Improve sdk with v2

.gitignore

@@ -8,3 +8,7 @@ node_modules
 /coverage/
 /tsbuild
 /types
+/.vscode
+/packages/webrtc-sdk/docs/*
+/packages/webrtc-sdk/dist
+/packages/webrtc-sdk/docs

package.json

@@ -9,9 +9,10 @@
     "doc": "doc"
   },
   "scripts": {
-    "compile": "npm run cleanup:tsbuild && npm run compile:js && npm run compile:ts && copy-files-from-to && npm run cleanup:tsbuild",
+    "compile": "npm run cleanup:tsbuild && npm run compile:js && npm run compile:ts && npm run compile:webrtc-sdk && copy-files-from-to && npm run cleanup:tsbuild",
     "compile:js": "rollup -c rollup.config.module.cjs && rollup -c rollup.config.browser.cjs ",
     "compile:ts": "tsc -p ./tsconfig.json && api-extractor run",
+    "compile:webrtc-sdk": "npm install --prefix packages/webrtc-sdk && npm run build --prefix packages/webrtc-sdk",
     "cleanup:tsbuild": "rimraf ./tsbuild",
     "test": "karma start karma.conf.cjs",
     "codecov": "codecov"
@@ -28,6 +29,38 @@
     {
       "from": "./dist/es/*",
       "to": "./src/main/webapp/js/"
+    },
+    {
+      "from": "./packages/webrtc-sdk/dist/**/*",
+      "to": "./src/main/webapp/v2/js/"
+    },
+    {
+      "from": "./packages/webrtc-sdk/examples/*",
+      "to": "./src/main/webapp/v2/"
+    },
+    {
+      "from": "./src/main/webapp/js/external/**/*",
+      "to": "./src/main/webapp/v2/js/external/"
+    },
+    {
+      "from": "./src/main/webapp/js/external/**/*",
+      "to": "./src/main/webapp/v2/js/external/"
+    },
+    {
+      "from": "./src/main/webapp/js/fetch.stream.js",
+      "to": "./src/main/webapp/v2/js/"
+    },
+    {
+      "from": "./src/main/webapp/js/utility.js",
+      "to": "./src/main/webapp/v2/js/"
+    },
+    {
+      "from": "./src/main/webapp/js/external/**/*",
+      "to": "./src/main/webapp/v2/js/external/"
+    },
+    {
+      "from": "./src/main/webapp/css/**/*",
+      "to": "./src/main/webapp/v2/css/"
     }
   ],
   "copyFilesSettings": {

packages/webrtc-sdk/.prettierrc.json

@@ -0,0 +1,11 @@
+{
+  "semi": true,
+  "trailingComma": "es5",
+  "singleQuote": false,
+  "printWidth": 100,
+  "tabWidth": 2,
+  "useTabs": false,
+  "bracketSpacing": true,
+  "arrowParens": "avoid",
+  "endOfLine": "lf"
+}

