@auxilium/datalynk-client 1.1.6 → 1.1.8
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/README.md +17 -11
- package/dist/index.js +33 -21
- package/dist/index.mjs +33 -21
- package/dist/webrtc.d.ts +15 -12
- package/dist/webrtc.d.ts.map +1 -1
- package/package.json +1 -1
package/README.md
CHANGED
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@@ -411,18 +411,24 @@ Connect to a WebRTC chatroom to send audio/video to peers
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411
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```js
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// Video elements to display stream on
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414
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-
const local = document.querySelector('#video1');
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-
const remote = document.querySelector('#
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416
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-
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const roomID = 'TEST_ROOM'
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418
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const session = await api.webrtc.connect('test', (peer) => { // This callback fires everytime a new peer connects
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419
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// Set remote audio/video stream to first track once availible
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420
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-
const streams = peer.stream.getRemoteStreams();
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421
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-
if(streams[0]) remoteVideo.srcObject = streams[0];
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422
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-
else peer.stream.ontrack = ({streams}) => remoteVideo.srcObject = streams[0];
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-
}, true, false); // audio = true, video = false
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-
local.srcObject = session.stream; // Set local audio/video stream
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const local = document.querySelector('#video1'); // <video id="video1" autoplay muted playsinline></video>
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const remote = document.querySelector('#video2'); // <video id="video2" autoplay playsinline></video>
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// Connect to a WebRTC room
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const roomID = 'TEST_ROOM';
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const session = await api.webrtc.connect(roomID, true, true); // audio = true, video = true
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+
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// Set local audio/video stream
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local.srcObject = session.stream;
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+
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// Track is null on the initial connection, will fire again when the stream is ready
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session.onConnected = (peer, stream) => {
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// Set remote audio/video stream if ready
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if(stream) remote.srcObject = stream;
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}
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+
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// Disconnect from WebRTC room
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session.disconnect();
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```
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</details>
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package/dist/index.js
CHANGED
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@@ -3245,20 +3245,23 @@ class Superuser {
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} });
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}
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}
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-
const version = "1.1.
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+
const version = "1.1.8";
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class WebRtc {
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constructor(api) {
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__publicField(this, "ice");
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-
var _a, _b, _c;
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+
var _a, _b, _c, _d;
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this.api = api;
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-
this.ice =
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urls: [(_a = this.api.options.webrtc) == null ? void 0 : _a.url],
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-
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this.ice = [
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{ urls: ["stun:" + ((_a = this.api.options.webrtc) == null ? void 0 : _a.url)] },
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clean({
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urls: ["turn:" + ((_b = this.api.options.webrtc) == null ? void 0 : _b.url)],
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username: (_c = this.api.options.webrtc) == null ? void 0 : _c.username,
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credential: (_d = this.api.options.webrtc) == null ? void 0 : _d.password
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})
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];
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}
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async answer(offer, stream) {
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-
const rtc = new RTCPeerConnection({ iceServers:
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+
const rtc = new RTCPeerConnection({ iceServers: this.ice });
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stream.getTracks().forEach((track) => rtc.addTrack(track, stream));
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await rtc.setRemoteDescription(new RTCSessionDescription(offer));
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const answer = await rtc.createAnswer();
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@@ -3272,7 +3275,7 @@ class WebRtc {
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return rtc;
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}
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async offer(stream) {
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-
const rtc = new RTCPeerConnection({ iceServers:
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const rtc = new RTCPeerConnection({ iceServers: this.ice });
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stream.getTracks().forEach((track) => rtc.addTrack(track, stream));
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await rtc.setLocalDescription(await rtc.createOffer());
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await new Promise((res) => {
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@@ -3286,19 +3289,28 @@ class WebRtc {
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/**
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* Create a new WebRTC connection to a room
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* @param {string} id Room ID
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-
* @param {(peer) => any} onConnect Callback when peers connect
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* @param {boolean} audio Stream local audio to room/peers
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* @param {boolean} video Stream local video to room/peers
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* @returns {Promise<WebRTCSession>} Connection session
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*/
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-
async connect(id = randomStringBuilder(16, true, true),
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async connect(id = randomStringBuilder(16, true, true), audio = true, video = true) {
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const session = {
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open: true,
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id,
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uid: randomStringBuilder(8, true, true),
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peers: {},
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stream: await navigator.mediaDevices.getUserMedia({ audio, video }),
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-
disconnect: null
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disconnect: null,
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onConnected: null
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};
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const handleTracks = (session2, uid) => {
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if (!session2.open || !session2.onConnected) return;
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const peer = session2.peers[uid];
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const streams = peer.connection.getRemoteStreams();
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if (streams[0]) session2.onConnected(peer, streams[0]);
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else peer.connection.ontrack = ({ streams: streams2 }) => {
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if (session2.open && session2.onConnected) session2.onConnected(peer, streams2[0]);
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};
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};
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const unsubscribe = this.api.socket.addListener(async (event) => {
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var _a, _b, _c, _d;
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@@ -3309,34 +3321,34 @@ class WebRtc {
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session.peers[payload.uid] = {
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uid: payload.uid,
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username: payload.username,
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-
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+
connection: await this.offer(session.stream)
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};
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this.api.socket.send({ [`webrtc/${id}`]: { offer: {
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to: payload.uid,
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uid: session.uid,
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username: (_a = this.api.auth.user) == null ? void 0 : _a.login,
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-
offer: session.peers[payload.uid].
