@ariaflowagents/livekit-plugin-transport-ws 0.9.0

package/dist/realtime_bridge.js

@@ -0,0 +1,192 @@
+// ─── PCM Conversion Utilities ──────────────────────────────────────────────
+/**
+ * Convert float32 PCM samples to int16 PCM bytes (Uint8Array).
+ * GeminiLiveSession and RealtimeRuntime expect int16 PCM.
+ * LiveKit AudioFrame uses float32 samples internally.
+ */
+export function float32ToInt16Bytes(float32) {
+    const int16 = new Int16Array(float32.length);
+    for (let i = 0; i < float32.length; i++) {
+        const clamped = Math.max(-1, Math.min(1, float32[i] ?? 0));
+        int16[i] = Math.round(clamped * 0x7fff);
+    }
+    return new Uint8Array(int16.buffer);
+}
+/**
+ * Convert int16 PCM bytes (Uint8Array) to float32 PCM samples.
+ */
+export function int16BytesToFloat32(pcmBytes) {
+    const int16View = new Int16Array(pcmBytes.buffer, pcmBytes.byteOffset, Math.floor(pcmBytes.byteLength / 2));
+    const float32 = new Float32Array(int16View.length);
+    for (let i = 0; i < int16View.length; i++) {
+        float32[i] = (int16View[i] ?? 0) / 0x7fff;
+    }
+    return float32;
+}
+// ─── Raw WebSocket Bridge ──────────────────────────────────────────────────
+/**
+ * Bridge a raw WebSocket connection to a RealtimeTransportSession.
+ *
+ * This is the primary bridge used by WebSocketAgentServer.startNativeSession().
+ * Binary WS messages are passed through directly as Uint8Array (int16 PCM)
+ * without intermediate float32 conversion.
+ *
+ * The WebSocket client and the model client must agree on sample rate and
+ * encoding. No resampling is performed in this bridge.
+ */
+export function bridgeWebSocketToRealtimeTransport(ws, options) {
+    const audioHandlers = [];
+    const closeHandlers = [];
+    let closed = false;
+    function fireClose() {
+        if (closed)
+            return;
+        closed = true;
+        for (const handler of closeHandlers) {
+            handler();
+        }
+    }
+    // Named handlers so they can be removed on close()
+    const onMessage = (data, isBinary) => {
+        if (!isBinary || closed)
+            return;
+        const uint8 = new Uint8Array(data.buffer, data.byteOffset, data.byteLength);
+        for (const handler of audioHandlers) {
+            handler(uint8);
+        }
+    };
+    const onClose = () => fireClose();
+    const onError = () => fireClose();
+    ws.on('message', onMessage);
+    ws.on('close', onClose);
+    ws.on('error', onError);
+    function removeListeners() {
+        ws.removeListener('message', onMessage);
+        ws.removeListener('close', onClose);
+        ws.removeListener('error', onError);
+    }
+    const interruptHandlers = [];
+    return {
+        sendAudio(data) {
+            if (closed || ws.readyState !== ws.OPEN)
+                return;
+            try {
+                ws.send(Buffer.from(data), { binary: true });
+            }
+            catch {
+                // WS already closed
+            }
+        },
+        onAudio(handler) {
+            audioHandlers.push(handler);
+        },
+        onClose(handler) {
+            closeHandlers.push(handler);
+        },
+        onInterrupted(handler) {
+            interruptHandlers.push(handler);
+        },
+        clearAudioBuffer() {
+            // Signal the client to stop playback.
+            // Send a JSON control message that the frontend can act on.
+            if (closed || ws.readyState !== ws.OPEN)
+                return;
+            try {
+                ws.send(JSON.stringify({ type: 'clear_audio' }));
+            }
+            catch {
+                // WS closed
+            }
+            for (const handler of interruptHandlers) {
+                handler();
+            }
+        },
+        close() {
+            if (closed)
+                return;
+            closed = true;
+            removeListeners();
+            if (ws.readyState === ws.OPEN) {
+                ws.close(1000, 'Session ended');
+            }
+        },
+    };
+}
+// ─── Adapter-Based Bridge ──────────────────────────────────────────────────
+/**
+ * Bridge a WebSocketTransportAdapter to a RealtimeTransportSession.
+ *
+ * This variant works at the adapter level, converting between LiveKit's
+ * float32 AudioFrame format and RealtimeRuntime's int16 PCM Uint8Array.
+ *
+ * Use this when you already have a WebSocketTransportAdapter instance
+ * (e.g., from the server's onConnection callback) and want to route it
+ * through RealtimeRuntime instead of AriaFlowVoiceSession.
+ *
+ * Note: This bridge subscribes to the raw WS binary messages directly
+ * (same as the simple bridge) because WebSocketAudioInput doesn't expose
+ * a frame event. Output is sent as raw binary on the WS, bypassing
+ * WebSocketAudioOutput's AudioFrame machinery.
+ */
+export function bridgeAdapterToRealtimeTransport(adapter) {
+    const audioHandlers = [];
+    const closeHandlers = [];
+    let closed = false;
+    // Use the public rawSocket getter to access the underlying WebSocket.
+    // The adapter's audioInput already listens on the WS for binary messages
+    // and converts to AudioFrames. For the realtime bridge, we need the raw
+    // int16 PCM bytes. We attach a second binary listener on the same WS.
+    const ws = adapter.rawSocket;
+    function fireClose() {
+        if (closed)
+            return;
+        closed = true;
+        for (const handler of closeHandlers) {
+            handler();
+        }
+    }
+    const onMessage = (data, isBinary) => {
+        if (!isBinary || closed)
+            return;
+        const uint8 = new Uint8Array(data.buffer, data.byteOffset, data.byteLength);
+        for (const handler of audioHandlers) {
+            handler(uint8);
+        }
+    };
+    const onClose = () => fireClose();
+    const onError = () => fireClose();
+    ws.on('message', onMessage);
+    ws.on('close', onClose);
+    ws.on('error', onError);
+    function removeListeners() {
+        ws.removeListener('message', onMessage);
+        ws.removeListener('close', onClose);
+        ws.removeListener('error', onError);
+    }
+    return {
+        sendAudio(data) {
+            if (closed || !adapter.isOpen)
+                return;
+            try {
+                ws.send(data, { binary: true });
+            }
+            catch {
+                // WS closed
+            }
+        },
+        onAudio(handler) {
+            audioHandlers.push(handler);
+        },
+        onClose(handler) {
+            closeHandlers.push(handler);
+        },
+        close() {
+            if (closed)
+                return;
+            closed = true;
+            removeListeners();
+            adapter.close().catch(() => { });
+        },
+    };
+}
+//# sourceMappingURL=realtime_bridge.js.map

