoaded</i> event triggered, changing configuration will have no effect.</p>
* <p>The config can be passed on creation of the GstWebRTCAPI class.</p>
* <p>For example:
* <pre>
* const signalingProtocol = window.location.protocol.startsWith("https") ? "wss" : "ws";
* window.gstWebRTCConfig = {
* const api = new GstWebRTCAPI({
* meta: { name: `WebClient-${Date.now()}` },
* signalingServerUrl: `${signalingProtocol}://${window.location.host}/webrtc`
* };
* });
* </pre></p>
* @typedef {object} gstWebRTCConfig
* @typedef {object} GstWebRTCConfig
* @property {object} meta=null - Client free-form information that will be exchanged with all peers through the
* signaling <i>meta</i> property, its content depends on your application.
* @property {string} signalingServerUrl=ws://127.0.0.1:8443 - The WebRTC signaling server URL.

32
net/webrtc/gstwebrtc-api/src/consumer-session.js
View file

@ -15,28 +15,28 @@ import RemoteController from "./remote-controller.js";
/**
* Event name: "streamsChanged".<br>
* Triggered when the underlying media streams of a {@link gstWebRTCAPI.ConsumerSession} change.
* @event gstWebRTCAPI#StreamsChangedEvent
* Triggered when the underlying media streams of a {@link GstWebRTCAPI.ConsumerSession} change.
* @event GstWebRTCAPI#StreamsChangedEvent
* @type {external:Event}
* @see gstWebRTCAPI.ConsumerSession#streams
* @see GstWebRTCAPI.ConsumerSession#streams
*/
/**
* Event name: "remoteControllerChanged".<br>
* Triggered when the underlying remote controller of a {@link gstWebRTCAPI.ConsumerSession} changes.
* @event gstWebRTCAPI#RemoteControllerChangedEvent
* Triggered when the underlying remote controller of a {@link GstWebRTCAPI.ConsumerSession} changes.
* @event GstWebRTCAPI#RemoteControllerChangedEvent
* @type {external:Event}
* @see gstWebRTCAPI.ConsumerSession#remoteController
* @see GstWebRTCAPI.ConsumerSession#remoteController
*/
/**
* @class gstWebRTCAPI.ConsumerSession
* @class GstWebRTCAPI.ConsumerSession
* @hideconstructor
* @classdesc Consumer session managing a peer-to-peer WebRTC channel between a remote producer and this client
* instance.
* <p>Call {@link gstWebRTCAPI#createConsumerSession} to create a ConsumerSession instance.</p>
* @extends {gstWebRTCAPI.WebRTCSession}
* @fires {@link gstWebRTCAPI#event:StreamsChangedEvent}
* @fires {@link gstWebRTCAPI#event:RemoteControllerChangedEvent}
* <p>Call {@link GstWebRTCAPI#createConsumerSession} to create a ConsumerSession instance.</p>
* @extends {GstWebRTCAPI.WebRTCSession}
* @fires {@link GstWebRTCAPI#event:StreamsChangedEvent}
* @fires {@link GstWebRTCAPI#event:RemoteControllerChangedEvent}
*/
export default class ConsumerSession extends WebRTCSession {
constructor(peerId, comChannel) {
@ -55,7 +55,7 @@ export default class ConsumerSession extends WebRTCSession {
/**
* The array of remote media streams consumed locally through this WebRTC channel.
* @member {external:MediaStream[]} gstWebRTCAPI.ConsumerSession#streams
* @member {external:MediaStream[]} GstWebRTCAPI.ConsumerSession#streams
* @readonly
*/
get streams() {
@ -65,7 +65,7 @@ export default class ConsumerSession extends WebRTCSession {
/**
* The remote controller associated with this WebRTC consumer session. Value may be null if consumer session
* has no remote controller.
* @member {gstWebRTCAPI.RemoteController} gstWebRTCAPI.ConsumerSession#remoteController
* @member {GstWebRTCAPI.RemoteController} GstWebRTCAPI.ConsumerSession#remoteController
* @readonly
*/
get remoteController() {
@ -77,10 +77,10 @@ export default class ConsumerSession extends WebRTCSession {
* This method must be called after creating the consumer session in order to start receiving the remote streams.
* It registers this consumer session to the signaling server and gets ready to receive audio/video streams.
* <p>Even on success, streaming can fail later if any error occurs during or after connection. In order to know
* the effective streaming state, you should be listening to the [error]{@link gstWebRTCAPI#event:ErrorEvent},
* [stateChanged]{@link gstWebRTCAPI#event:StateChangedEvent} and/or [closed]{@link gstWebRTCAPI#event:ClosedEvent}
* the effective streaming state, you should be listening to the [error]{@link GstWebRTCAPI#event:ErrorEvent},
* [stateChanged]{@link GstWebRTCAPI#event:StateChangedEvent} and/or [closed]{@link GstWebRTCAPI#event:ClosedEvent}
* events.</p>
* @method gstWebRTCAPI.ConsumerSession#connect
* @method GstWebRTCAPI.ConsumerSession#connect
* @returns {boolean} true in case of success (may fail later during or after connection) or false in case of
* immediate error (wrong session state or no connection to the signaling server).
