Files
clawdbot/extensions/voice-call/src/providers/stt-openai-realtime.ts
2026-01-12 21:44:19 +00:00

304 lines
8.3 KiB
TypeScript

/**
* OpenAI Realtime STT Provider
*
* Uses the OpenAI Realtime API for streaming transcription with:
* - Direct mu-law audio support (no conversion needed)
* - Built-in server-side VAD for turn detection
* - Low-latency streaming transcription
* - Partial transcript callbacks for real-time UI updates
*/
import WebSocket from "ws";
/**
* Configuration for OpenAI Realtime STT.
*/
export interface RealtimeSTTConfig {
/** OpenAI API key */
apiKey: string;
/** Model to use (default: gpt-4o-transcribe) */
model?: string;
/** Silence duration in ms before considering speech ended (default: 800) */
silenceDurationMs?: number;
/** VAD threshold 0-1 (default: 0.5) */
vadThreshold?: number;
}
/**
* Session for streaming audio and receiving transcripts.
*/
export interface RealtimeSTTSession {
/** Connect to the transcription service */
connect(): Promise<void>;
/** Send mu-law audio data (8kHz mono) */
sendAudio(audio: Buffer): void;
/** Wait for next complete transcript (after VAD detects end of speech) */
waitForTranscript(timeoutMs?: number): Promise<string>;
/** Set callback for partial transcripts (streaming) */
onPartial(callback: (partial: string) => void): void;
/** Set callback for final transcripts */
onTranscript(callback: (transcript: string) => void): void;
/** Close the session */
close(): void;
/** Check if session is connected */
isConnected(): boolean;
}
/**
* Provider factory for OpenAI Realtime STT sessions.
*/
export class OpenAIRealtimeSTTProvider {
readonly name = "openai-realtime";
private apiKey: string;
private model: string;
private silenceDurationMs: number;
private vadThreshold: number;
constructor(config: RealtimeSTTConfig) {
if (!config.apiKey) {
throw new Error("OpenAI API key required for Realtime STT");
}
this.apiKey = config.apiKey;
this.model = config.model || "gpt-4o-transcribe";
this.silenceDurationMs = config.silenceDurationMs || 800;
this.vadThreshold = config.vadThreshold || 0.5;
}
/**
* Create a new realtime transcription session.
*/
createSession(): RealtimeSTTSession {
return new OpenAIRealtimeSTTSession(
this.apiKey,
this.model,
this.silenceDurationMs,
this.vadThreshold,
);
}
}
/**
* WebSocket-based session for real-time speech-to-text.
*/
class OpenAIRealtimeSTTSession implements RealtimeSTTSession {
private static readonly MAX_RECONNECT_ATTEMPTS = 5;
private static readonly RECONNECT_DELAY_MS = 1000;
private ws: WebSocket | null = null;
private connected = false;
private closed = false;
private reconnectAttempts = 0;
private pendingTranscript = "";
private onTranscriptCallback: ((transcript: string) => void) | null = null;
private onPartialCallback: ((partial: string) => void) | null = null;
constructor(
private readonly apiKey: string,
private readonly model: string,
private readonly silenceDurationMs: number,
private readonly vadThreshold: number,
) {}
async connect(): Promise<void> {
this.closed = false;
this.reconnectAttempts = 0;
return this.doConnect();
}
private async doConnect(): Promise<void> {
return new Promise((resolve, reject) => {
const url = "wss://api.openai.com/v1/realtime?intent=transcription";
this.ws = new WebSocket(url, {
headers: {
Authorization: `Bearer ${this.apiKey}`,
"OpenAI-Beta": "realtime=v1",
},
});
this.ws.on("open", () => {
console.log("[RealtimeSTT] WebSocket connected");
this.connected = true;
this.reconnectAttempts = 0;
// Configure the transcription session
this.sendEvent({
type: "transcription_session.update",
session: {
input_audio_format: "g711_ulaw",
input_audio_transcription: {