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+
offer: session.peers[payload.uid].connection.localDescription
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} } });
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} else if (event[key].disconnected) {
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-
session.peers[event[key].disconnected.uid].
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session.peers[event[key].disconnected.uid].connection.close();
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delete session.peers[event[key].disconnected.uid];
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} else if (event[key].offer && event[key].offer.to == session.uid) {
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const payload = event[key].offer;
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session.peers[payload.uid] = {
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uid: payload.uid,
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username: payload.username,
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-
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connection: await this.answer(payload.offer, session.stream)
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};
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this.api.socket.send({ [`webrtc/${id}`]: { answer: {
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to: payload.uid,
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uid: session.uid,
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username: (_b = this.api.auth.user) == null ? void 0 : _b.login,
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-
answer: session.peers[payload.uid].
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+
answer: session.peers[payload.uid].connection.localDescription
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} } });
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-
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+
handleTracks(session, payload.uid);
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} else if (event[key].answer && event[key].answer.to == session.uid) {
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-
(_d = (_c = session.peers[event[key].answer.uid]) == null ? void 0 : _c.
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3339
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-
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+
(_d = (_c = session.peers[event[key].answer.uid]) == null ? void 0 : _c.connection) == null ? void 0 : _d.setRemoteDescription(new RTCSessionDescription(event[key].answer.answer));
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+
handleTracks(session, event[key].answer.uid);
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}
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}
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}, () => {
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@@ -3410,7 +3422,7 @@ const _Api = class _Api {
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saveSession: true,
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serviceWorker: "/service.worker.mjs",
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webrtc: {
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-
url: "
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+
url: "scarborough.auxilium.world:3478",
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username: "auxilium-webrtc",
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password: "UU3qkD95Tg54eF"
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},
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package/dist/index.mjs
CHANGED
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@@ -3243,20 +3243,23 @@ class Superuser {
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3243
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} });
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3244
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}
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3245
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}
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3246
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-
const version = "1.1.
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3246
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+
const version = "1.1.8";
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3247
3247
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class WebRtc {
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constructor(api) {
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3249
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__publicField(this, "ice");
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3250
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-
var _a, _b, _c;
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+
var _a, _b, _c, _d;
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3251
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this.api = api;
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3252
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-
this.ice =
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3253
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-
urls: [(_a = this.api.options.webrtc) == null ? void 0 : _a.url],
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3254
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-
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3255
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-
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3256
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-
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3252
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+
this.ice = [
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3253
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+
{ urls: ["stun:" + ((_a = this.api.options.webrtc) == null ? void 0 : _a.url)] },
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3254
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+
clean({
|
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3255
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+
urls: ["turn:" + ((_b = this.api.options.webrtc) == null ? void 0 : _b.url)],
|
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3256
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+
username: (_c = this.api.options.webrtc) == null ? void 0 : _c.username,
|
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3257
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+
credential: (_d = this.api.options.webrtc) == null ? void 0 : _d.password
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3258
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+
})
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3259
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+
];
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3257
3260
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}
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3258
3261
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async answer(offer, stream) {
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|
3259
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-
const rtc = new RTCPeerConnection({ iceServers:
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3262
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+
const rtc = new RTCPeerConnection({ iceServers: this.ice });
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3260
3263
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stream.getTracks().forEach((track) => rtc.addTrack(track, stream));
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3261
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await rtc.setRemoteDescription(new RTCSessionDescription(offer));
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3262
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const answer = await rtc.createAnswer();
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|
@@ -3270,7 +3273,7 @@ class WebRtc {