*/

606
net/webrtc/gstwebrtc-api/src/gstwebrtc-api.js
View file

@ -13,367 +13,337 @@ import defaultConfig from "./config.js";
import ComChannel from "./com-channel.js";
import SessionState from "./session-state.js";
const apiState = {
config: null,
channel: null,
producers: {},
connectionListeners: [],
producersListeners: []
};
/**
* @class GstWebRTCAPI
* @classdesc The API entry point that manages a WebRTC.
*/
export default class GstWebRTCAPI {
/**
* @constructor GstWebRTCAPI
* @param {GstWebRTCConfig} [userConfig] - The user configuration.<br>Only the parameters different from the default
* ones need to be provided.
*/
constructor(userConfig) {
this._channel = null;
this._producers = {};
this._connectionListeners = [];
this._producersListeners = [];
/**
* @interface gstWebRTCAPI.ConnectionListener
*/
/**
* Callback method called when this client connects to the WebRTC signaling server.
* The callback implementation should not throw any exception.
* @method gstWebRTCAPI.ConnectionListener#connected
* @abstract
* @param {string} clientId - The unique identifier of this WebRTC client.<br>This identifier is provided by the
* signaling server to uniquely identify each connected peer.
*/
/**
* Callback method called when this client disconnects from the WebRTC signaling server.
* The callback implementation should not throw any exception.
* @method gstWebRTCAPI.ConnectionListener#disconnected
* @abstract
*/
const config = Object.assign({}, defaultConfig);
if (userConfig && (typeof (userConfig) === "object")) {
Object.assign(config, userConfig);
}
/**
* Registers a connection listener that will be called each time the WebRTC API connects to or disconnects from the
* signaling server.
* @function
* @memberof gstWebRTCAPI
* @instance
* @param {gstWebRTCAPI.ConnectionListener} listener - The connection listener to register.
* @returns {boolean} true in case of success (or if the listener was already registered), or false if the listener
* doesn't implement all callback functions and cannot be registered.
*/
function registerConnectionListener(listener) {
if (!listener || (typeof (listener) !== "object") ||
(typeof (listener.connected) !== "function") ||
(typeof (listener.disconnected) !== "function")) {
return false;
if (typeof (config.meta) !== "object") {
config.meta = null;
}
this._config = config;
this.connectChannel();
}
if (!apiState.connectionListeners.includes(listener)) {
apiState.connectionListeners.push(listener);
}
/**
* @interface GstWebRTCAPI.ConnectionListener
*/
/**
* Callback method called when this client connects to the WebRTC signaling server.
* The callback implementation should not throw any exception.
* @method GstWebRTCAPI.ConnectionListener#connected
* @abstract
* @param {string} clientId - The unique identifier of this WebRTC client.<br>This identifier is provided by the
* signaling server to uniquely identify each connected peer.
*/
/**
* Callback method called when this client disconnects from the WebRTC signaling server.
* The callback implementation should not throw any exception.
* @method GstWebRTCAPI.ConnectionListener#disconnected
* @abstract
*/
return true;
}
/**
* Registers a connection listener that will be called each time the WebRTC API connects to or disconnects from the
* signaling server.
* @method GstWebRTCAPI#registerConnectionListener
* @param {GstWebRTCAPI.ConnectionListener} listener - The connection listener to register.
* @returns {boolean} true in case of success (or if the listener was already registered), or false if the listener
* doesn't implement all callback functions and cannot be registered.
*/
registerConnectionListener(listener) {
if (!listener || (typeof (listener) !== "object") ||
(typeof (listener.connected) !== "function") ||
(typeof (listener.disconnected) !== "function")) {
return false;
}
if (!this._connectionListeners.includes(listener)) {
this._connectionListeners.push(listener);
}
/**
* Unregisters a connection listener.<br>
* The removed listener will never be called again and can be garbage collected.
* @function
* @memberof gstWebRTCAPI
* @instance
* @param {gstWebRTCAPI.ConnectionListener} listener - The connection listener to unregister.
* @returns {boolean} true if the listener is found and unregistered, or false if the listener was not previously
* registered.
*/
function unregisterConnectionListener(listener) {
const idx = apiState.connectionListeners.indexOf(listener);
if (idx >= 0) {
apiState.connectionListeners.splice(idx, 1);
return true;
}
return false;
}
/**
* Unregisters a connection listener.<br>
* The removed listener will never be called again and can be garbage collected.
* @method GstWebRTCAPI#unregisterConnectionListener
* @param {GstWebRTCAPI.ConnectionListener} listener - The connection listener to unregister.
* @returns {boolean} true if the listener is found and unregistered, or false if the listener was not previously
* registered.
*/
unregisterConnectionListener(listener) {
const idx = this._connectionListeners.indexOf(listener);
if (idx >= 0) {
this._connectionListeners.splice(idx, 1);
return true;
}
/**
* Unregisters all previously registered connection listeners.
* @function
* @memberof gstWebRTCAPI
* @instance
*/
function unregisterAllConnectionListeners() {
apiState.connectionListeners = [];
}
/**
* Creates a new producer session.
* <p>You can only create a producer session at once.<br>
* To request streaming from a new stream you will first need to close the previous producer session.</p>
* <p>You can only request a producer session while you are connected to the signaling server. You can use the
* {@link gstWebRTCAPI.ConnectionListener} interface and {@link gstWebRTCAPI#registerConnectionListener} function to
* listen to the connection state.</p>
* @function
* @memberof gstWebRTCAPI
* @instance
* @param {external:MediaStream} stream - The audio/video stream to offer as a producer through WebRTC.
