voice-agent-ai-sdk 0.1.0

package/LICENSE

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+MIT License
+
+Copyright (c) 2026 Bijit Mondal
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

package/README.md

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+# voice-agent-ai-sdk
+
+[![npm version](https://badge.fury.io/js/voice-agent-ai-sdk.svg)](https://www.npmjs.com/package/voice-agent-ai-sdk)
+
+Streaming voice/text agent SDK built on [AI SDK](https://sdk.vercel.ai/) with optional WebSocket transport.
+
+## Features
+
+- **Streaming text generation** via AI SDK `streamText` with multi-step tool calling.
+- **Chunked streaming TTS** — text is split at sentence boundaries and converted to speech in parallel as the LLM streams, giving low time-to-first-audio.
+- **Audio transcription** via AI SDK `experimental_transcribe` (e.g. Whisper).
+- **Barge-in / interruption** — user speech cancels both the in-flight LLM stream and pending TTS, saving tokens and latency.
+- **Memory management** — configurable sliding-window on conversation history (`maxMessages`, `maxTotalChars`) and audio input size limits.
+- **Serial request queue** — concurrent `sendText` / audio inputs are queued and processed one at a time, preventing race conditions.
+- **Graceful lifecycle** — `disconnect()` aborts all in-flight work; `destroy()` permanently releases every resource.
+- **WebSocket transport** with a full protocol of stream, tool, and speech lifecycle events.
+- **Works without WebSocket** — call `sendText()` directly for text-only or server-side use.
+
+## Prerequisites
+
+- Node.js 20+
+- pnpm
+- OpenAI API key
+
+## Setup
+
+1. Install dependencies:
+
+   pnpm install
+
+2. Configure environment variables in `.env`:
+
+   OPENAI_API_KEY=your_openai_api_key
+   VOICE_WS_ENDPOINT=ws://localhost:8080
+
+`VOICE_WS_ENDPOINT` is optional for text-only usage.
+
+## VoiceAgent usage (as in the demo)
+
+Minimal end-to-end example using AI SDK tools, streaming text, and streaming TTS:
+
+```ts
+import "dotenv/config";
+import { VoiceAgent } from "./src";
+import { tool } from "ai";
+import { z } from "zod";
+import { openai } from "@ai-sdk/openai";
+
+const weatherTool = tool({
+   description: "Get the weather in a location",
+   inputSchema: z.object({ location: z.string() }),
+   execute: async ({ location }) => ({ location, temperature: 72, conditions: "sunny" }),
+});
+
+const agent = new VoiceAgent({
+   model: openai("gpt-4o"),
+   transcriptionModel: openai.transcription("whisper-1"),
+   speechModel: openai.speech("gpt-4o-mini-tts"),
+   instructions: "You are a helpful voice assistant.",
+   voice: "alloy",
+   speechInstructions: "Speak in a friendly, natural conversational tone.",
+   outputFormat: "mp3",
+   streamingSpeech: {
+      minChunkSize: 40,
+      maxChunkSize: 180,
+      parallelGeneration: true,
+      maxParallelRequests: 2,
+   },
+   // Memory management (new in 0.1.0)
+   history: {
+      maxMessages: 50,       // keep last 50 messages
+      maxTotalChars: 100_000, // or trim when total chars exceed 100k
+   },
+   maxAudioInputSize: 5 * 1024 * 1024, // 5 MB limit
+   endpoint: process.env.VOICE_WS_ENDPOINT,
+   tools: { getWeather: weatherTool },
+});
+
+agent.on("text", ({ role, text }) => {
+   const prefix = role === "user" ? "👤" : "🤖";
+   console.log(prefix, text);
+});
+
+agent.on("chunk:text_delta", ({ text }) => process.stdout.write(text));