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|
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3270
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return rtc;
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3271
3274
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}
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3272
3275
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async offer(stream) {
|
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3273
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-
const rtc = new RTCPeerConnection({ iceServers:
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|
3276
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+
const rtc = new RTCPeerConnection({ iceServers: this.ice });
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3274
3277
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stream.getTracks().forEach((track) => rtc.addTrack(track, stream));
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3275
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await rtc.setLocalDescription(await rtc.createOffer());
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3276
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await new Promise((res) => {
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|
@@ -3284,19 +3287,28 @@ class WebRtc {
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|
|
3284
3287
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/**
|
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3285
3288
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* Create a new WebRTC connection to a room
|
|
3286
3289
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* @param {string} id Room ID
|
|
3287
|
-
* @param {(peer) => any} onConnect Callback when peers connect
|
|
3288
3290
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* @param {boolean} audio Stream local audio to room/peers
|
|
3289
3291
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* @param {boolean} video Stream local video to room/peers
|
|
3290
3292
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* @returns {Promise<WebRTCSession>} Connection session
|
|
3291
3293
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*/
|
|
3292
|
-
async connect(id = randomStringBuilder(16, true, true),
|
|
3294
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+
async connect(id = randomStringBuilder(16, true, true), audio = true, video = true) {
|
|
3293
3295
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const session = {
|
|
3294
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open: true,
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3295
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id,
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uid: randomStringBuilder(8, true, true),
|
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3297
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peers: {},
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3298
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stream: await navigator.mediaDevices.getUserMedia({ audio, video }),
|
|
3299
|
-
disconnect: null
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3301
|
+
disconnect: null,
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3302
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+
onConnected: null
|
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3303
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+
};
|
|
3304
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+
const handleTracks = (session2, uid) => {
|
|
3305
|
+
if (!session2.open || !session2.onConnected) return;
|
|
3306
|
+
const peer = session2.peers[uid];
|
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3307
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+
const streams = peer.connection.getRemoteStreams();
|
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3308
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+
if (streams[0]) session2.onConnected(peer, streams[0]);
|
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3309
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+
else peer.connection.ontrack = ({ streams: streams2 }) => {
|
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3310
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+
if (session2.open && session2.onConnected) session2.onConnected(peer, streams2[0]);
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3311
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+
};
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3300
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};
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3301
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const unsubscribe = this.api.socket.addListener(async (event) => {
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var _a, _b, _c, _d;
|
|
@@ -3307,34 +3319,34 @@ class WebRtc {
|
|
|
3307
3319
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session.peers[payload.uid] = {
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|
3308
3320
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uid: payload.uid,
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3321
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username: payload.username,
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|
3310
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-
|
|
3322
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+
connection: await this.offer(session.stream)
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3311
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};
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|
3312
3324
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this.api.socket.send({ [`webrtc/${id}`]: { offer: {
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3313
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to: payload.uid,
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3314
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uid: session.uid,
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3315
3327
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username: (_a = this.api.auth.user) == null ? void 0 : _a.login,
|
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3316
|
-
offer: session.peers[payload.uid].
|
|
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+
offer: session.peers[payload.uid].connection.localDescription
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3317
3329
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} } });
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|
3318
3330
|
} else if (event[key].disconnected) {
|
|
3319
|
-
session.peers[event[key].disconnected.uid].
|
|
3331
|
+
session.peers[event[key].disconnected.uid].connection.close();
|
|
3320
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|
delete session.peers[event[key].disconnected.uid];
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3321
3333
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} else if (event[key].offer && event[key].offer.to == session.uid) {
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3322
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const payload = event[key].offer;
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3323
3335
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session.peers[payload.uid] = {
|
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3336
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uid: payload.uid,
|
|
3325
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username: payload.username,
|
|
3326
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-
|
|
3338
|
+
connection: await this.answer(payload.offer, session.stream)
|
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3327
3339
|
};
|
|
3328
3340
|
this.api.socket.send({ [`webrtc/${id}`]: { answer: {
|
|
3329
3341
|
to: payload.uid,
|
|
3330
3342
|
uid: session.uid,
|
|
3331
3343
|
username: (_b = this.api.auth.user) == null ? void 0 : _b.login,
|
|
3332
|
-
answer: session.peers[payload.uid].
