@llmrtc/llmrtc-provider-openai 1.1.0 → 1.2.0

package/CHANGELOG.md

@@ -1,5 +1,77 @@
 # @llmrtc/llmrtc-provider-openai
 
+## 1.2.0
+
+### Minor Changes
+
+- fd91344: Anthropic modernization and richer stop reasons.
+  - The StopReason union gains `refusal`, `content_filter`, `pause_turn`, and
+    `context_overflow`. Note for TypeScript consumers with exhaustive switches:
+    this widens the union. OpenAI-compatible providers now report
+    `content_filter` for filtered responses (previously mis-reported as
+    `stop_sequence`); Gemini safety blocks and Bedrock guardrail interventions
+    also map to `content_filter`.
+  - The Anthropic provider defaults to `claude-sonnet-5` and automatically
+    omits temperature/top_p for model families that reject them (Sonnet 5,
+    Opus 4.7+, Fable tier), with a `samplingParamsSupported` override. The
+    Bedrock provider applies the same guard for Claude models.
+  - Orchestrators log a warning when a turn ends with an unusual stop reason.
+  - CLI and example fallbacks move off retired model ids.
+
+- 319bb47: OpenAI voice updates: steerable TTS and current transcription models.
+  - TTSConfig gains an optional `instructions` field for instructable TTS
+    models. OpenAITTSProvider accepts `instructions` at the constructor and
+    per call, sends them for gpt-\* models (e.g. gpt-4o-mini-tts), and
+    ignores them with a one-time warning on tts-1/tts-1-hd.
+  - OpenAITTSVoice widened to the current voice roster (adds ash, ballad,
+    coral, sage, verse) while accepting any string for forward
+    compatibility; TTS model accepts any string. Note: code doing
+    exhaustive switches over OpenAITTSVoice will need adjusting, matching
+    the same widening in the official openai SDK. ballad/verse require
+    gpt-4o-mini-tts; the other nine voices work on all TTS models.
+  - OpenAIWhisperProvider documents gpt-4o-transcribe /
+    gpt-4o-mini-transcribe support.
+  - CLI: new OPENAI_STT_MODEL, OPENAI_TTS_MODEL, and
+    OPENAI_TTS_INSTRUCTIONS environment variables.
+
+- a7d1ecc: Streaming speech-to-text: live interim transcripts while the user speaks.
+  - New ElevenLabsScribeProvider: batch STT via Scribe v2 and realtime
+    streaming via the Scribe v2 Realtime WebSocket (sub-150ms partials).
+  - New OpenAIRealtimeSTTProvider: streaming transcription over the OpenAI
+    Realtime API (gpt-realtime-whisper by default; audio-duration billing).
+  - Opt-in server mode streamingSTT (CLI: STREAMING_STT=true) streams mic
+    audio to the STT provider from VAD speech start, relaying partial
+    transcripts to the client over the existing transcript/isFinal
+    protocol; falls back to buffered STT when the provider can't stream.
+  - Both orchestrators gain runTurnStreamFromAudioStream; STTProvider
+    gains an optional streamingInputSampleRate hint; core exports an
+    AsyncEventQueue utility. Buffered STT remains the default.
+  - Streaming sockets carry watchdog timeouts (connect/inactivity/final)
+    and surface early closes or transcription failures as turn errors
+    instead of silently dropping the utterance.
+  - Behavior notes: the onSpeechStart server hook now fires after barge-in
+    cancellation, so a slow hook can no longer delay interruptions;
+    STT_PROVIDER=elevenlabs now selects ElevenLabs Scribe (previously it
+    silently fell back to OpenAI Whisper).
+
+### Patch Changes
+
+- Updated dependencies [fd91344]
+- Updated dependencies [319bb47]
+- Updated dependencies [a7d1ecc]
+  - @llmrtc/llmrtc-core@1.2.0
+
+## 1.1.1
+
+### Patch Changes
+
+- b35b4e1: Fix streaming TTS failing with "response.body?.getReader is not a function"
+  when the OpenAI SDK returns a Node Readable body (Node < 18 or apps loading
+  the SDK's Node shims). The provider now handles web ReadableStream bodies,
+  async-iterable Node stream bodies, and buffered fallback. Previously the
+  error was caught upstream and every turn silently fell back to
+  non-streaming TTS, adding seconds of latency.