+agent.on("speech_start", ({ streaming }) => console.log("speech_start", streaming));
+agent.on("audio_chunk", ({ chunkId, format, uint8Array }) => {
+   console.log("audio_chunk", chunkId, format, uint8Array.length);
+});
+
+await agent.sendText("What's the weather in San Francisco?");
+
+if (process.env.VOICE_WS_ENDPOINT) {
+   await agent.connect(process.env.VOICE_WS_ENDPOINT);
+}
+```
+
+### Configuration options
+
+The agent accepts:
+
+| Option | Required | Default | Description |
+|---|---|---|---|
+| `model` | **yes** | — | AI SDK chat model (e.g. `openai("gpt-4o")`) |
+| `transcriptionModel` | no | — | AI SDK transcription model (e.g. `openai.transcription("whisper-1")`) |
+| `speechModel` | no | — | AI SDK speech model (e.g. `openai.speech("gpt-4o-mini-tts")`) |
+| `instructions` | no | `"You are a helpful voice assistant."` | System prompt |
+| `stopWhen` | no | `stepCountIs(5)` | Stopping condition for multi-step tool loops |
+| `tools` | no | `{}` | AI SDK tools map |
+| `endpoint` | no | — | Default WebSocket URL for `connect()` |
+| `voice` | no | `"alloy"` | TTS voice |
+| `speechInstructions` | no | — | Style instructions passed to the speech model |
+| `outputFormat` | no | `"mp3"` | Audio output format (`mp3`, `opus`, `wav`, …) |
+| `streamingSpeech` | no | see below | Streaming TTS chunk tuning |
+| `history` | no | see below | Conversation memory limits |
+| `maxAudioInputSize` | no | `10485760` (10 MB) | Maximum accepted audio input in bytes |
+
+#### `streamingSpeech`
+
+| Key | Default | Description |
+|---|---|---|
+| `minChunkSize` | `50` | Min characters before a sentence is sent to TTS |
+| `maxChunkSize` | `200` | Max characters per chunk (force-split at clause boundary) |
+| `parallelGeneration` | `true` | Start TTS for upcoming chunks while the current one plays |
+| `maxParallelRequests` | `3` | Cap on concurrent TTS requests |
+
+#### `history`
+
+| Key | Default | Description |
+|---|---|---|
+| `maxMessages` | `100` | Max messages kept in history (0 = unlimited). Oldest are trimmed in pairs. |
+| `maxTotalChars` | `0` (unlimited) | Max total characters across all messages. Oldest are trimmed when exceeded. |
+
+### Methods
+
+| Method | Description |
+|---|---|
+| `sendText(text)` | Process text input. Returns a promise with the full assistant response. Requests are queued serially. |
+| `sendAudio(base64Audio)` | Transcribe base64 audio and process the result. |
+| `sendAudioBuffer(buffer)` | Same as above, accepts a raw `Buffer` / `Uint8Array`. |
+| `transcribeAudio(buffer)` | Transcribe audio to text without generating a response. |
+| `generateAndSendSpeechFull(text)` | Non-streaming TTS fallback (entire text at once). |
+| `interruptSpeech(reason?)` | Cancel in-flight TTS only (LLM stream keeps running). |
+| `interruptCurrentResponse(reason?)` | Cancel **both** the LLM stream and TTS. Used for barge-in. |
+| `connect(url?)` / `handleSocket(ws)` | Establish or attach a WebSocket. Safe to call multiple times. |
+| `disconnect()` | Close the socket and abort all in-flight work. |
+| `destroy()` | Permanently release all resources. The agent cannot be reused. |
+| `clearHistory()` | Clear conversation history. |
+| `getHistory()` / `setHistory(msgs)` | Read or restore conversation history. |
+| `registerTools(tools)` | Merge additional tools into the agent. |
+
+### Read-only properties
+