model: this.model,
},
turn_detection: {
type: "server_vad",
threshold: this.vadThreshold,
prefix_padding_ms: 300,
silence_duration_ms: this.silenceDurationMs,
},
},
});
resolve();
});
this.ws.on("message", (data: Buffer) => {
try {
const event = JSON.parse(data.toString());
this.handleEvent(event);
} catch (e) {
console.error("[RealtimeSTT] Failed to parse event:", e);
}
});
this.ws.on("error", (error) => {
console.error("[RealtimeSTT] WebSocket error:", error);
if (!this.connected) reject(error);
});
this.ws.on("close", (code, reason) => {
console.log(
`[RealtimeSTT] WebSocket closed (code: ${code}, reason: ${reason?.toString() || "none"})`,
);
this.connected = false;
// Attempt reconnection if not intentionally closed
if (!this.closed) {
void this.attemptReconnect();
}
});
setTimeout(() => {
if (!this.connected) {
reject(new Error("Realtime STT connection timeout"));
}
}, 10000);
});
}
private async attemptReconnect(): Promise<void> {
if (this.closed) {
return;
}
if (
this.reconnectAttempts >= OpenAIRealtimeSTTSession.MAX_RECONNECT_ATTEMPTS
) {
console.error(
`[RealtimeSTT] Max reconnect attempts (${OpenAIRealtimeSTTSession.MAX_RECONNECT_ATTEMPTS}) reached`,
);
return;
}
this.reconnectAttempts++;
const delay =
OpenAIRealtimeSTTSession.RECONNECT_DELAY_MS *
2 ** (this.reconnectAttempts - 1);
console.log(
`[RealtimeSTT] Reconnecting ${this.reconnectAttempts}/${OpenAIRealtimeSTTSession.MAX_RECONNECT_ATTEMPTS} in ${delay}ms...`,
);
await new Promise((resolve) => setTimeout(resolve, delay));
if (this.closed) {
return;
}
try {
await this.doConnect();
console.log("[RealtimeSTT] Reconnected successfully");
} catch (error) {
console.error("[RealtimeSTT] Reconnect failed:", error);
}
}
private handleEvent(event: {
type: string;
delta?: string;
transcript?: string;
error?: unknown;
}): void {
switch (event.type) {
case "transcription_session.created":
case "transcription_session.updated":
case "input_audio_buffer.speech_stopped":
case "input_audio_buffer.committed":
console.log(`[RealtimeSTT] ${event.type}`);
break;
case "conversation.item.input_audio_transcription.delta":
if (event.delta) {
this.pendingTranscript += event.delta;
this.onPartialCallback?.(this.pendingTranscript);
}
break;
case "conversation.item.input_audio_transcription.completed":
if (event.transcript) {
console.log(`[RealtimeSTT] Transcript: ${event.transcript}`);
this.onTranscriptCallback?.(event.transcript);
}
this.pendingTranscript = "";
break;
case "input_audio_buffer.speech_started":
console.log("[RealtimeSTT] Speech started");
this.pendingTranscript = "";
break;
case "error":
console.error("[RealtimeSTT] Error:", event.error);
break;
}
}
private sendEvent(event: unknown): void {
if (this.ws?.readyState === WebSocket.OPEN) {
this.ws.send(JSON.stringify(event));
}
}
sendAudio(muLawData: Buffer): void {
if (!this.connected) return;
this.sendEvent({
type: "input_audio_buffer.append",
audio: muLawData.toString("base64"),
});
}
onPartial(callback: (partial: string) => void): void {
this.onPartialCallback = callback;
}
onTranscript(callback: (transcript: string) => void): void {
this.onTranscriptCallback = callback;
}
async waitForTranscript(timeoutMs = 30000): Promise<string> {
return new Promise((resolve, reject) => {
const timeout = setTimeout(() => {
this.onTranscriptCallback = null;
reject(new Error("Transcript timeout"));
}, timeoutMs);
this.onTranscriptCallback = (transcript) => {
clearTimeout(timeout);
this.onTranscriptCallback = null;
resolve(transcript);
};
});
}
close(): void {
this.closed = true;
if (this.ws) {
this.ws.close();
this.ws = null;
}
this.connected = false;
}
isConnected(): boolean {
return this.connected;
}
}