@ariaflowagents/livekit-plugin-transport-ws 0.9.0

package/dist/audio_input.d.ts

@@ -0,0 +1,25 @@
+import { AudioInput } from '@ariaflow/livekit-plugin';
+import type { WebSocket } from 'ws';
+/**
+ * Receives audio from a WebSocket connection and provides it as a
+ * ReadableStream<AudioFrame> to the AgentSession's STT pipeline.
+ *
+ * Binary WebSocket messages are raw PCM bytes. AudioByteStream accumulates
+ * bytes and emits AudioFrame objects at 100ms intervals.
+ */
+export declare class WebSocketAudioInput extends AudioInput {
+    private ws;
+    private byteStream;
+    private currentStreamId;
+    private streamWriter;
+    private closed;
+    constructor(ws: WebSocket, sampleRate?: number, numChannels?: number);
+    private setupListeners;
+    private handleAudioData;
+    private startNewStream;
+    private endCurrentStream;
+    /** Called when client sends 'end_of_audio'. Flushes remaining audio. */
+    endOfAudio(): void;
+    close(): Promise<void>;
+}
+//# sourceMappingURL=audio_input.d.ts.map

package/dist/audio_input.d.ts.map

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

@@ -0,0 +1,93 @@
+import { AudioInput, AudioByteStream } from '@ariaflow/livekit-plugin';
+import { TransformStream } from 'node:stream/web';
+/**
+ * Receives audio from a WebSocket connection and provides it as a
+ * ReadableStream<AudioFrame> to the AgentSession's STT pipeline.
+ *
+ * Binary WebSocket messages are raw PCM bytes. AudioByteStream accumulates
+ * bytes and emits AudioFrame objects at 100ms intervals.
+ */
+export class WebSocketAudioInput extends AudioInput {
+    ws;
+    byteStream;
+    currentStreamId = null;
+    streamWriter = null;
+    closed = false;
+    constructor(ws, sampleRate = 24000, numChannels = 1) {
+        super();
+        this.ws = ws;
+        this.byteStream = new AudioByteStream(sampleRate, numChannels);
+        this.setupListeners();
+    }
+    setupListeners() {
+        this.ws.on('message', (data, isBinary) => {
+            if (!isBinary || this.closed)
+                return;
+            this.handleAudioData(data);
+        });
+        this.ws.on('close', () => {
+            this.endCurrentStream();
+        });
+        this.ws.on('error', () => {
+            this.endCurrentStream();
+        });
+    }
+    handleAudioData(data) {
+        if (!this.currentStreamId) {
+            this.startNewStream();
+        }
+        const frames = this.byteStream.write(data.buffer.slice(data.byteOffset, data.byteOffset + data.byteLength));
+        for (const frame of frames) {
+            if (this.streamWriter) {
+                this.streamWriter.write(frame).catch((err) => {
+                    console.error('[WebSocketAudioInput] Error writing frame:', {
+                        error: err instanceof Error ? err.message : String(err),
+                        streamId: this.currentStreamId,
+                        timestamp: new Date().toISOString(),
+                    });
+                    this.endCurrentStream();
+                });
+            }
+        }
+    }
+    startNewStream() {
+        const { readable, writable } = new TransformStream();
+        this.streamWriter = writable.getWriter();
+        this.currentStreamId = this.multiStream.addInputStream(readable);
+    }
+    endCurrentStream() {
+        if (this.streamWriter) {
+            this.streamWriter.close().catch((err) => {
+                console.error('[WebSocketAudioInput] Error closing writer:', {
+                    error: err instanceof Error ? err.message : String(err),
+                    streamId: this.currentStreamId,
+                    timestamp: new Date().toISOString(),
+                });
+            });
+            this.streamWriter = null;
+        }
+        this.currentStreamId = null;
+    }
+    /** Called when client sends 'end_of_audio'. Flushes remaining audio. */
+    endOfAudio() {
+        const frames = this.byteStream.flush();
+        for (const frame of frames) {
+            if (this.streamWriter) {
+                this.streamWriter.write(frame).catch((err) => {
+                    console.error('[WebSocketAudioInput] Error writing flush frame:', {
+                        error: err instanceof Error ? err.message : String(err),
+                        streamId: this.currentStreamId,
+                        timestamp: new Date().toISOString(),
+                    });
+                });
+            }
+        }
+        this.endCurrentStream();
+    }
+    async close() {
+        this.closed = true;
+        this.endCurrentStream();
+        await super.close();
+    }
+}
+//# sourceMappingURL=audio_input.js.map

package/dist/audio_input.js.map

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package/dist/audio_output.d.ts