* @returns {gstWebRTCAPI.ProducerSession} The created producer session or null in case of error. To start streaming,
* you still need to call {@link gstWebRTCAPI.ProducerSession#start} after adding on the returned session all the event
* listeners you may need.
*/
function createProducerSession(stream) {
if (apiState.channel) {
return apiState.channel.createProducerSession(stream);
} else {
return null;
}
}
/**
* Information about a remote producer registered by the signaling server.
* @typedef {object} gstWebRTCAPI.Producer
* @readonly
* @property {string} id - The remote producer unique identifier set by the signaling server (always non-empty).
* @property {object} meta - Free-form object containing extra information about the remote producer (always non-null,
* but may be empty). Its content depends on your application.
*/
/**
* Gets the list of all remote WebRTC producers available on the signaling server.
* <p>The remote producers list is only populated once you've connected to the signaling server. You can use the
* {@link gstWebRTCAPI.ConnectionListener} interface and {@link gstWebRTCAPI#registerConnectionListener} function to
* listen to the connection state.</p>
* @function
* @memberof gstWebRTCAPI
* @instance
* @returns {gstWebRTCAPI.Producer[]} The list of remote WebRTC producers available.
*/
function getAvailableProducers() {
return Object.values(apiState.producers);
}
/**
* @interface gstWebRTCAPI.ProducersListener
*/
/**
* Callback method called when a remote producer is added on the signaling server.
* The callback implementation should not throw any exception.
* @method gstWebRTCAPI.ProducersListener#producerAdded
* @abstract
* @param {gstWebRTCAPI.Producer} producer - The remote producer added on server-side.
*/
/**
* Callback method called when a remote producer is removed from the signaling server.
* The callback implementation should not throw any exception.
* @method gstWebRTCAPI.ProducersListener#producerRemoved
* @abstract
* @param {gstWebRTCAPI.Producer} producer - The remote producer removed on server-side.
*/
/**
* Registers a producers listener that will be called each time a producer is added or removed on the signaling
* server.
* @function
* @memberof gstWebRTCAPI
* @instance
* @param {gstWebRTCAPI.ProducersListener} listener - The producer listener to register.
* @returns {boolean} true in case of success (or if the listener was already registered), or false if the listener
* doesn't implement all callback functions and cannot be registered.
*/
function registerProducersListener(listener) {
if (!listener || (typeof (listener) !== "object") ||
(typeof (listener.producerAdded) !== "function") ||
(typeof (listener.producerRemoved) !== "function")) {
return false;
}
if (!apiState.producersListeners.includes(listener)) {
apiState.producersListeners.push(listener);
/**
* Unregisters all previously registered connection listeners.
* @method GstWebRTCAPI#unregisterAllConnectionListeners
*/
unregisterAllConnectionListeners() {
this._connectionListeners = [];
}
return true;
}
/**
* Creates a new producer session.
* <p>You can only create one producer session at a time.<br>
* To request streaming from a new stream you will first need to close the previous producer session.</p>
* <p>You can only request a producer session while you are connected to the signaling server. You can use the
* {@link GstWebRTCAPI.ConnectionListener} interface and {@link GstWebRTCAPI#registerConnectionListener} method to
* listen to the connection state.</p>
* @method GstWebRTCAPI#createProducerSession
* @param {external:MediaStream} stream - The audio/video stream to offer as a producer through WebRTC.
* @returns {GstWebRTCAPI.ProducerSession} The created producer session or null in case of error. To start streaming,
* you still need to call {@link GstWebRTCAPI.ProducerSession#start} after adding on the returned session all the event
* listeners you may need.
*/
createProducerSession(stream) {
if (this._channel) {
return this._channel.createProducerSession(stream);
}
return null;
}
/**
* Information about a remote producer registered by the signaling server.
* @typedef {object} GstWebRTCAPI.Producer
* @readonly
* @property {string} id - The remote producer unique identifier set by the signaling server (always non-empty).
* @property {object} meta - Free-form object containing extra information about the remote producer (always non-null,
* but may be empty). Its content depends on your application.
*/
/**
* Gets the list of all remote WebRTC producers available on the signaling server.
* <p>The remote producers list is only populated once you've connected to the signaling server. You can use the
* {@link GstWebRTCAPI.ConnectionListener} interface and {@link GstWebRTCAPI#registerConnectionListener} method to
* listen to the connection state.</p>
* @method GstWebRTCAPI#getAvailableProducers
* @returns {GstWebRTCAPI.Producer[]} The list of remote WebRTC producers available.
*/
getAvailableProducers() {
return Object.values(this._producers);
}
/**
* @interface GstWebRTCAPI.ProducersListener
*/
/**
* Callback method called when a remote producer is added on the signaling server.
* The callback implementation should not throw any exception.
* @method GstWebRTCAPI.ProducersListener#producerAdded
* @abstract
* @param {GstWebRTCAPI.Producer} producer - The remote producer added on server-side.
*/
/**
* Callback method called when a remote producer is removed from the signaling server.
* The callback implementation should not throw any exception.
* @method GstWebRTCAPI.ProducersListener#producerRemoved
* @abstract
* @param {GstWebRTCAPI.Producer} producer - The remote producer removed on server-side.
*/
/**
* Registers a producers listener that will be called each time a producer is added or removed on the signaling
* server.
* @method GstWebRTCAPI#registerProducersListener
* @param {GstWebRTCAPI.ProducersListener} listener - The producer listener to register.