package/dist/realtime_bridge.js.map

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package/dist/server.d.ts

@@ -0,0 +1,73 @@
+import { SessionManager, type AriaFlowVoiceSession } from '@ariaflow/livekit-plugin';
+import type { voice } from '@livekit/agents';
+import type { RealtimeRuntime, RealtimeAudioClient, RealtimeSessionHandle } from '@ariaflowagents/core/realtime';
+import { WebSocketTransportAdapter } from './transport_adapter.js';
+import type { WebSocketServerOptions } from './types.js';
+/**
+ * Options for starting a native audio session through RealtimeRuntime.
+ */
+export interface NativeSessionOptions {
+    /** The RealtimeRuntime instance to route audio through. */
+    runtime: RealtimeRuntime;
+    /** Factory that creates a RealtimeAudioClient for each session. */
+    createModelClient: () => RealtimeAudioClient;
+    /** Optional session ID (generated if not provided). */
+    sessionId?: string;
+    /** Optional user ID for session scoping. */
+    userId?: string;
+    /** Optional agent ID override. */
+    agentId?: string;
+}
+/**
+ * A WebSocket server that accepts connections and creates agent sessions.
+ *
+ * Usage:
+ *   const server = new WebSocketAgentServer({ port: 8080 });
+ *
+ *   server.onConnection(async (transport) => {
+ *     const voiceSession = new AriaFlowVoiceSession({
+ *       runtime: runtime,
+ *       stt: new GeminiLiveSTT(),
+ *       tts: new GeminiLiveTTS(),
+ *     });
+ *     await server.startSession(transport, voiceSession);
+ *   });
+ *
+ *   await server.listen();
+ */
+export declare class WebSocketAgentServer {
+    private options;
+    private wss;
+    private sessionManager;
+    private nativeSessions;
+    private connectionHandler;
+    constructor(options?: WebSocketServerOptions);
+    onConnection(handler: (adapter: WebSocketTransportAdapter) => void | Promise<void>): void;
+    startSession(adapter: WebSocketTransportAdapter, voiceSession: AriaFlowVoiceSession): Promise<voice.AgentSession>;
+    /**
+     * Start a native audio session that routes audio directly through
+     * RealtimeRuntime instead of the cascaded STT→LLM→TTS pipeline.
+     *
+     * This uses the WS-to-RealtimeTransport bridge to convert the raw
+     * WebSocket connection into a RealtimeTransportSession, then starts
+     * a session via RealtimeRuntime.startSession().
+     *
+     * The model client handles STT, reasoning, and TTS in a single
+     * persistent connection (e.g., Gemini Live, OpenAI Realtime).
+     *
+     * @example
+     * ```typescript
+     * server.onConnection(async (transport) => {
+     *   await server.startNativeSession(transport, {
+     *     runtime: realtimeRuntime,
+     *     createModelClient: () => new GeminiLiveSession({ apiKey, model }),
+     *   });
+     * });
+     * ```
+     */
+    startNativeSession(adapter: WebSocketTransportAdapter, options: NativeSessionOptions): Promise<RealtimeSessionHandle>;
+    listen(): Promise<void>;
+    get sessions(): SessionManager;
+    close(): Promise<void>;
+}
+//# sourceMappingURL=server.d.ts.map