+
 ## 1.1.0
 
 ### Minor Changes

package/README.md

@@ -17,7 +17,7 @@ import { OpenAILLMProvider } from '@llmrtc/llmrtc-provider-openai';
 
 const llm = new OpenAILLMProvider({
   apiKey: process.env.OPENAI_API_KEY!,
-  model: 'gpt-4o'
+  model: 'gpt-5.6-terra' // or 'gpt-5.6-sol' (flagship), 'gpt-5.6-luna' (fast)
 });
 ```
 

package/dist/index.d.ts

@@ -16,6 +16,11 @@ export declare class OpenAILLMProvider implements LLMProvider {
 export interface OpenAIWhisperConfig {
     apiKey: string;
     baseURL?: string;
+    /**
+     * Transcription model (default: 'whisper-1'). The newer
+     * 'gpt-4o-transcribe' and 'gpt-4o-mini-transcribe' models run on the
+     * same endpoint with better accuracy, especially in noisy audio.
+     */
     model?: string;
     language?: string;
 }
@@ -27,32 +32,46 @@ export declare class OpenAIWhisperProvider implements STTProvider {
     constructor(config: OpenAIWhisperConfig);
     transcribe(audio: Buffer, config?: STTConfig): Promise<STTResult>;
 }
-export type OpenAITTSVoice = 'alloy' | 'echo' | 'fable' | 'onyx' | 'nova' | 'shimmer';
+export type OpenAITTSVoice = 'alloy' | 'ash' | 'ballad' | 'coral' | 'echo' | 'fable' | 'nova' | 'onyx' | 'sage' | 'shimmer' | 'verse' | (string & {});
 export type OpenAITTSFormat = 'mp3' | 'opus' | 'aac' | 'flac' | 'wav' | 'pcm';
+export type OpenAITTSModel = 'tts-1' | 'tts-1-hd' | 'gpt-4o-mini-tts' | (string & {});
 export interface OpenAITTSConfig {
     /** OpenAI API key */
     apiKey: string;
     /** Base URL for API (optional, for Azure OpenAI or proxies) */
     baseURL?: string;
     /** TTS model (default: 'tts-1') */
-    model?: 'tts-1' | 'tts-1-hd' | 'gpt-4o-mini-tts';
+    model?: OpenAITTSModel;
     /** Default voice (default: 'alloy') */
     voice?: OpenAITTSVoice;
-    /** Speech speed multiplier 0.25-4.0 (default: 1.0) */
+    /** Speech speed multiplier 0.25-4.0 (default: 1.0). No effect on gpt-4o-mini-tts - use `instructions` to direct pacing there. */
     speed?: number;
+    /**
+     * Default delivery instructions (tone, pacing, emotion, accent) for
+     * instructable models such as gpt-4o-mini-tts, e.g. "Speak like a
+     * calm customer-support agent". Instructions are only sent when the
+     * effective model name starts with 'gpt-'; on any other name
+     * (tts-1, tts-1-hd, proxy/deployment aliases) they are ignored with
+     * a one-time warning, since the API rejects them there.
+     */
+    instructions?: string;
 }
 /**
  * OpenAI Text-to-Speech Provider.
  *
- * Available voices: alloy, echo, fable, onyx, nova, shimmer
- * Available models: tts-1 (fast), tts-1-hd (quality), gpt-4o-mini-tts (instructable)
+ * Available voices: alloy, ash, coral, echo, fable, nova, onyx, sage,
+ * shimmer on all models; ballad and verse additionally on
+ * gpt-4o-mini-tts.
+ * Available models: tts-1 (fast), tts-1-hd (quality), gpt-4o-mini-tts
+ * (instructable - accepts natural-language delivery `instructions`).