* @returns {boolean} true in case of success (or if the listener was already registered), or false if the listener
* doesn't implement all callback functions and cannot be registered.
*/
registerProducersListener(listener) {
if (!listener || (typeof (listener) !== "object") ||
(typeof (listener.producerAdded) !== "function") ||
(typeof (listener.producerRemoved) !== "function")) {
return false;
}
if (!this._producersListeners.includes(listener)) {
this._producersListeners.push(listener);
}
/**
* Unregisters a producers listener.<br>
* The removed listener will never be called again and can be garbage collected.
* @function
* @memberof gstWebRTCAPI
* @instance
* @param {gstWebRTCAPI.ProducersListener} listener - The producers listener to unregister.
* @returns {boolean} true if the listener is found and unregistered, or false if the listener was not previously
* registered.
*/
function unregisterProducersListener(listener) {
const idx = apiState.producersListeners.indexOf(listener);
if (idx >= 0) {
apiState.producersListeners.splice(idx, 1);
return true;
}
return false;
}
/**
* Unregisters a producers listener.<br>
* The removed listener will never be called again and can be garbage collected.
* @method GstWebRTCAPI#unregisterProducersListener
* @param {GstWebRTCAPI.ProducersListener} listener - The producers listener to unregister.
* @returns {boolean} true if the listener is found and unregistered, or false if the listener was not previously
* registered.
*/
unregisterProducersListener(listener) {
const idx = this._producersListeners.indexOf(listener);
if (idx >= 0) {
this._producersListeners.splice(idx, 1);
return true;
}
/**
* Unregisters all previously registered producers listeners.
* @function
* @memberof gstWebRTCAPI
* @instance
*/
function unregisterAllProducersListeners() {
apiState.producersListeners = [];
}
return false;
}
/**
* Creates a consumer session by connecting the local client to a remote WebRTC producer.
* <p>You can only create one consumer session per remote producer.</p>
* <p>You can only request a new consumer session while you are connected to the signaling server. You can use the
* {@link gstWebRTCAPI.ConnectionListener} interface and {@link gstWebRTCAPI#registerConnectionListener} function to
* listen to the connection state.</p>
* @function
* @memberof gstWebRTCAPI
* @instance
* @param {string} producerId - The unique identifier of the remote producer to connect to.
* @returns {gstWebRTCAPI.ConsumerSession} The WebRTC session between the selected remote producer and this local
* consumer, or null in case of error. To start connecting and receiving the remote streams, you still need to call
* {@link gstWebRTCAPI.ConsumerSession#connect} after adding on the returned session all the event listeners you may
* need.
*/
function createConsumerSession(producerId) {
if (apiState.channel) {
return apiState.channel.createConsumerSession(producerId);
} else {
/**
* Unregisters all previously registered producers listeners.
* @method GstWebRTCAPI#unregisterAllProducersListeners
*/
unregisterAllProducersListeners() {
this._producersListeners = [];
}
/**
* Creates a consumer session by connecting the local client to a remote WebRTC producer.
* <p>You can only create one consumer session per remote producer.</p>
* <p>You can only request a new consumer session while you are connected to the signaling server. You can use the
* {@link GstWebRTCAPI.ConnectionListener} interface and {@link GstWebRTCAPI#registerConnectionListener} method to
* listen to the connection state.</p>
* @method GstWebRTCAPI#createConsumerSession
* @param {string} producerId - The unique identifier of the remote producer to connect to.
* @returns {GstWebRTCAPI.ConsumerSession} The WebRTC session between the selected remote producer and this local
* consumer, or null in case of error. To start connecting and receiving the remote streams, you still need to call
* {@link GstWebRTCAPI.ConsumerSession#connect} after adding on the returned session all the event listeners you may
* need.
*/
createConsumerSession(producerId) {
if (this._channel) {
return this._channel.createConsumerSession(producerId);
}
return null;
}
}
/**
* The GStreamer WebRTC Javascript API.