package/dist/server.d.ts.map

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package/dist/server.js

@@ -0,0 +1,247 @@
+import { WebSocketServer as WSServer } from 'ws';
+import { SessionManager } from '@ariaflow/livekit-plugin';
+import { WebSocketTransportAdapter } from './transport_adapter.js';
+import { bridgeWebSocketToRealtimeTransport } from './realtime_bridge.js';
+import { parseClientMessage, serializeServerMessage } from './protocol.js';
+/**
+ * A WebSocket server that accepts connections and creates agent sessions.
+ *
+ * Usage:
+ *   const server = new WebSocketAgentServer({ port: 8080 });
+ *
+ *   server.onConnection(async (transport) => {
+ *     const voiceSession = new AriaFlowVoiceSession({
+ *       runtime: runtime,
+ *       stt: new GeminiLiveSTT(),
+ *       tts: new GeminiLiveTTS(),
+ *     });
+ *     await server.startSession(transport, voiceSession);
+ *   });
+ *
+ *   await server.listen();
+ */
+export class WebSocketAgentServer {
+    options;
+    wss = null;
+    sessionManager = new SessionManager();
+    nativeSessions = new Map();
+    connectionHandler = null;
+    constructor(options = {}) {
+        this.options = options;
+    }
+    onConnection(handler) {
+        this.connectionHandler = handler;
+    }
+    async startSession(adapter, voiceSession) {
+        return this.sessionManager.startSession(adapter, voiceSession);
+    }
+    /**
+     * Start a native audio session that routes audio directly through
+     * RealtimeRuntime instead of the cascaded STT→LLM→TTS pipeline.
+     *
+     * This uses the WS-to-RealtimeTransport bridge to convert the raw
+     * WebSocket connection into a RealtimeTransportSession, then starts
+     * a session via RealtimeRuntime.startSession().
+     *
+     * The model client handles STT, reasoning, and TTS in a single
+     * persistent connection (e.g., Gemini Live, OpenAI Realtime).
+     *
+     * @example
+     * ```typescript
+     * server.onConnection(async (transport) => {
+     *   await server.startNativeSession(transport, {
+     *     runtime: realtimeRuntime,
+     *     createModelClient: () => new GeminiLiveSession({ apiKey, model }),
+     *   });
+     * });
+     * ```
+     */
+    async startNativeSession(adapter, options) {
+        const ws = adapter.rawSocket;
+        const realtimeTransport = bridgeWebSocketToRealtimeTransport(ws, {
+            sessionId: adapter.id,
+        });
+        const modelClient = options.createModelClient();
+        const handle = await options.runtime.startSession({
+            modelClient,
+            transport: realtimeTransport,
+            sessionId: options.sessionId ?? adapter.id,
+            userId: options.userId,
+            agentId: options.agentId,
+        });
+        this.nativeSessions.set(adapter.id, handle);
+        // Wire cleanup on WS close
+        ws.on('close', () => {
+            const nativeHandle = this.nativeSessions.get(adapter.id);
+            if (nativeHandle) {
+                this.nativeSessions.delete(adapter.id);
+                nativeHandle.stop().catch((err) => {
+                    console.error('[WebSocketServer] native session stop error:', {
+                        error: err instanceof Error ? err.message : String(err),
+                        adapterId: adapter.id,
+                        timestamp: new Date().toISOString(),
+                    });
+                });
+            }
+        });
+        console.info('[WebSocketServer] native session started', {
+            adapterId: adapter.id,
+            sessionId: handle.sessionId,
+            callId: handle.callId,
+            timestamp: new Date().toISOString(),
+        });
+        return handle;
+    }
+    async listen() {
+        const port = this.options.port ?? 8080;
+        const host = this.options.host ?? '0.0.0.0';
+        this.wss = new WSServer({ port, host });
+        this.wss.on('connection', async (ws, req) => {
+            const connectedAt = Date.now();
+            let binaryMessageCount = 0;
+            let firstBinaryAt = null;
+            // Authentication
+            if (this.options.authenticate) {
+                const allowed = await this.options.authenticate(req);
+                if (!allowed) {
+                    ws.close(4001, 'Unauthorized');
+                    return;
+                }
+            }
+            const sampleRate = this.options.defaultSampleRate ?? 24000;
+            const numChannels = this.options.defaultNumChannels ?? 1;
+            const adapter = new WebSocketTransportAdapter(ws, {
+                sampleRate,
+                numChannels,
+            });
+            console.info('[WebSocketServer] client connected', {
+                adapterId: adapter.id,
+                remoteAddress: req.socket.remoteAddress,
+                timestamp: new Date().toISOString(),
+            });
+            // Send session started
+            ws.send(serializeServerMessage({
+                type: 'session_started',
+                sessionId: adapter.id,
+                config: {
+                    sampleRate,
+                    numChannels,
+                    encoding: 'pcm_s16le',
+                },
+            }));