@@ -0,0 +1,26 @@
+import { AudioOutput, type AudioFrame } from '@ariaflow/livekit-plugin';
+import type { WebSocket } from 'ws';
+/**
+ * Sends audio frames from the TTS pipeline to the WebSocket client
+ * as binary messages containing raw PCM data.
+ */
+export declare class WebSocketAudioOutput extends AudioOutput {
+    private ws;
+    private contextLabel;
+    private sendQueue;
+    private sending;
+    private segmentSamplesSent;
+    private flushed;
+    private closed;
+    private segmentIndex;
+    private segmentStartedAt;
+    private frameCountInSegment;
+    constructor(ws: WebSocket, contextLabel?: string, sampleRate?: number);
+    captureFrame(frame: AudioFrame): Promise<void>;
+    private processSendQueue;
+    private drainAsync;
+    flush(): void;
+    clearBuffer(): void;
+    close(): Promise<void>;
+}
+//# sourceMappingURL=audio_output.d.ts.map

package/dist/audio_output.d.ts.map

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

@@ -0,0 +1,134 @@
+import { AudioOutput } from '@ariaflow/livekit-plugin';
+/**
+ * Sends audio frames from the TTS pipeline to the WebSocket client
+ * as binary messages containing raw PCM data.
+ */
+export class WebSocketAudioOutput extends AudioOutput {
+    ws;
+    contextLabel;
+    sendQueue = [];
+    sending = false;
+    segmentSamplesSent = 0;
+    flushed = false;
+    closed = false;
+    segmentIndex = 0;
+    segmentStartedAt = null;
+    frameCountInSegment = 0;
+    constructor(ws, contextLabel = 'unknown', sampleRate = 24000) {
+        super(sampleRate);
+        this.ws = ws;
+        this.contextLabel = contextLabel;
+    }
+    async captureFrame(frame) {
+        await super.captureFrame(frame);
+        if (this.closed)
+            return;
+        this.sendQueue.push(frame);
+        if (!this.sending) {
+            this.processSendQueue();
+        }
+    }
+    processSendQueue() {
+        if (this.sending || this.closed)
+            return;
+        this.sending = true;
+        this.drainAsync();
+    }
+    drainAsync() {
+        if (this.closed || this.sendQueue.length === 0) {
+            this.sending = false;
+            if (this.flushed) {
+                const playbackDuration = this.sampleRate
+                    ? this.segmentSamplesSent / this.sampleRate
+                    : 0;
+                this.onPlaybackFinished({
+                    playbackPosition: playbackDuration,
+                    interrupted: false,
+                });
+                this.segmentSamplesSent = 0;
+                this.flushed = false;
+            }
+            return;
+        }
+        const frame = this.sendQueue.shift();
+        if (this.segmentSamplesSent === 0) {
+            this.segmentIndex += 1;
+            this.segmentStartedAt = Date.now();
+            this.frameCountInSegment = 0;
+            this.onPlaybackStarted(Date.now());
+            console.info('[WebSocketAudioOutput] first frame enqueued', {
+                context: this.contextLabel,
+                segmentIndex: this.segmentIndex,
+                frameBytes: frame.data.byteLength,
+                timestamp: new Date().toISOString(),
+            });
+        }
+        const buffer = Buffer.from(frame.data.buffer, frame.data.byteOffset, frame.data.byteLength);
+        try {
+            this.ws.send(buffer, { binary: true });
+        }
+        catch {
+            this.sending = false;
+            return;
+        }
+        this.segmentSamplesSent += frame.samplesPerChannel;
+        this.frameCountInSegment += 1;
+        // Yield to the event loop between sends to avoid blocking
+        setImmediate(() => this.drainAsync());
+    }
+    flush() {
+        super.flush();
+        this.flushed = true;
+        if (this.sendQueue.length === 0) {
+            const playbackDuration = this.sampleRate
+                ? this.segmentSamplesSent / this.sampleRate
+                : 0;
+            this.onPlaybackFinished({
+                playbackPosition: playbackDuration,
+                interrupted: false,
+            });
+            this.segmentSamplesSent = 0;
+            this.flushed = false;
+            if (this.segmentStartedAt !== null) {
+                console.info('[WebSocketAudioOutput] segment flushed', {
+                    context: this.contextLabel,
+                    segmentIndex: this.segmentIndex,
+                    frames: this.frameCountInSegment,
+                    durationMs: Date.now() - this.segmentStartedAt,
+                    playbackDurationMs: Math.round(playbackDuration * 1000),
+                    timestamp: new Date().toISOString(),
+                });
+            }
+            this.segmentStartedAt = null;
+            this.frameCountInSegment = 0;
+        }
+    }
+    clearBuffer() {
+        const playbackDuration = this.sampleRate
+            ? this.segmentSamplesSent / this.sampleRate
+            : 0;
+        this.sendQueue = [];
+        this.onPlaybackFinished({
+            playbackPosition: playbackDuration,
+            interrupted: true,
+        });
+        if (this.segmentStartedAt !== null) {
+            console.warn('[WebSocketAudioOutput] segment interrupted', {
+                context: this.contextLabel,
+                segmentIndex: this.segmentIndex,
+                frames: this.frameCountInSegment,
+                durationMs: Date.now() - this.segmentStartedAt,
+                timestamp: new Date().toISOString(),
+            });
+        }
+        this.segmentSamplesSent = 0;
+        this.flushed = false;
+        this.segmentStartedAt = null;
+        this.frameCountInSegment = 0;
+    }
+    async close() {
+        this.closed = true;
+        this.sendQueue = [];
+    }
+}
+//# sourceMappingURL=audio_output.js.map