  *
  * @example
  * ```typescript
  * const provider = new OpenAITTSProvider({
  *   apiKey: 'sk-...',
- *   model: 'tts-1',
- *   voice: 'nova'
+ *   model: 'gpt-4o-mini-tts',
+ *   voice: 'coral',
+ *   instructions: 'Speak warmly, at a relaxed pace.'
  * });
  * ```
  */
@@ -65,7 +84,10 @@ export declare class OpenAITTSProvider implements TTSProvider {
     private model;
     private voice;
     private speed;
+    private instructions?;
+    private warnedInstructionsUnsupported;
     constructor(config: OpenAITTSConfig);
+    private speechParams;
     speak(text: string, overrideConfig?: TTSConfig): Promise<TTSResult>;
     /**
      * Streaming TTS - returns audio chunks as they are generated.
@@ -76,3 +98,75 @@ export declare class OpenAITTSProvider implements TTSProvider {
      */
     speakStream(text: string, overrideConfig?: TTSConfig): AsyncIterable<Buffer>;
 }
+export interface OpenAIRealtimeSTTConfig {
+    /** OpenAI API key */
+    apiKey: string;
+    /**
+     * Transcription model (default: 'gpt-realtime-whisper' - native
+     * streaming). 'gpt-4o-transcribe' / 'gpt-4o-mini-transcribe' also work
+     * over the Realtime API.
+     */
+    model?: string;
+    /** ISO language code hint */
+    language?: string;
+    /**
+     * Latency/accuracy trade-off for gpt-realtime-whisper
+     * (default: server-side default).
+     */
+    delay?: 'minimal' | 'low' | 'medium' | 'high' | 'xhigh';
+    /**
+     * Realtime WebSocket URL
+     * (default: 'wss://api.openai.com/v1/realtime?intent=transcription')
+     */
+    url?: string;
+    /**
+     * Socket watchdog timeouts in ms. Defaults: connect 10000, inactivity
+     * 30000, final-after-commit 15000. Mostly useful in tests.
+     */
+    timeoutsMs?: {
+        connect?: number;
+        inactivity?: number;
+        final?: number;
+    };
+}
+/**
+ * OpenAI Realtime Speech-to-Text Provider.
+ *
+ * Streams audio to a transcription-type Realtime API session and yields
+ * interim transcripts as the user speaks. Partial results carry the
+ * accumulated transcript-so-far; the final result carries the complete
+ * transcript. Billing follows the transcription model's audio pricing,
+ * not realtime LLM tokens.
+ *
+ * Input frames must be 24kHz mono 16-bit signed LE PCM
+ * (streamingInputSampleRate). transcribe() accepts a 16-bit mono PCM WAV
+ * buffer for compatibility with the buffered pipeline; for general batch
+ * transcription use OpenAIWhisperProvider instead.
+ *
+ * @example
+ * ```typescript
+ * const stt = new OpenAIRealtimeSTTProvider({
+ *   apiKey: process.env.OPENAI_API_KEY!,
+ *   model: 'gpt-realtime-whisper'
+ * });
+ * ```
+ */
+export declare class OpenAIRealtimeSTTProvider implements STTProvider {
+    readonly name = "openai-realtime-stt";
+    /** The Realtime API consumes 24kHz mono 16-bit PCM */
+    readonly streamingInputSampleRate = 24000;
+    private readonly apiKey;
+    private readonly model;
+    private readonly language?;
+    private readonly delay?;
+    private readonly url;
+    private readonly timeoutsMs;
+    constructor(config: OpenAIRealtimeSTTConfig);
+    transcribeStream(audio: AsyncIterable<Buffer>, config?: STTConfig): AsyncIterable<STTResult>;
+    /**
+     * One-shot transcription of a 16-bit mono PCM WAV buffer (the format
+     * produced by the LLMRTC voice pipeline), streamed through a realtime
+     * session. For arbitrary containers use OpenAIWhisperProvider.