+            console.info('[WebSocketServer] session_started sent', {
+                adapterId: adapter.id,
+                sampleRate,
+                numChannels,
+                timestamp: new Date().toISOString(),
+            });
+            // Handle text and control messages
+            ws.on('message', (data, isBinary) => {
+                if (isBinary) {
+                    binaryMessageCount += 1;
+                    if (!firstBinaryAt) {
+                        firstBinaryAt = Date.now();
+                        console.info('[WebSocketServer] first binary audio packet received', {
+                            adapterId: adapter.id,
+                            bytes: data.byteLength,
+                            msSinceConnect: firstBinaryAt - connectedAt,
+                            timestamp: new Date().toISOString(),
+                        });
+                    }
+                    else if (binaryMessageCount % 100 === 0) {
+                        console.debug('[WebSocketServer] binary audio packet progress', {
+                            adapterId: adapter.id,
+                            count: binaryMessageCount,
+                            bytes: data.byteLength,
+                            timestamp: new Date().toISOString(),
+                        });
+                    }
+                    return;
+                }
+                const msg = parseClientMessage(data.toString());
+                if (!msg) {
+                    console.warn('[WebSocketServer] unrecognized client message', {
+                        adapterId: adapter.id,
+                        payloadPreview: data.toString().slice(0, 120),
+                        timestamp: new Date().toISOString(),
+                    });
+                    return;
+                }
+                if (msg.type === 'end_of_audio') {
+                    console.info('[WebSocketServer] end_of_audio received', {
+                        adapterId: adapter.id,
+                        binaryMessageCount,
+                        timestamp: new Date().toISOString(),
+                    });
+                    adapter.audioInput.endOfAudio();
+                }
+                else if (msg.type === 'user_text') {
+                    console.info('[WebSocketServer] user_text received', {
+                        adapterId: adapter.id,
+                        chars: msg.text.length,
+                        timestamp: new Date().toISOString(),
+                    });
+                    const session = this.sessionManager.getSession(adapter.id);
+                    if (session) {
+                        try {
+                            const handle = session.generateReply({ userInput: msg.text });
+                            handle.addDoneCallback(() => {
+                                // Intentionally noop: callback is used to attach completion lifecycle
+                                // in one place for future telemetry hooks.
+                            });
+                        }
+                        catch (err) {
+                            console.error('[WebSocketServer] generateReply error:', {
+                                error: err instanceof Error ? err.message : String(err),
+                                adapterId: adapter.id,
+                                timestamp: new Date().toISOString(),
+                            });
+                        }
+                    }
+                }
+            });
+            // Handle disconnect
+            ws.on('close', (code, reasonBuffer) => {
+                const disconnectedAt = Date.now();
+                console.info('[WebSocketServer] client disconnected', {
+                    adapterId: adapter.id,
+                    code,
+                    reason: reasonBuffer.toString(),
+                    binaryMessageCount,
+                    sessionDurationMs: disconnectedAt - connectedAt,
+                    timestamp: new Date().toISOString(),
+                });
+                this.sessionManager.closeSession(adapter.id).catch((err) => {
+                    console.error('[WebSocketServer] Error closing session:', {
+                        error: err instanceof Error ? err.message : String(err),
+                        adapterId: adapter.id,
+                        timestamp: new Date().toISOString(),
+                    });
+                });
+            });
+            if (this.connectionHandler) {
+                await this.connectionHandler(adapter);
+            }
+        });
+        return new Promise((resolve) => {
+            this.wss.on('listening', () => {
+                resolve();
+            });
+        });
+    }
+    get sessions() {
+        return this.sessionManager;
+    }
+    async close() {
+        // Stop all native sessions
+        const nativeStops = Array.from(this.nativeSessions.values()).map((handle) => handle.stop().catch(() => { }));
+        await Promise.allSettled(nativeStops);
+        this.nativeSessions.clear();
+        // Close all cascaded sessions
+        await this.sessionManager.closeAll();
+        if (this.wss) {
+            this.wss.close();
+        }
+    }
+}
+//# sourceMappingURL=server.js.map