+| Property | Type | Description |
+|---|---|---|
+| `connected` | `boolean` | Whether a WebSocket is connected |
+| `processing` | `boolean` | Whether a request is currently being processed |
+| `speaking` | `boolean` | Whether audio is currently being generated / sent |
+| `pendingSpeechChunks` | `number` | Number of queued TTS chunks |
+| `destroyed` | `boolean` | Whether `destroy()` has been called |
+
+### Events
+
+| Event | Payload | When |
+|---|---|---|
+| `text` | `{ role, text }` | User input received or full assistant response ready |
+| `chunk:text_delta` | `{ id, text }` | Each streaming text token from the LLM |
+| `chunk:reasoning_delta` | `{ id, text }` | Each reasoning token (models that support it) |
+| `chunk:tool_call` | `{ toolName, toolCallId, input }` | Tool invocation detected |
+| `tool_result` | `{ name, toolCallId, result }` | Tool execution finished |
+| `speech_start` | `{ streaming }` | TTS generation begins |
+| `speech_complete` | `{ streaming }` | All TTS chunks sent |
+| `speech_interrupted` | `{ reason }` | Speech was cancelled (barge-in, disconnect, error) |
+| `speech_chunk_queued` | `{ id, text }` | A text chunk entered the TTS queue |
+| `audio_chunk` | `{ chunkId, data, format, text, uint8Array }` | One TTS chunk is ready |
+| `audio` | `{ data, format, uint8Array }` | Full non-streaming TTS audio |
+| `transcription` | `{ text, language }` | Audio transcription result |
+| `audio_received` | `{ size }` | Raw audio input received (before transcription) |
+| `history_trimmed` | `{ removedCount, reason }` | Oldest messages evicted from history |
+| `connected` / `disconnected` | — | WebSocket lifecycle |
+| `warning` | `string` | Non-fatal issues (empty input, etc.) |
+| `error` | `Error` | Errors from LLM, TTS, transcription, or WebSocket |
+
+## Run (text-only check)
+
+This validates LLM + tool + streaming speech without requiring WebSocket:
+
+pnpm demo
+
+Expected logs include `text`, `chunk:text_delta`, tool events, and speech chunk events.
+
+## Run (WebSocket check)
+
+1. Start the local WS server:
+
+       pnpm ws:server
+
+2. In another terminal, run the demo:
+
+       pnpm demo
+
+The demo will:
+- run `sendText()` first (text-only sanity check), then
+- connect to `VOICE_WS_ENDPOINT` if provided,
+- emit streaming protocol messages (`text_delta`, `tool_call`, `audio_chunk`, `response_complete`, etc.).
+
+## Browser voice client (HTML)
+
+A simple browser client is available at [example/voice-client.html](example/voice-client.html).
+
+What it does:
+- captures microphone speech using Web Speech API (speech-to-text)
+- sends transcript to the agent via WebSocket (`type: "transcript"`)
+- receives streaming `audio_chunk` messages and plays them in order
+
+How to use:
+1. Start your agent server/WebSocket endpoint.
+2. Open [example/voice-client.html](example/voice-client.html) in a browser (Chrome/Edge recommended).
+3. Connect to `ws://localhost:8080` (or your endpoint), then click **Start Mic**.
+
+## Scripts
+
+- `pnpm build` – build TypeScript
+- `pnpm dev` – watch TypeScript
+- `pnpm demo` – run demo client
+- `pnpm ws:server` – run local test WebSocket server
+
+## Notes
+
+- If `VOICE_WS_ENDPOINT` is empty, WebSocket connect is skipped.
+- The sample WS server sends a mock `transcript` message for end-to-end testing.
+- Streaming TTS uses chunk queueing and supports interruption (`interrupt`).