+     */
+    transcribe(audio: Buffer, config?: STTConfig): Promise<STTResult>;
+}

package/dist/index.js

@@ -1,4 +1,6 @@
 import OpenAI, { toFile } from 'openai';
+import WebSocket from 'ws';
+import { AsyncEventQueue } from '@llmrtc/llmrtc-core';
 import { mapToolsToOpenAI, mapToolChoiceToOpenAI, parseToolCallsFromOpenAI, mapStopReasonFromOpenAI, processToolCallDelta, finalizeToolCalls, } from './tool-adapter.js';
 export class OpenAILLMProvider {
     constructor(config) {
@@ -191,18 +193,26 @@ function mapMessages(messages) {
         return { role: m.role, content: [{ type: 'text', text: m.content }, ...imageParts] };
     });
 }
+/** The speech endpoint only accepts `instructions` on gpt-* TTS models. */
+function ttsModelSupportsInstructions(model) {
+    return model.toLowerCase().startsWith('gpt-');
+}
 /**
  * OpenAI Text-to-Speech Provider.
  *
- * Available voices: alloy, echo, fable, onyx, nova, shimmer
- * Available models: tts-1 (fast), tts-1-hd (quality), gpt-4o-mini-tts (instructable)
+ * Available voices: alloy, ash, coral, echo, fable, nova, onyx, sage,
+ * shimmer on all models; ballad and verse additionally on
+ * gpt-4o-mini-tts.
+ * Available models: tts-1 (fast), tts-1-hd (quality), gpt-4o-mini-tts
+ * (instructable - accepts natural-language delivery `instructions`).
  *
  * @example
  * ```typescript
  * const provider = new OpenAITTSProvider({
  *   apiKey: 'sk-...',
- *   model: 'tts-1',
- *   voice: 'nova'
+ *   model: 'gpt-4o-mini-tts',
+ *   voice: 'coral',
+ *   instructions: 'Speak warmly, at a relaxed pace.'
  * });
  * ```
  */
@@ -251,21 +261,48 @@ export class OpenAITTSProvider {
             writable: true,
             value: void 0
         });
+        Object.defineProperty(this, "instructions", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "warnedInstructionsUnsupported", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: false
+        });
         this.client = new OpenAI({ apiKey: config.apiKey, baseURL: config.baseURL });
         this.model = config.model ?? 'tts-1';
         this.voice = config.voice ?? 'alloy';
         this.speed = config.speed ?? 1.0;
+        this.instructions = config.instructions;
     }
-    async speak(text, overrideConfig) {
-        const voice = overrideConfig?.voice ?? this.voice;
-        const format = mapFormat(overrideConfig?.format);
-        const response = await this.client.audio.speech.create({
-            model: overrideConfig?.model ?? this.model,
-            voice,
+    speechParams(text, overrideConfig) {
+        const model = overrideConfig?.model ?? this.model;
+        const params = {
+            model,
+            voice: overrideConfig?.voice ?? this.voice,
             input: text,
-            response_format: format,
+            response_format: mapFormat(overrideConfig?.format),
             speed: this.speed
-        });
+        };
+        const instructions = overrideConfig?.instructions ?? this.instructions;
+        if (instructions) {
+            if (ttsModelSupportsInstructions(model)) {
+                params.instructions = instructions;
+            }
+            else if (!this.warnedInstructionsUnsupported) {
+                this.warnedInstructionsUnsupported = true;
+                console.warn(`[openai-tts] Model "${model}" does not support TTS instructions - ignoring them. ` +
+                    `Use an instructable model such as gpt-4o-mini-tts.`);
+            }
+        }
+        return params;
+    }
+    async speak(text, overrideConfig) {
+        const response = await this.client.audio.speech.create(this.speechParams(text, overrideConfig));
         const buffer = Buffer.from(await response.arrayBuffer());
         return {
             audio: buffer,
@@ -281,29 +318,32 @@ export class OpenAITTSProvider {
      * This is the recommended format for lowest latency (no decode step needed).