* @namespace gstWebRTCAPI
*/
const gstWebRTCAPI = Object.freeze({
SessionState: SessionState,
registerConnectionListener: registerConnectionListener,
unregisterConnectionListener: unregisterConnectionListener,
unregisterAllConnectionListeners: unregisterAllConnectionListeners,
createProducerSession: createProducerSession,
getAvailableProducers: getAvailableProducers,
registerProducersListener: registerProducersListener,
unregisterProducersListener: unregisterProducersListener,
unregisterAllProducersListeners: unregisterAllProducersListeners,
createConsumerSession: createConsumerSession
});
function triggerConnected(clientId) {
for (const listener of apiState.connectionListeners) {
try {
listener.connected(clientId);
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
connectChannel() {
if (this._channel) {
const oldChannel = this._channel;
this._channel = null;
oldChannel.close();
for (const key in this._producers) {
this.triggerProducerRemoved(key);
}
this._producers = {};
this.triggerDisconnected();
}
}
}
function triggerDisconnected() {
for (const listener of apiState.connectionListeners) {
try {
listener.disconnected();
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
}
}
}
this._channel = new ComChannel(
this._config.signalingServerUrl,
this._config.meta,
this._config.webrtcConfig
);
function triggerProducerAdded(producer) {
if (producer.id in apiState.producers) {
return;
this._channel.addEventListener("error", (event) => {
if (event.target === this._channel) {
console.error(event.message, event.error);
}
});
this._channel.addEventListener("closed", (event) => {
if (event.target !== this._channel) {
return;
}
this._channel = null;
for (const key in this._producers) {
this.triggerProducerRemoved(key);
}
this._producers = {};
this.triggerDisconnected();
if (this._config.reconnectionTimeout > 0) {
window.setTimeout(() => {
this.connectChannel();
}, this._config.reconnectionTimeout);
}
});
this._channel.addEventListener("ready", (event) => {
if (event.target === this._channel) {
this.triggerConnected(this._channel.channelId);
}
});
this._channel.addEventListener("producerAdded", (event) => {
if (event.target === this._channel) {
this.triggerProducerAdded(event.detail);
}
});
this._channel.addEventListener("producerRemoved", (event) => {
if (event.target === this._channel) {
this.triggerProducerRemoved(event.detail.id);
}
});
}
apiState.producers[producer.id] = producer;
for (const listener of apiState.producersListeners) {
try {
listener.producerAdded(producer);
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
}
}
}
function triggerProducerRemoved(producerId) {
if (producerId in apiState.producers) {
const producer = apiState.producers[producerId];
delete apiState.producers[producerId];
for (const listener of apiState.producersListeners) {
triggerConnected(clientId) {
for (const listener of this._connectionListeners) {
try {
listener.producerRemoved(producer);
listener.connected(clientId);
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
}
}
}
}
function connectChannel() {
if (apiState.channel) {
const oldChannel = apiState.channel;
apiState.channel = null;
oldChannel.close();
for (const key in apiState.producers) {
triggerProducerRemoved(key);
}
apiState.producers = {};
triggerDisconnected();
}
apiState.channel = new ComChannel(
apiState.config.signalingServerUrl,
apiState.config.meta,
apiState.config.webrtcConfig);
apiState.channel.addEventListener("error", (event) => {
if (event.target === apiState.channel) {
console.error(event.message, event.error);
}
});
apiState.channel.addEventListener("closed", (event) => {
if (event.target === apiState.channel) {
apiState.channel = null;
for (const key in apiState.producers) {
triggerProducerRemoved(key);
}
apiState.producers = {};
triggerDisconnected();
if (apiState.config.reconnectionTimeout > 0) {
window.setTimeout(connectChannel, apiState.config.reconnectionTimeout);
triggerDisconnected() {
for (const listener of this._connectionListeners) {
try {
listener.disconnected();
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
}
}
});
}
apiState.channel.addEventListener("ready", (event) => {
if (event.target === apiState.channel) {
triggerConnected(apiState.channel.channelId);
triggerProducerAdded(producer) {
if (producer.id in this._producers) {
return;
}
});
apiState.channel.addEventListener("producerAdded", (event) => {
if (event.target === apiState.channel) {
triggerProducerAdded(event.detail);
this._producers[producer.id] = producer;
for (const listener of this._producersListeners) {
try {
listener.producerAdded(producer);
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
}
}
});
}
apiState.channel.addEventListener("producerRemoved", (event) => {
if (event.target === apiState.channel) {
triggerProducerRemoved(event.detail.id);
triggerProducerRemoved(producerId) {
if (producerId in this._producers) {
const producer = this._producers[producerId];
delete this._producers[producerId];
for (const listener of this._producersListeners) {
try {
listener.producerRemoved(producer);
} catch (ex) {
console.error("a listener callback should not throw any exception", ex);
}
}
}
});
}
}
function start(userConfig) {
if (apiState.config) {
throw new Error("GstWebRTC API is already started");
}
const config = Object.assign({}, defaultConfig);
if (userConfig && (typeof (userConfig) === "object")) {
Object.assign(config, userConfig);
}
if (typeof (config.meta) !== "object") {
config.meta = null;
}
apiState.config = config;
connectChannel();
}
export { gstWebRTCAPI, start };
GstWebRTCAPI.SessionState = SessionState;

9
net/webrtc/gstwebrtc-api/src/index.js
View file

@ -10,7 +10,7 @@
*/
import "webrtc-adapter";
import { gstWebRTCAPI, start } from "./gstwebrtc-api.js";
import GstWebRTCAPI from "./gstwebrtc-api.js";
/**
* @external MediaStream
@ -49,9 +49,6 @@ import { gstWebRTCAPI, start } from "./gstwebrtc-api.js";
* @see https://developer.mozilla.org/en-US/docs/Web/API/HTMLVideoElement
*/
if (!window.gstWebRTCAPI) {
window.gstWebRTCAPI = gstWebRTCAPI;
window.addEventListener("DOMContentLoaded", () => {
start(window.gstWebRTCConfig);
});
if (!window.GstWebRTCAPI) {
window.GstWebRTCAPI = GstWebRTCAPI;
}

46
net/webrtc/gstwebrtc-api/src/producer-session.js
View file

@ -13,10 +13,10 @@ import WebRTCSession from "./webrtc-session.js";
import SessionState from "./session-state.js";
/**
* @class gstWebRTCAPI.ClientSession
* @class GstWebRTCAPI.ClientSession
* @hideconstructor
* @classdesc Client session representing a link between a remote consumer and a local producer session.
* @extends {gstWebRTCAPI.WebRTCSession}
* @extends {GstWebRTCAPI.WebRTCSession}
*/
class ClientSession extends WebRTCSession {
constructor(peerId, sessionId, comChannel, stream) {
@ -102,33 +102,33 @@ class ClientSession extends WebRTCSession {
/**
* Event name: "clientConsumerAdded".<br>
* Triggered when a remote consumer peer connects to a local {@link gstWebRTCAPI.ProducerSession}.