package/dist/VoiceAgent.d.ts

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+import { WebSocket } from "ws";
+import { EventEmitter } from "events";
+import { streamText, LanguageModel, type Tool, type ModelMessage, type TranscriptionModel, type SpeechModel } from "ai";
+/**
+ * Configuration for streaming speech behavior
+ */
+interface StreamingSpeechConfig {
+    /** Minimum characters before generating speech for a chunk */
+    minChunkSize: number;
+    /** Maximum characters per chunk (will split at sentence boundary before this) */
+    maxChunkSize: number;
+    /** Whether to enable parallel TTS generation */
+    parallelGeneration: boolean;
+    /** Maximum number of parallel TTS requests */
+    maxParallelRequests: number;
+}
+/**
+ * Configuration for conversation history memory management
+ */
+interface HistoryConfig {
+    /** Maximum number of messages to keep in history. When exceeded, oldest messages are trimmed. Set to 0 for unlimited. */
+    maxMessages: number;
+    /** Maximum total character count across all messages. When exceeded, oldest messages are trimmed. Set to 0 for unlimited. */
+    maxTotalChars: number;
+}
+export interface VoiceAgentOptions {
+    model: LanguageModel;
+    transcriptionModel?: TranscriptionModel;
+    speechModel?: SpeechModel;
+    instructions?: string;
+    stopWhen?: NonNullable<Parameters<typeof streamText>[0]["stopWhen"]>;
+    tools?: Record<string, Tool>;
+    endpoint?: string;
+    voice?: string;
+    speechInstructions?: string;
+    outputFormat?: string;
+    /** Configuration for streaming speech generation */
+    streamingSpeech?: Partial<StreamingSpeechConfig>;
+    /** Configuration for conversation history memory limits */
+    history?: Partial<HistoryConfig>;
+    /** Maximum audio input size in bytes (default: 10 MB) */
+    maxAudioInputSize?: number;
+}
+export declare class VoiceAgent extends EventEmitter {
+    private socket?;
+    private tools;
+    private model;
+    private transcriptionModel?;
+    private speechModel?;
+    private instructions;
+    private stopWhen;
+    private endpoint?;
+    private isConnected;
+    private conversationHistory;
+    private voice;
+    private speechInstructions?;
+    private outputFormat;
+    private isProcessing;
+    private isDestroyed;
+    private inputQueue;
+    private processingQueue;
+    private currentStreamAbortController?;
+    private historyConfig;
+    private maxAudioInputSize;
+    private streamingSpeechConfig;
+    private currentSpeechAbortController?;
+    private speechChunkQueue;
+    private nextChunkId;
+    private isSpeaking;
+    private pendingTextBuffer;
+    private speechQueueDonePromise?;
+    private speechQueueDoneResolve?;
+    constructor(options: VoiceAgentOptions);
+    /**
+     * Ensure the agent has not been destroyed. Throws if it has.
+     */
+    private ensureNotDestroyed;
+    private setupListeners;
+    /**
+     * Clean up all in-flight state when the connection drops.
+     */
+    private cleanupOnDisconnect;
+    registerTools(tools: Record<string, Tool>): void;
+    /**
+     * Transcribe audio data to text using the configured transcription model
+     */
+    transcribeAudio(audioData: Buffer | Uint8Array): Promise<string>;
+    /**
+     * Generate speech from text using the configured speech model
+     * @param abortSignal Optional signal to cancel the speech generation
+     */
+    generateSpeechFromText(text: string, abortSignal?: AbortSignal): Promise<Uint8Array>;
+    /**
+     * Interrupt ongoing speech generation and playback (barge-in support).
+     * This only interrupts TTS — the LLM stream is left running.
+     */
+    interruptSpeech(reason?: string): void;
+    /**
+     * Interrupt both the current LLM stream and ongoing speech.
+     * Use this for barge-in scenarios where the entire response should be cancelled.
+     */
+    interruptCurrentResponse(reason?: string): void;
+    /**
+     * Extract complete sentences from text buffer
+     * Returns [extractedSentences, remainingBuffer]
+     */
+    private extractSentences;
+    /**
+     * Trim conversation history to stay within configured limits.
+     * Removes oldest messages (always in pairs to preserve user/assistant turns).