      */
     async *speakStream(text, overrideConfig) {
-        const voice = overrideConfig?.voice ?? this.voice;
-        const format = mapFormat(overrideConfig?.format);
-        const response = await this.client.audio.speech.create({
-            model: overrideConfig?.model ?? this.model,
-            voice,
-            input: text,
-            response_format: format,
-            speed: this.speed
-        });
-        // Response is a Response-like object with body stream
-        const reader = response.body?.getReader();
-        if (!reader) {
-            // Fallback: return the whole buffer if streaming not available
-            const buffer = Buffer.from(await response.arrayBuffer());
-            yield buffer;
+        const response = await this.client.audio.speech.create(this.speechParams(text, overrideConfig));
+        // The SDK's response body is a web ReadableStream when a global fetch
+        // is available, but a Node Readable when the SDK runs with its Node
+        // shims (e.g. Node < 18 or an app importing 'openai/shims/node').
+        // Handle both instead of assuming getReader() exists.
+        const body = response.body;
+        if (body && typeof body.getReader === 'function') {
+            const reader = body.getReader();
+            while (true) {
+                const { done, value } = await reader.read();
+                if (done)
+                    break;
+                yield Buffer.from(value);
+            }
             return;
         }
-        while (true) {
-            const { done, value } = await reader.read();
-            if (done)
-                break;
-            yield Buffer.from(value);
+        if (body &&
+            typeof body[Symbol.asyncIterator] === 'function') {
+            for await (const chunk of body) {
+                yield Buffer.from(chunk);
+            }
+            return;
         }
+        // Fallback: return the whole buffer if streaming is not available
+        const buffer = Buffer.from(await response.arrayBuffer());
+        yield buffer;
     }
 }
 /**
@@ -322,4 +362,296 @@ function mapFormat(format) {
             return 'mp3';
     }
 }
+const REALTIME_STT_TIMEOUTS = { connect: 10000, inactivity: 30000, final: 15000 };
+/**
+ * OpenAI Realtime Speech-to-Text Provider.
+ *
+ * Streams audio to a transcription-type Realtime API session and yields
+ * interim transcripts as the user speaks. Partial results carry the
+ * accumulated transcript-so-far; the final result carries the complete
+ * transcript. Billing follows the transcription model's audio pricing,
+ * not realtime LLM tokens.
+ *
+ * Input frames must be 24kHz mono 16-bit signed LE PCM
+ * (streamingInputSampleRate). transcribe() accepts a 16-bit mono PCM WAV
+ * buffer for compatibility with the buffered pipeline; for general batch
+ * transcription use OpenAIWhisperProvider instead.
+ *
+ * @example
+ * ```typescript
+ * const stt = new OpenAIRealtimeSTTProvider({
+ *   apiKey: process.env.OPENAI_API_KEY!,
+ *   model: 'gpt-realtime-whisper'
+ * });
+ * ```
+ */
+export class OpenAIRealtimeSTTProvider {
+    constructor(config) {
+        Object.defineProperty(this, "name", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 'openai-realtime-stt'
+        });
+        /** The Realtime API consumes 24kHz mono 16-bit PCM */
+        Object.defineProperty(this, "streamingInputSampleRate", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: 24000
+        });
+        Object.defineProperty(this, "apiKey", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "model", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "language", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "delay", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "url", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        Object.defineProperty(this, "timeoutsMs", {
+            enumerable: true,
+            configurable: true,
+            writable: true,
+            value: void 0
+        });
+        this.apiKey = config.apiKey;
+        this.model = config.model ?? 'gpt-realtime-whisper';
+        this.language = config.language;
+        this.delay = config.delay;
+        // intent=transcription opens a transcription-type session directly,
+        // without naming an (unused) realtime LLM session model
+        this.url = config.url ?? 'wss://api.openai.com/v1/realtime?intent=transcription';
+        this.timeoutsMs = {
+            connect: config.timeoutsMs?.connect ?? REALTIME_STT_TIMEOUTS.connect,
+            inactivity: config.timeoutsMs?.inactivity ?? REALTIME_STT_TIMEOUTS.inactivity,
+            final: config.timeoutsMs?.final ?? REALTIME_STT_TIMEOUTS.final
+        };
+    }
+    async *transcribeStream(audio, config) {
+        const model = config?.model ?? this.model;
+        const ws = new WebSocket(this.url, {
+            headers: { Authorization: `Bearer ${this.apiKey}` }
+        });
+        const queue = new AsyncEventQueue();
+        let runningText = '';
+        let completed = false;
+        let commitSent = false;
+        // Watchdog: a silently-stalled socket must not hang the voice turn
+        let watchdog = null;
+        const armWatchdog = (ms, waitingFor) => {
+            if (watchdog)
+                clearTimeout(watchdog);
+            watchdog = setTimeout(() => {
+                queue.fail(new Error(`OpenAI Realtime STT timed out waiting for ${waitingFor}`));
+                ws.terminate();
+            }, ms);
+            watchdog.unref?.();
+        };
+        const disarmWatchdog = () => {
+            if (watchdog)