packages/webrtc-sdk/README.md

@@ -0,0 +1,229 @@
+# WebRTC SDK v2 (TypeScript)
+
+Modern, strictly-typed client SDK for Ant Media Server.
+
+## Install
+
+This package is currently private for development. Build locally:
+
+```bash
+npm install
+npm run build
+```
+
+## Quick start
+
+```ts
+import { WebRTCClient, getWebSocketURL } from 'webrtc-sdk';
+
+// One-liner session helper (recommended for most apps)
+const { client } = await WebRTCClient.createSession({
+  websocketURL: getWebSocketURL('wss://example.com:5443/LiveApp/websocket'),
+  role: 'publisher',
+  streamId: 'stream1',
+  localVideo: document.getElementById('local') as HTMLVideoElement,
+  remoteVideo: document.getElementById('remote') as HTMLVideoElement,
+  mediaConstraints: { audio: true, video: true },
+  autoPlay: true, // attempt autoplay on the remote element after join
+});
+
+client.on('play_started', ({ streamId }) => console.log('playing', streamId));
+```
+
+
+
+### Common operations
+
+```ts
+// Stop a session
+await client.stop('stream1');
+
+// Send text over data channel
+await client.sendData('stream1', 'hello');
+
+// One-off stats snapshot
+const stats = await client.getStats('stream1');
+```
+
+## Events
+
+### Events quick reference
+
+Common events emitted by the SDK (see TypeDoc for full list):
+
+- `initialized`: signaling is ready
+- `publish_started` / `publish_finished`
+- `play_started` / `play_finished`
+- `newTrackAvailable` { stream, track, streamId }
+- `ice_connection_state_changed` { state, streamId }
+- `data_channel_opened` / `data_channel_closed`
+- `data_received` { streamId, data: string | ArrayBuffer }
+- `updated_stats` PeerStats
+- `devices_updated` GroupedDevices
+- `room_joined` / `room_left`
+- `room_information`, `broadcast_object`, `subscriber_count`, `subscriber_list`
+- `video_track_assignments`
+- `reconnection_attempt_for_publisher` / `reconnection_attempt_for_player`
+- `error` { error, message? }
+
+### Listening to events (v2)
+
+```ts
+client.on('initialized', () => console.log('ready'));
+client.on('publish_started', ({ streamId }) => console.log('publishing', streamId));
+client.on('play_started', ({ streamId }) => console.log('playing', streamId));
+client.on('data_received', ({ streamId, data }) => console.log('dc <-', streamId, data));
+client.on('ice_connection_state_changed', ({ state, streamId }) => console.log('ice', state, streamId));
+client.on('reconnected', ({ streamId }) => console.log('reconnected', streamId));
+
+// AMS server notifications exposed as typed events and also under notification:<name>
+client.on('broadcast_object', (obj) => console.log('broadcast_object', obj));
+client.on('notification:subscriberCount', (payload) => console.log('subscriberCount', payload));
+```
+
+## Documentation
+
+Generated API docs are available in the `docs/` folder. To regenerate:
+
+```bash
+npm run docs
+```
+
+Open `docs/index.html` in a browser.
+
+### Architecture and usage guidance
+
+`WebRTCClient` is the primary API surface. It composes:
+
+- `WebSocketAdaptor`: handles signaling with Ant Media Server (WS commands, notifications).
+- `MediaManager`: handles local media (getUserMedia, device switching, screen share).
+
+For most applications, call methods on `WebRTCClient` only. It exposes the common
+operations you need: `join()`, `publish()`, `play()`, `stop()`, `listDevices()`,
+`selectVideoInput()`, `selectAudioInput()`, `startScreenShare()`, `stopScreenShare()`,
+`sendData()`, `enableStats()/disableStats()`, room/multitrack helpers, and emits typed events.
+
+Note: Methods internally wait for signaling readiness; you don't need to call `ready()` yourself.
+
+Only use `WebSocketAdaptor` or `MediaManager` directly if you have advanced
+customization needs (e.g., custom signaling transport or bespoke media capture).
+Otherwise, prefer the higher-level `WebRTCClient` methods.
+
+### Room / Multitrack quick start
+
+```ts
+// Join a room
+await client.joinRoom({ roomId: 'my-room', streamId: 'publisher1' });
+
+// Selectively play only some subtracks of a main stream
+await client.playSelective({
+  streamId: 'mainStreamId',
+  enableTracks: ['camera_user1', 'screen_user2'],
+  disableTracksByDefault: true,
+});
+
+// Enable/disable a specific subtrack
+client.enableTrack('mainStreamId', 'camera_user3', true);
+
+// Force quality (ABR)
+client.forceStreamQuality('mainStreamId', 720); // or 'auto'
+```
+
+### Device management and screen share
+
+```ts
+// Switch devices without renegotiation
+await client.selectVideoInput('camera-device-id');
+await client.selectAudioInput('mic-device-id');
+
+// Camera on/off keeps sender alive (black dummy track)
+await client.turnOffLocalCamera();
+await client.turnOnLocalCamera();
+
+// Screen share and overlay (PIP camera)
+await client.startScreenShare();
+await client.stopScreenShare();
+await client.startScreenWithCameraOverlay();
+await client.stopScreenWithCameraOverlay();
+```
+
+### Data channel
+
+```ts
+// Text
+await client.sendData('s1', 'hello');
+// Binary
+await client.sendData('s1', new Uint8Array([1,2,3]).buffer);
+// Optional sanitize received strings
+client.setSanitizeDataChannelStrings(true);
+client.on('data_received', ({ data }) => console.log('rx', data));
+```
+
+## Feature examples
+
+### Reconnect configuration
+
+```ts
+client.configureReconnect({ backoff: "exp", baseMs: 500, maxMs: 10000, jitter: 0.3 });
+client.on("reconnected", ({ streamId }) => console.log("reconnected", streamId));
+```
+
+### Device hot-swap and track controls
+
+```ts
+client.on("device_hotswapped", e => console.log("hotswapped", e));
+
+// Pause/resume tracks without renegotiation
+client.pauseTrack("audio");
+client.resumeTrack("audio");
+```
+
+### Remote audio level (viewer)
+
+```ts
+await client.enableRemoteAudioLevel("s1", level => console.log(level), 200);
+client.disableRemoteAudioLevel("s1");
+```
+
+### Data-only publish
+
+```ts
+const { client } = await WebRTCClient.createSession({
+  websocketURL: getWebSocketURL('wss://example.com:5443/LiveApp/websocket'),
+  role: 'publisher',
+  streamId: 'data-only',
+  onlyDataChannel: true,
+});
+await client.sendJSON('data-only', { hello: 'world' });
+```
+
+### Stats helpers
+
+```ts
+// One-off snapshot and event
+const stats = await client.getStats('s1');
+client.on('updated_stats', (ps) => console.log(ps));
+
+// Poll every 2s
+client.enableStats('s1', 2000);
+```
+
+## Troubleshooting
+
+- Autoplay blocked: browsers may block autoplay with sound. Use `autoPlay: true` and call `videoEl.play()` on a user gesture if needed.
+- HTTPS required: `getUserMedia` and `getDisplayMedia` need HTTPS (or localhost). Serve examples over HTTPS.
+- TURN/ICE: if connections fail across networks, configure proper TURN servers on Ant Media Server and pass `peerConfig.iceServers` if needed.
+
+## Examples
+
+- `examples/publish.html`
+- `examples/play.html`
+- `examples/room.html` (rooms/multitrack, enable/disable subtracks, force quality, selective play)
+
+## Development
+
+- Lint: `npm run lint` (ESLint)
+- Format: Prettier (integrated; run `npm run lint:fix` for quick fixes)
+- Tests: `npm test`
+- Docs: `npm run docs`
+