package/dist/server.js.map

@@ -0,0 +1 @@
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package/dist/text_output.d.ts

@@ -0,0 +1,17 @@
+import { TextOutput, type TimedString } from '@ariaflow/livekit-plugin';
+import type { WebSocket } from 'ws';
+/**
+ * Sends agent transcription text to the WebSocket client as JSON messages.
+ */
+export declare class WebSocketTextOutput extends TextOutput {
+    private ws;
+    private contextLabel;
+    private closed;
+    private segmentIndex;
+    private segmentStartedAt;
+    constructor(ws: WebSocket, contextLabel?: string, nextInChain?: TextOutput);
+    captureText(text: string | TimedString): Promise<void>;
+    flush(): void;
+    close(): Promise<void>;
+}
+//# sourceMappingURL=text_output.d.ts.map

package/dist/text_output.d.ts.map

@@ -0,0 +1 @@
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package/dist/text_output.js

@@ -0,0 +1,77 @@
+import { TextOutput, isTimedString } from '@ariaflow/livekit-plugin';
+import { serializeServerMessage } from './protocol.js';
+/**
+ * Sends agent transcription text to the WebSocket client as JSON messages.
+ */
+export class WebSocketTextOutput extends TextOutput {
+    ws;
+    contextLabel;
+    closed = false;
+    segmentIndex = 0;
+    segmentStartedAt = null;
+    constructor(ws, contextLabel = 'unknown', nextInChain) {
+        super(nextInChain);
+        this.ws = ws;
+        this.contextLabel = contextLabel;
+    }
+    async captureText(text) {
+        if (this.closed)
+            return;
+        const textContent = isTimedString(text) ? text.text : text;
+        if (textContent.length > 0 && this.segmentStartedAt === null) {
+            this.segmentIndex += 1;
+            this.segmentStartedAt = Date.now();
+            console.info('[WebSocketTextOutput] first chunk', {
+                context: this.contextLabel,
+                segmentIndex: this.segmentIndex,
+                chars: textContent.length,
+                timestamp: new Date().toISOString(),
+            });
+        }
+        const msg = {
+            type: 'agent_text',
+            text: textContent,
+            isFinal: false,
+        };
+        try {
+            this.ws.send(serializeServerMessage(msg));
+        }
+        catch {
+            // WebSocket closed
+        }
+        if (this.nextInChain) {
+            await this.nextInChain.captureText(text);
+        }
+    }
+    flush() {
+        if (!this.closed) {
+            if (this.segmentStartedAt !== null) {
+                console.info('[WebSocketTextOutput] segment flushed', {
+                    context: this.contextLabel,
+                    segmentIndex: this.segmentIndex,
+                    durationMs: Date.now() - this.segmentStartedAt,
+                    timestamp: new Date().toISOString(),
+                });
+            }
+            this.segmentStartedAt = null;
+            const msg = {
+                type: 'agent_text',
+                text: '',
+                isFinal: true,
+            };
+            try {
+                this.ws.send(serializeServerMessage(msg));
+            }
+            catch {
+                // WebSocket closed
+            }
+        }
+        if (this.nextInChain) {
+            this.nextInChain.flush();
+        }
+    }
+    async close() {
+        this.closed = true;
+    }
+}
+//# sourceMappingURL=text_output.js.map

package/dist/text_output.js.map

@@ -0,0 +1 @@
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