package/dist/audio_output.js.map

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

@@ -0,0 +1,11 @@
+export { WebSocketAgentServer } from './server.js';
+export type { NativeSessionOptions } from './server.js';
+export type { WebSocketServerOptions } from './types.js';
+export { WebSocketTransportAdapter } from './transport_adapter.js';
+export { WebSocketAudioInput } from './audio_input.js';
+export { WebSocketAudioOutput } from './audio_output.js';
+export { WebSocketTextOutput } from './text_output.js';
+export { bridgeWebSocketToRealtimeTransport, bridgeAdapterToRealtimeTransport, float32ToInt16Bytes, int16BytesToFloat32, } from './realtime_bridge.js';
+export type { ClientMessage, ServerMessage, ConfigureMessage, UserTextMessage, EndOfAudioMessage, SessionStartedMessage, AgentTextMessage, UserTranscriptionMessage, AgentStateMessage, ErrorMessage, SessionEndedMessage, } from './protocol.js';
+export { parseClientMessage, serializeServerMessage } from './protocol.js';
+//# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

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

@@ -0,0 +1,8 @@
+export { WebSocketAgentServer } from './server.js';
+export { WebSocketTransportAdapter } from './transport_adapter.js';
+export { WebSocketAudioInput } from './audio_input.js';
+export { WebSocketAudioOutput } from './audio_output.js';
+export { WebSocketTextOutput } from './text_output.js';
+export { bridgeWebSocketToRealtimeTransport, bridgeAdapterToRealtimeTransport, float32ToInt16Bytes, int16BytesToFloat32, } from './realtime_bridge.js';
+export { parseClientMessage, serializeServerMessage } from './protocol.js';
+//# sourceMappingURL=index.js.map