+     */
+    private trimHistory;
+    /**
+     * Queue a text chunk for speech generation
+     */
+    private queueSpeechChunk;
+    /**
+     * Generate audio for a single chunk
+     */
+    private generateChunkAudio;
+    /**
+     * Process the speech queue and send audio chunks in order
+     */
+    private processSpeechQueue;
+    /**
+     * Process text delta for streaming speech.
+     * Call this as text chunks arrive from LLM.
+     */
+    private processTextForStreamingSpeech;
+    /**
+     * Flush any remaining text in the buffer to speech
+     * Call this when stream ends
+     */
+    private flushStreamingSpeech;
+    /**
+     * Process incoming audio data: transcribe and generate response
+     */
+    private processAudioInput;
+    connect(url?: string): Promise<void>;
+    /**
+     * Attach an existing WebSocket (server-side usage).
+     * Use this when a WS server accepts a connection and you want the
+     * agent to handle messages on that socket.
+     */
+    handleSocket(socket: WebSocket): void;
+    /**
+     * Send text input for processing (bypasses transcription).
+     * Requests are queued and processed serially to prevent race conditions.
+     */
+    sendText(text: string): Promise<string>;
+    /**
+     * Send audio data to be transcribed and processed
+     * @param audioData Base64 encoded audio data
+     */
+    sendAudio(audioData: string): Promise<void>;
+    /**
+     * Send raw audio buffer to be transcribed and processed
+     */
+    sendAudioBuffer(audioBuffer: Buffer | Uint8Array): Promise<void>;
+    /**
+     * Enqueue a text input for serial processing.
+     * This ensures only one processUserInput runs at a time, preventing
+     * race conditions on conversationHistory, fullText accumulation, etc.
+     */
+    private enqueueInput;
+    /**
+     * Drain the input queue, processing one request at a time.
+     */
+    private drainInputQueue;
+    /**
+     * Process user input with streaming text generation.
+     * Handles the full pipeline: text -> LLM (streaming) -> TTS -> WebSocket.
+     *
+     * This method is designed to be called serially via drainInputQueue().
+     */
+    private processUserInput;
+    /**
+     * Generate speech for full text at once (non-streaming fallback)
+     * Useful when you want to bypass streaming speech for short responses
+     */
+    generateAndSendSpeechFull(text: string): Promise<void>;
+    /**
+     * Send a message via WebSocket if connected.
+     * Gracefully handles send failures (e.g., socket closing mid-send).
+     */
+    private sendWebSocketMessage;
+    /**
+     * Start listening for voice input
+     */
+    startListening(): void;
+    /**
+     * Stop listening for voice input
+     */
+    stopListening(): void;
+    /**
+     * Clear conversation history
+     */
+    clearHistory(): void;
+    /**
+     * Get current conversation history
+     */
+    getHistory(): ModelMessage[];
+    /**
+     * Set conversation history (useful for restoring sessions)
+     */
+    setHistory(history: ModelMessage[]): void;
+    /**
+     * Internal helper to close and clean up the current socket.
+     */
+    private disconnectSocket;
+    /**
+     * Disconnect from WebSocket and stop all in-flight work.
+     */
+    disconnect(): void;
+    /**
+     * Permanently destroy the agent, releasing all resources.
+     * After calling this, the agent cannot be reused.
+     */
+    destroy(): void;
+    /**
+     * Check if agent is connected to WebSocket
+     */
+    get connected(): boolean;
+    /**
+     * Check if agent is currently processing a request
+     */
+    get processing(): boolean;
+    /**
+     * Check if agent is currently speaking (generating/playing audio)
+     */
+    get speaking(): boolean;
+    /**
+     * Get the number of pending speech chunks in the queue
+     */
+    get pendingSpeechChunks(): number;
+    /**
+     * Check if agent has been permanently destroyed
+     */
+    get destroyed(): boolean;
+}
+export {};
+//# sourceMappingURL=VoiceAgent.d.ts.map

package/dist/VoiceAgent.d.ts.map

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