|
|
3344
|
+
answer: session.peers[payload.uid].connection.localDescription
|
|
3333
3345
|
} } });
|
|
3334
|
-
|
|
3346
|
+
handleTracks(session, payload.uid);
|
|
3335
3347
|
} else if (event[key].answer && event[key].answer.to == session.uid) {
|
|
3336
|
-
(_d = (_c = session.peers[event[key].answer.uid]) == null ? void 0 : _c.
|
|
3337
|
-
|
|
3348
|
+
(_d = (_c = session.peers[event[key].answer.uid]) == null ? void 0 : _c.connection) == null ? void 0 : _d.setRemoteDescription(new RTCSessionDescription(event[key].answer.answer));
|
|
3349
|
+
handleTracks(session, event[key].answer.uid);
|
|
3338
3350
|
}
|
|
3339
3351
|
}
|
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3340
3352
|
}, () => {
|
|
@@ -3408,7 +3420,7 @@ const _Api = class _Api {
|
|
|
3408
3420
|
saveSession: true,
|
|
3409
3421
|
serviceWorker: "/service.worker.mjs",
|
|
3410
3422
|
webrtc: {
|
|
3411
|
-
url: "
|
|
3423
|
+
url: "scarborough.auxilium.world:3478",
|
|
3412
3424
|
username: "auxilium-webrtc",
|
|
3413
3425
|
password: "UU3qkD95Tg54eF"
|
|
3414
3426
|
},
|
package/dist/webrtc.d.ts
CHANGED
|
@@ -1,5 +1,14 @@
|
|
|
1
1
|
import { Unsubscribe } from '@ztimson/utils';
|
|
2
2
|
import { Api } from './api';
|
|
3
|
+
/** WebRTC peer */
|
|
4
|
+
export type WebRTCPeer = {
|
|
5
|
+
/** Peer connection UID */
|
|
6
|
+
uid: string;
|
|
7
|
+
/** Peer username */
|
|
8
|
+
username: string;
|
|
9
|
+
/** Remote connection to stream */
|
|
10
|
+
connection: RTCPeerConnection;
|
|
11
|
+
};
|
|
3
12
|
/** WebRTC session */
|
|
4
13
|
export type WebRTCSession = {
|
|
5
14
|
/** Is the connection active? */
|
|
@@ -10,19 +19,14 @@ export type WebRTCSession = {
|
|
|
10
19
|
uid: string;
|
|
11
20
|
/** Local media stream */
|
|
12
21
|
stream: MediaStream;
|
|
13
|
-
/** Connected peers */
|
|
22
|
+
/** Connected peers sorted by UID */
|
|
14
23
|
peers: {
|
|
15
|
-
[key: string]:
|
|
16
|
-
/** Peer connection UID */
|
|
17
|
-
uid: string;
|
|
18
|
-
/** Peer username */
|
|
19
|
-
username: string;
|
|
20
|
-
/** Remote connection to stream */
|
|
21
|
-
stream: RTCPeerConnection;
|
|
22
|
-
};
|
|
24
|
+
[key: string]: WebRTCPeer;
|
|
23
25
|
};
|
|
24
26
|
/** Disconnection session */
|
|
25
27
|
disconnect: Unsubscribe;
|
|
28
|
+
/** Callback when peers connect */
|
|
29
|
+
onConnected?: (peer: WebRTCPeer, track: any) => any;
|
|
26
30
|
};
|
|
27
31
|
export declare class WebRtc {
|
|
28
32
|
private api;
|
|
@@ -30,18 +34,17 @@ export declare class WebRtc {
|
|
|
30
34
|
urls: string[];
|
|
31
35
|
username?: string;
|
|
32
36
|
credential?: string;
|
|
33
|
-
};
|
|
37
|
+
}[];
|
|
34
38
|