package/dist/index.js.map

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

@@ -0,0 +1,65 @@
+export interface ConfigureMessage {
+    type: 'configure';
+    sampleRate?: number;
+    numChannels?: number;
+    encoding?: 'pcm_s16le' | 'mulaw' | 'alaw';
+}
+export interface UserTextMessage {
+    type: 'user_text';
+    text: string;
+}
+export interface EndOfAudioMessage {
+    type: 'end_of_audio';
+}
+export interface SessionStartedMessage {
+    type: 'session_started';
+    sessionId: string;
+    config: {
+        sampleRate: number;
+        numChannels: number;
+        encoding: string;
+    };
+}
+export interface AgentTextMessage {
+    type: 'agent_text';
+    text: string;
+    isFinal: boolean;
+}
+/**
+ * @experimental Not currently emitted by the runtime. STT transcription events
+ * flow through the LiveKit AgentSession internal pipeline and are not surfaced
+ * to the WS client. Reserved for future observability needs.
+ */
+export interface UserTranscriptionMessage {
+    type: 'user_transcription';
+    text: string;
+    isFinal: boolean;
+}
+/**
+ * @experimental Not currently emitted by the runtime. State machine transitions
+ * are implicit inside the LiveKit AgentSession. Reserved for future UI state
+ * indicator support.
+ */
+export interface AgentStateMessage {
+    type: 'agent_state';
+    state: 'initializing' | 'idle' | 'listening' | 'thinking' | 'speaking';
+}
+export interface ErrorMessage {
+    type: 'error';
+    message: string;
+    code?: string;
+}
+/**
+ * @experimental Not currently emitted by the runtime. Session end is communicated
+ * via the WebSocket close frame. The client discovers session end when the WS
+ * connection closes. Reserved for explicit lifecycle signaling if needed.
+ */
+export interface SessionEndedMessage {
+    type: 'session_ended';
+    reason: 'completed' | 'error' | 'client_disconnect' | 'server_shutdown';
+}
+export type ClientMessage = ConfigureMessage | UserTextMessage | EndOfAudioMessage;
+export type ServerMessage = SessionStartedMessage | AgentTextMessage | UserTranscriptionMessage | AgentStateMessage | ErrorMessage | SessionEndedMessage;
+export declare function parseClientMessage(data: string): ClientMessage | null;
+export declare function serializeServerMessage(msg: ServerMessage): string;
+//# sourceMappingURL=protocol.d.ts.map

package/dist/protocol.d.ts.map

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

@@ -0,0 +1,44 @@
+// --- Client -> Server Messages ---
+function isFinitePositiveNumber(value) {
+    return typeof value === 'number' && Number.isFinite(value) && value > 0;
+}
+function isValidEncoding(value) {
+    return value === 'pcm_s16le' || value === 'mulaw' || value === 'alaw';
+}
+export function parseClientMessage(data) {
+    try {
+        const msg = JSON.parse(data);
+        if (!msg || typeof msg.type !== 'string')
+            return null;
+        switch (msg.type) {
+            case 'configure': {
+                if (msg.sampleRate !== undefined && !isFinitePositiveNumber(msg.sampleRate)) {
+                    return null;
+                }
+                if (msg.numChannels !== undefined && !isFinitePositiveNumber(msg.numChannels)) {
+                    return null;
+                }
+                if (msg.encoding !== undefined && !isValidEncoding(msg.encoding)) {
+                    return null;
+                }
+                return msg;
+            }
+            case 'user_text':
+                if (typeof msg.text !== 'string') {
+                    return null;
+                }
+                return msg;
+            case 'end_of_audio':
+                return msg;
+            default:
+                return null;
+        }
+    }
+    catch {
+        return null;
+    }
+}
+export function serializeServerMessage(msg) {
+    return JSON.stringify(msg);
+}
+//# sourceMappingURL=protocol.js.map

package/dist/protocol.js.map

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

@@ -0,0 +1,43 @@
+import type { WebSocket } from 'ws';
+import type { RealtimeTransportSession } from '@ariaflowagents/core/realtime';
+import type { WebSocketTransportAdapter } from './transport_adapter.js';
+/**
+ * Convert float32 PCM samples to int16 PCM bytes (Uint8Array).
+ * GeminiLiveSession and RealtimeRuntime expect int16 PCM.
+ * LiveKit AudioFrame uses float32 samples internally.
+ */
+export declare function float32ToInt16Bytes(float32: Float32Array): Uint8Array;
+/**
+ * Convert int16 PCM bytes (Uint8Array) to float32 PCM samples.
+ */
+export declare function int16BytesToFloat32(pcmBytes: Uint8Array): Float32Array;
+/**
+ * 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 declare function bridgeWebSocketToRealtimeTransport(ws: WebSocket, options?: {
+    sessionId?: string;
+}): RealtimeTransportSession;
+/**
+ * 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 declare function bridgeAdapterToRealtimeTransport(adapter: WebSocketTransportAdapter): RealtimeTransportSession;
+//# sourceMappingURL=realtime_bridge.d.ts.map

package/dist/realtime_bridge.d.ts.map

@@ -0,0 +1 @@
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