+                clearTimeout(watchdog);
+            watchdog = null;
+        };
+        armWatchdog(this.timeoutsMs.connect, 'connection');
+        ws.on('message', (data) => {
+            armWatchdog(commitSent ? this.timeoutsMs.final : this.timeoutsMs.inactivity, commitSent ? 'the final transcript' : 'server messages');
+            let event;
+            try {
+                event = JSON.parse(data.toString());
+            }
+            catch {
+                return;
+            }
+            switch (event.type) {
+                case 'conversation.item.input_audio_transcription.delta':
+                    runningText += event.delta ?? '';
+                    queue.push({ text: runningText, isFinal: false, raw: event });
+                    break;
+                case 'conversation.item.input_audio_transcription.completed':
+                    completed = true;
+                    queue.push({ text: event.transcript ?? runningText, isFinal: true, raw: event });
+                    queue.end();
+                    ws.close();
+                    break;
+                case 'conversation.item.input_audio_transcription.failed':
+                    queue.fail(new Error(`OpenAI Realtime STT transcription failed: ${event.error?.message ?? JSON.stringify(event.error)}`));
+                    ws.close();
+                    break;
+                case 'error':
+                    queue.fail(new Error(`OpenAI Realtime STT error: ${event.error?.message ?? JSON.stringify(event.error)}`));
+                    ws.close();
+                    break;
+                default:
+                    break;
+            }
+        });
+        ws.on('error', (err) => {
+            queue.fail(new Error(`OpenAI Realtime STT socket error: ${err.message}`));
+        });
+        ws.on('close', () => {
+            disarmWatchdog();
+            // A close before the final transcript means the utterance was lost;
+            // surface it instead of silently producing an empty transcript
+            // (no-op when the queue already ended or failed)
+            if (!completed) {
+                queue.fail(new Error('OpenAI Realtime STT connection closed before the final transcript'));
+            }
+        });
+        const opened = new Promise((resolve, reject) => {
+            ws.once('open', () => resolve());
+            ws.once('error', (err) => reject(err));
+            ws.once('close', () => reject(new Error('socket closed before opening')));
+        });
+        const language = config?.language ?? this.language;
+        const sendLoop = (async () => {
+            await opened;
+            armWatchdog(this.timeoutsMs.inactivity, 'server messages');
+            // Transcription-type session; our own VAD segments the audio, so
+            // server turn detection stays off and we commit manually
+            ws.send(JSON.stringify({
+                type: 'session.update',
+                session: {
+                    type: 'transcription',
+                    audio: {
+                        input: {
+                            format: { type: 'audio/pcm', rate: this.streamingInputSampleRate },
+                            transcription: {
+                                model,
+                                ...(language && { language }),
+                                ...(this.delay && { delay: this.delay })
+                            },
+                            turn_detection: null
+                        }
+                    }
+                }
+            }));
+            let sentAny = false;
+            for await (const frame of audio) {
+                if (ws.readyState !== WebSocket.OPEN)
+                    return;
+                if (frame.length === 0)
+                    continue;
+                sentAny = true;
+                ws.send(JSON.stringify({ type: 'input_audio_buffer.append', audio: frame.toString('base64') }));
+            }
+            if (ws.readyState !== WebSocket.OPEN)
+                return;
+            if (sentAny) {
+                commitSent = true;
+                armWatchdog(this.timeoutsMs.final, 'the final transcript');
+                ws.send(JSON.stringify({ type: 'input_audio_buffer.commit' }));
+            }
+            else {
+                // Nothing to transcribe; committing an empty buffer errors
+                completed = true;
+                queue.end();
+                ws.close();
+            }
+        })().catch((err) => {
+            queue.fail(err);
+            ws.close();
+        });
+        try {
+            yield* queue;
+        }
+        finally {
+            disarmWatchdog();
+            // Close first: an abandoned sendLoop exits at its next frame once
+            // the socket is no longer OPEN, instead of feeding a dead session
+            if (ws.readyState === WebSocket.OPEN || ws.readyState === WebSocket.CONNECTING) {
+                ws.close();
+            }
+            // sendLoop never rejects (errors route into the queue); the bounded
+            // wait guards against a caller-owned frame iterable that never ends
+            await Promise.race([
+                sendLoop,
+                new Promise((resolve) => {
+                    const t = setTimeout(resolve, 5000);
+                    t.unref?.();
+                })
+            ]);
+        }
+    }
+    /**
+     * One-shot transcription of a 16-bit mono PCM WAV buffer (the format
+     * produced by the LLMRTC voice pipeline), streamed through a realtime
+     * session. For arbitrary containers use OpenAIWhisperProvider.