* @event gstWebRTCAPI#ClientConsumerAddedEvent
* Triggered when a remote consumer peer connects to a local {@link GstWebRTCAPI.ProducerSession}.
* @event GstWebRTCAPI#ClientConsumerAddedEvent
* @type {external:CustomEvent}
* @property {gstWebRTCAPI.ClientSession} detail - The WebRTC session associated with the added consumer peer.
* @see gstWebRTCAPI.ProducerSession
* @property {GstWebRTCAPI.ClientSession} detail - The WebRTC session associated with the added consumer peer.
* @see GstWebRTCAPI.ProducerSession
*/
/**
* Event name: "clientConsumerRemoved".<br>
* Triggered when a remote consumer peer disconnects from a local {@link gstWebRTCAPI.ProducerSession}.
* @event gstWebRTCAPI#ClientConsumerRemovedEvent
* Triggered when a remote consumer peer disconnects from a local {@link GstWebRTCAPI.ProducerSession}.
* @event GstWebRTCAPI#ClientConsumerRemovedEvent
* @type {external:CustomEvent}
* @property {gstWebRTCAPI.ClientSession} detail - The WebRTC session associated with the removed consumer peer.
* @see gstWebRTCAPI.ProducerSession
* @property {GstWebRTCAPI.ClientSession} detail - The WebRTC session associated with the removed consumer peer.
* @see GstWebRTCAPI.ProducerSession
*/
/**
* @class gstWebRTCAPI.ProducerSession
* @class GstWebRTCAPI.ProducerSession
* @hideconstructor
* @classdesc Producer session managing the streaming out of a local {@link external:MediaStream}.<br>
* It manages all underlying WebRTC connections to each peer client consuming the stream.
* <p>Call {@link gstWebRTCAPI#createProducerSession} to create a ProducerSession instance.</p>
* <p>Call {@link GstWebRTCAPI#createProducerSession} to create a ProducerSession instance.</p>
* @extends {external:EventTarget}
* @fires {@link gstWebRTCAPI#event:ErrorEvent}
* @fires {@link gstWebRTCAPI#event:StateChangedEvent}
* @fires {@link gstWebRTCAPI#event:ClosedEvent}
* @fires {@link gstWebRTCAPI#event:ClientConsumerAddedEvent}
* @fires {@link gstWebRTCAPI#event:ClientConsumerRemovedEvent}
* @fires {@link GstWebRTCAPI#event:ErrorEvent}
* @fires {@link GstWebRTCAPI#event:StateChangedEvent}
* @fires {@link GstWebRTCAPI#event:ClosedEvent}
* @fires {@link GstWebRTCAPI#event:ClientConsumerAddedEvent}
* @fires {@link GstWebRTCAPI#event:ClientConsumerRemovedEvent}
*/
export default class ProducerSession extends EventTarget {
constructor(comChannel, stream) {
@ -142,7 +142,7 @@ export default class ProducerSession extends EventTarget {
/**
* The local stream produced out by this session.
* @member {external:MediaStream} gstWebRTCAPI.ProducerSession#stream
* @member {external:MediaStream} GstWebRTCAPI.ProducerSession#stream
* @readonly
*/
get stream() {
@ -151,7 +151,7 @@ export default class ProducerSession extends EventTarget {
/**
* The current producer session state.
* @member {gstWebRTCAPI.SessionState} gstWebRTCAPI.ProducerSession#state
* @member {GstWebRTCAPI.SessionState} GstWebRTCAPI.ProducerSession#state
* @readonly
*/
get state() {
@ -163,10 +163,10 @@ export default class ProducerSession extends EventTarget {
* This method must be called after creating the producer session in order to start streaming. It registers this
* producer session to the signaling server and gets ready to serve peer requests from consumers.
* <p>Even on success, streaming can fail later if any error occurs during or after connection. In order to know
* the effective streaming state, you should be listening to the [error]{@link gstWebRTCAPI#event:ErrorEvent},
* [stateChanged]{@link gstWebRTCAPI#event:StateChangedEvent} and/or [closed]{@link gstWebRTCAPI#event:ClosedEvent}
* the effective streaming state, you should be listening to the [error]{@link GstWebRTCAPI#event:ErrorEvent},
* [stateChanged]{@link GstWebRTCAPI#event:StateChangedEvent} and/or [closed]{@link GstWebRTCAPI#event:ClosedEvent}
* events.</p>
* @method gstWebRTCAPI.ProducerSession#start
* @method GstWebRTCAPI.ProducerSession#start
* @returns {boolean} true in case of success (may fail later during or after connection) or false in case of
* immediate error (wrong session state or no connection to the signaling server).
*/
@ -203,7 +203,7 @@ export default class ProducerSession extends EventTarget {
* Terminates the producer session.<br>
* It immediately disconnects all peer consumers attached to this producer session and unregisters the producer
* from the signaling server.