packages/webrtc-sdk/documents/Cookbook.md

@@ -0,0 +1,83 @@
+## WebRTC SDK v2 Cookbook
+
+### 1) Publish/Play with one-liner
+```ts
+const { client } = await WebRTCClient.createSession({ websocketURL, role: 'publisher', streamId: 's1', localVideo, remoteVideo, mediaConstraints: { audio: true, video: true }, autoPlay: true });
+```
+
+### 1b) Event listeners
+```ts
+client.on('publish_started', ({ streamId }) => console.log('publishing', streamId));
+client.on('data_received', ({ data }) => console.log('dc <-', data));
+
+```
+
+### 2) Device switching
+```ts
+await client.selectVideoInput(cameraId);
+await client.selectAudioInput(micId);
+```
+
+### 3) Screen share and PIP overlay
+```ts
+await client.startScreenShare();
+// later
+await client.stopScreenShare();
+
+await client.startScreenWithCameraOverlay();
+await client.stopScreenWithCameraOverlay();
+```
+
+### 4) Data channel helpers
+```ts
+await client.sendData('s1', 'hello');
+await client.sendJSON('s1', { type: 'chat', text: 'hi' });
+```
+
+### 5) Reconnect policy and events
+```ts
+client.configureReconnect({ backoff: 'exp', baseMs: 500, maxMs: 10000, jitter: 0.3 });
+client.on('reconnected', ({ streamId }) => console.log('reconnected', streamId));
+```
+
+### 6) Room / multitrack helpers
+```ts
+await client.joinRoom({ roomId: 'room1', streamId: 'pub1' });
+await client.playSelective({ streamId: 'main', enableTracks: ['camera_u1'], disableTracksByDefault: true });
+client.enableTrack('main', 'camera_u1', true);
+client.forceStreamQuality('main', 720); // or 'auto'
+```
+
+### 7) Bandwidth/quality
+```ts
+await client.changeBandwidth('s1', 600);
+await client.changeBandwidth('s1', 'unlimited');
+await client.setDegradationPreference('s1', 'maintain-framerate');
+```
+
+### 8) Stats
+```ts
+client.on('updated_stats', ps => console.log(ps));
+client.enableStats('s1', 2000);
+```
+
+### 9) Audio output (sinkId) and meters
+```ts
+await client.setAudioOutput(deviceId, remoteVideo);
+await client.enableRemoteAudioLevel('s1', level => console.log(level));
+```
+
+### 10) Track controls
+```ts
+client.pauseTrack('audio');
+client.resumeTrack('audio');
+```
+
+### 11) Data-only publish
+```ts
+const { client } = await WebRTCClient.createSession({ websocketURL, role: 'publisher', streamId: 'data', onlyDataChannel: true });
+await client.sendJSON('data', { ping: true });
+```
+
+
+