constructor(api: Api);
|
|
35
39
|
private answer;
|
|
36
40
|
private offer;
|
|
37
41
|
/**
|
|
38
42
|
* Create a new WebRTC connection to a room
|
|
39
43
|
* @param {string} id Room ID
|
|
40
|
-
* @param {(peer) => any} onConnect Callback when peers connect
|
|
41
44
|
* @param {boolean} audio Stream local audio to room/peers
|
|
42
45
|
* @param {boolean} video Stream local video to room/peers
|
|
43
46
|
* @returns {Promise<WebRTCSession>} Connection session
|
|
44
47
|
*/
|
|
45
|
-
connect(id?: string,
|
|
48
|
+
connect(id?: string, audio?: boolean, video?: boolean): Promise<WebRTCSession>;
|
|
46
49
|
}
|
|
47
50
|
//# sourceMappingURL=webrtc.d.ts.map
|
package/dist/webrtc.d.ts.map
CHANGED
|
@@ -1 +1 @@
|
|
|
1
|
-
{"version":3,"file":"webrtc.d.ts","sourceRoot":"","sources":["../src/webrtc.ts"],"names":[],"mappings":"AAAA,OAAO,EAA6B,WAAW,EAAC,MAAM,gBAAgB,CAAC;AACvE,OAAO,EAAC,GAAG,EAAC,MAAM,OAAO,CAAC;AAE1B,qBAAqB;AACrB,MAAM,MAAM,aAAa,GAAG;IAC3B,gCAAgC;IAChC,IAAI,EAAE,OAAO,CAAC;IACd,cAAc;IACd,EAAE,EAAE,MAAM,CAAC;IACX,qBAAqB;IACrB,GAAG,EAAE,MAAM,CAAC;IACZ,yBAAyB;IACzB,MAAM,EAAE,WAAW,CAAC;IACpB,
|
|
1
|
+
{"version":3,"file":"webrtc.d.ts","sourceRoot":"","sources":["../src/webrtc.ts"],"names":[],"mappings":"AAAA,OAAO,EAA6B,WAAW,EAAC,MAAM,gBAAgB,CAAC;AACvE,OAAO,EAAC,GAAG,EAAC,MAAM,OAAO,CAAC;AAE1B,kBAAkB;AAClB,MAAM,MAAM,UAAU,GAAG;IACxB,0BAA0B;IAC1B,GAAG,EAAE,MAAM,CAAC;IACZ,oBAAoB;IACpB,QAAQ,EAAE,MAAM,CAAC;IACjB,kCAAkC;IAClC,UAAU,EAAE,iBAAiB,CAAA;CAC7B,CAAA;AAED,qBAAqB;AACrB,MAAM,MAAM,aAAa,GAAG;IAC3B,gCAAgC;IAChC,IAAI,EAAE,OAAO,CAAC;IACd,cAAc;IACd,EAAE,EAAE,MAAM,CAAC;IACX,qBAAqB;IACrB,GAAG,EAAE,MAAM,CAAC;IACZ,yBAAyB;IACzB,MAAM,EAAE,WAAW,CAAC;IACpB,oCAAoC;IACpC,KAAK,EAAE;QAAC,CAAC,GAAG,EAAE,MAAM,GAAG,UAAU,CAAA;KAAC,CAAA;IAClC,4BAA4B;IAC5B,UAAU,EAAE,WAAW,CAAA;IACvB,kCAAkC;IAClC,WAAW,CAAC,EAAE,CAAC,IAAI,EAAE,UAAU,EAAE,KAAK,EAAE,GAAG,KAAK,GAAG,CAAA;CACnD,CAAA;AAED,qBAAa,MAAM;IAGN,OAAO,CAAC,GAAG;IAFvB,SAAgB,GAAG,EAAG;QAAC,IAAI,EAAE,MAAM,EAAE,CAAC;QAAC,QAAQ,CAAC,EAAE,MAAM,CAAC;QAAC,UAAU,CAAC,EAAE,MAAM,CAAA;KAAC,EAAE,CAAC;gBAE7D,GAAG,EAAE,GAAG;YAWd,MAAM;YAaN,KAAK;IAWnB;;;;;;OAMG;IACG,OAAO,CAAC,EAAE,GAAE,MAA4C,EAAE,KAAK,UAAO,EAAE,KAAK,UAAO,GAAG,OAAO,CAAC,aAAa,CAAC;CAoEnH"}
|
package/package.json
CHANGED
|
@@ -2,7 +2,7 @@
|
|
|
2
2
|
"name": "@auxilium/datalynk-client",
|
|
3
3
|
"description": "Datalynk client library",
|
|
4
4
|
"repository": "https://gitlab.auxiliumgroup.com/auxilium/datalynk/datalynk-client",
|
|
5
|
-
"version": "1.1.
|
|
5
|
+
"version": "1.1.8",
|
|
6
6
|
"author": "Zak Timson <zaktimson@gmail.com>",
|
|
7
7
|
"private": false,
|
|
8
8
|
"main": "./dist/index.cjs",
|