* @method gstWebRTCAPI.ProducerSession#close
* @method GstWebRTCAPI.ProducerSession#close
*/
close() {
if (this._state !== SessionState.closed) {

22
net/webrtc/gstwebrtc-api/src/remote-controller.js
View file

@ -65,7 +65,7 @@ function getModifiers(event) {
}
/**
* @class gstWebRTCAPI.RemoteController
* @class GstWebRTCAPI.RemoteController
* @hideconstructor
* @classdesc Manages a specific WebRTC data channel created by a remote GStreamer webrtcsink producer and offering
* remote control of the producer through
@ -76,10 +76,10 @@ function getModifiers(event) {
* <p>You can attach an {@link external:HTMLVideoElement} to the remote controller, then all mouse and keyboard events
* emitted by this element will be automatically relayed to the remote producer.</p>
* @extends {external:EventTarget}
* @fires {@link gstWebRTCAPI#event:ErrorEvent}
* @fires {@link gstWebRTCAPI#event:ClosedEvent}
* @see gstWebRTCAPI.ConsumerSession#remoteController
* @see gstWebRTCAPI.RemoteController#attachVideoElement
* @fires {@link GstWebRTCAPI#event:ErrorEvent}
* @fires {@link GstWebRTCAPI#event:ClosedEvent}
* @see GstWebRTCAPI.ConsumerSession#remoteController
* @see GstWebRTCAPI.RemoteController#attachVideoElement
* @see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc/gstwebrtc-api#produce-a-gstreamer-interactive-webrtc-stream-with-remote-control
*/
export default class RemoteController extends EventTarget {
@ -114,7 +114,7 @@ export default class RemoteController extends EventTarget {
/**
* The underlying WebRTC data channel connected to a remote GStreamer webrtcsink producer offering remote control.
* The value may be null if the remote controller has been closed.
* @member {external:RTCDataChannel} gstWebRTCAPI.RemoteController#rtcDataChannel
* @member {external:RTCDataChannel} GstWebRTCAPI.RemoteController#rtcDataChannel
* @readonly
*/
get rtcDataChannel() {
@ -123,7 +123,7 @@ export default class RemoteController extends EventTarget {
/**
* The consumer session associated with this remote controller.
* @member {gstWebRTCAPI.ConsumerSession} gstWebRTCAPI.RemoteController#consumerSession
* @member {GstWebRTCAPI.ConsumerSession} GstWebRTCAPI.RemoteController#consumerSession
* @readonly
*/
get consumerSession() {
@ -133,9 +133,9 @@ export default class RemoteController extends EventTarget {
/**
* The video element that is currently used to send all mouse and keyboard events to the remote producer. Value may
* be null if no video element is attached.
* @member {external:HTMLVideoElement} gstWebRTCAPI.RemoteController#videoElement
* @member {external:HTMLVideoElement} GstWebRTCAPI.RemoteController#videoElement
* @readonly
* @see gstWebRTCAPI.RemoteController#attachVideoElement
* @see GstWebRTCAPI.RemoteController#attachVideoElement
*/
get videoElement() {
return this._videoElement;
@ -145,7 +145,7 @@ export default class RemoteController extends EventTarget {
* Associates a video element with this remote controller.<br>
* When a video element is attached to this remote controller, all mouse and keyboard events emitted by this
* element will be sent to the remote GStreamer webrtcink producer.
* @method gstWebRTCAPI.RemoteController#attachVideoElement
* @method GstWebRTCAPI.RemoteController#attachVideoElement
* @param {external:HTMLVideoElement|null} element - the video element to use to relay mouse and keyboard events,
* or null to detach any previously attached element. If the provided element parameter is not null and not a
* valid instance of an {@link external:HTMLVideoElement}, then the method does nothing.
@ -183,7 +183,7 @@ export default class RemoteController extends EventTarget {
* Closes the remote controller channel.<br>
* It immediately shuts down the underlying WebRTC data channel connected to a remote GStreamer webrtcsink
* producer and detaches any video element that may be used to relay mouse and keyboard events.
* @method gstWebRTCAPI.RemoteController#close
* @method GstWebRTCAPI.RemoteController#close
*/
close() {
this.attachVideoElement(null);

2
net/webrtc/gstwebrtc-api/src/session-state.js
View file

@ -12,7 +12,7 @@
/**
* Session states enumeration.<br>
* Session state always increases from idle to closed and never switches backwards.
* @typedef {enum} gstWebRTCAPI.SessionState
* @typedef {enum} GstWebRTCAPI.SessionState
* @readonly
* @property {0} idle - Default state when creating a new session, goes to <i>connecting</i> when starting
* the session.

54
net/webrtc/gstwebrtc-api/src/webrtc-session.js
View file

@ -15,53 +15,53 @@ import SessionState from "./session-state.js";
* Event name: "error".<br>
* Triggered when any kind of error occurs.
* <p>When emitted by a session, it is in general an unrecoverable error. Normally, the session is automatically closed
* but in the specific case of a {@link gstWebRTCAPI.ProducerSession}, when the error occurs on an underlying
* {@link gstWebRTCAPI.ClientSession} between the producer session and a remote client consuming the streamed media,
* then only the failing {@link gstWebRTCAPI.ClientSession} is closed. The producer session can keep on serving the
* but in the specific case of a {@link GstWebRTCAPI.ProducerSession}, when the error occurs on an underlying
* {@link GstWebRTCAPI.ClientSession} between the producer session and a remote client consuming the streamed media,
* then only the failing {@link GstWebRTCAPI.ClientSession} is closed. The producer session can keep on serving the
* other consumer peers.</p>
* @event gstWebRTCAPI#ErrorEvent
* @event GstWebRTCAPI#ErrorEvent
* @type {external:ErrorEvent}
* @property {string} message - The error message.
* @property {external:Error} error - The error exception.
* @see gstWebRTCAPI.WebRTCSession
* @see gstWebRTCAPI.RemoteController
* @see GstWebRTCAPI.WebRTCSession
* @see GstWebRTCAPI.RemoteController
*/
/**
* Event name: "stateChanged".<br>
* Triggered each time a session state changes.
* @event gstWebRTCAPI#StateChangedEvent
* @event GstWebRTCAPI#StateChangedEvent
* @type {external:Event}
* @see gstWebRTCAPI.WebRTCSession#state
* @see GstWebRTCAPI.WebRTCSession#state
*/
/**
* Event name: "rtcPeerConnectionChanged".<br>
* Triggered each time a session internal {@link external:RTCPeerConnection} changes. This can occur during the session
* connecting state when the peer-to-peer WebRTC connection is established, and when closing the
* {@link gstWebRTCAPI.WebRTCSession}.
* @event gstWebRTCAPI#RTCPeerConnectionChangedEvent
* {@link GstWebRTCAPI.WebRTCSession}.
* @event GstWebRTCAPI#RTCPeerConnectionChangedEvent
* @type {external:Event}
* @see gstWebRTCAPI.WebRTCSession#rtcPeerConnection
* @see GstWebRTCAPI.WebRTCSession#rtcPeerConnection
*/
/**
* Event name: "closed".<br>
* Triggered when a session is definitively closed (then it can be garbage collected as session instances are not
* reusable).
* @event gstWebRTCAPI#ClosedEvent
* @event GstWebRTCAPI#ClosedEvent
* @type {external:Event}
*/
/**
* @class gstWebRTCAPI.WebRTCSession
* @class GstWebRTCAPI.WebRTCSession
* @hideconstructor
* @classdesc Manages a WebRTC session between a producer and a consumer (peer-to-peer channel).
* @extends {external:EventTarget}
* @fires {@link gstWebRTCAPI#event:ErrorEvent}
* @fires {@link gstWebRTCAPI#event:StateChangedEvent}
* @fires {@link gstWebRTCAPI#event:RTCPeerConnectionChangedEvent}
* @fires {@link gstWebRTCAPI#event:ClosedEvent}
* @see gstWebRTCAPI.ConsumerSession
* @see gstWebRTCAPI.ProducerSession
* @see gstWebRTCAPI.ClientSession
* @fires {@link GstWebRTCAPI#event:ErrorEvent}
* @fires {@link GstWebRTCAPI#event:StateChangedEvent}
* @fires {@link GstWebRTCAPI#event:RTCPeerConnectionChangedEvent}
* @fires {@link GstWebRTCAPI#event:ClosedEvent}
* @see GstWebRTCAPI.ConsumerSession
* @see GstWebRTCAPI.ProducerSession
* @see GstWebRTCAPI.ClientSession
*/
export default class WebRTCSession extends EventTarget {
constructor(peerId, comChannel) {
@ -76,7 +76,7 @@ export default class WebRTCSession extends EventTarget {
/**
* Unique identifier of the remote peer to which this session is connected.
* @member {string} gstWebRTCAPI.WebRTCSession#peerId
* @member {string} GstWebRTCAPI.WebRTCSession#peerId
* @readonly
*/
get peerId() {
@ -87,8 +87,8 @@ export default class WebRTCSession extends EventTarget {
* Unique identifier of this session (defined by the signaling server).<br>
* The local session ID equals "" until it is created on server side. This is done during the connection handshake.
* The local session ID is guaranteed to be valid and to correctly reflect the signaling server value once
* session state has switched to {@link gstWebRTCAPI.SessionState#streaming}.
* @member {string} gstWebRTCAPI.WebRTCSession#sessionId
* session state has switched to {@link GstWebRTCAPI.SessionState#streaming}.
* @member {string} GstWebRTCAPI.WebRTCSession#sessionId
* @readonly
*/
get sessionId() {
@ -97,7 +97,7 @@ export default class WebRTCSession extends EventTarget {
/**
* The current WebRTC session state.
* @member {gstWebRTCAPI.SessionState} gstWebRTCAPI.WebRTCSession#state
* @member {GstWebRTCAPI.SessionState} GstWebRTCAPI.WebRTCSession#state
* @readonly
*/
get state() {
@ -107,9 +107,9 @@ export default class WebRTCSession extends EventTarget {
/**
* The internal {@link external:RTCPeerConnection} used to manage the underlying WebRTC connection with session
* peer. Value may be null if session has no active WebRTC connection. You can listen to the
* {@link gstWebRTCAPI#event:RTCPeerConnectionChangedEvent} event to be informed when the connection is established
* {@link GstWebRTCAPI#event:RTCPeerConnectionChangedEvent} event to be informed when the connection is established
* or destroyed.
* @member {external:RTCPeerConnection} gstWebRTCAPI.WebRTCSession#rtcPeerConnection
* @member {external:RTCPeerConnection} GstWebRTCAPI.WebRTCSession#rtcPeerConnection
* @readonly
*/
get rtcPeerConnection() {
@ -120,7 +120,7 @@ export default class WebRTCSession extends EventTarget {
* Terminates the WebRTC session.<br>
* It immediately disconnects the remote peer attached to this session and unregisters the session from the
* signaling server.
* @method gstWebRTCAPI.WebRTCSession#close
* @method GstWebRTCAPI.WebRTCSession#close
*/
close() {
if (this._state !== SessionState.closed) {