+     */
+    async transcribe(audio, config) {
+        const pcm = wavToRealtimePCM(audio, this.streamingInputSampleRate);
+        const frames = (async function* () {
+            // 100ms chunks
+            const chunkBytes = 4800;
+            for (let off = 0; off < pcm.length; off += chunkBytes) {
+                yield pcm.subarray(off, Math.min(off + chunkBytes, pcm.length));
+            }
+        })();
+        let last = { text: '', isFinal: true };
+        for await (const result of this.transcribeStream(frames, config)) {
+            if (result.isFinal) {
+                last = result;
+            }
+        }
+        return last;
+    }
+}
+/**
+ * Extract PCM from a 16-bit mono WAV buffer and linearly resample it to
+ * the target rate. Throws on non-WAV input.
+ */
+function wavToRealtimePCM(audio, targetRate) {
+    if (audio.length < 44 || audio.toString('ascii', 0, 4) !== 'RIFF' || audio.toString('ascii', 8, 12) !== 'WAVE') {
+        throw new Error('OpenAIRealtimeSTTProvider.transcribe expects a 16-bit mono PCM WAV buffer; use OpenAIWhisperProvider for other formats');
+    }
+    if (audio.toString('ascii', 36, 40) !== 'data') {
+        // Non-canonical header (extended fmt chunk, LIST metadata, ...) would
+        // silently misread the fields below - refuse instead
+        throw new Error('OpenAIRealtimeSTTProvider.transcribe expects a canonical 44-byte WAV header; use OpenAIWhisperProvider for other formats');
+    }
+    const sourceRate = audio.readUInt32LE(24);
+    const channels = audio.readUInt16LE(22);
+    const bits = audio.readUInt16LE(34);
+    if (channels !== 1 || bits !== 16) {
+        throw new Error(`OpenAIRealtimeSTTProvider.transcribe expects 16-bit mono WAV (got ${bits}-bit, ${channels}ch)`);
+    }
+    const pcm = audio.subarray(44);
+    if (sourceRate === targetRate) {
+        return pcm;
+    }
+    const sourceSamples = Math.floor(pcm.length / 2);
+    const targetSamples = Math.floor((sourceSamples * targetRate) / sourceRate);
+    const out = Buffer.alloc(targetSamples * 2);
+    for (let i = 0; i < targetSamples; i++) {
+        const pos = (i * sourceRate) / targetRate;
+        const i0 = Math.floor(pos);
+        const i1 = Math.min(i0 + 1, sourceSamples - 1);
+        const frac = pos - i0;
+        const s0 = pcm.readInt16LE(i0 * 2);
+        const s1 = pcm.readInt16LE(i1 * 2);
+        out.writeInt16LE(Math.round(s0 + (s1 - s0) * frac), i * 2);
+    }
+    return out;
+}
 //# sourceMappingURL=index.js.map

package/dist/index.js.map

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package/dist/tool-adapter.d.ts

@@ -5,7 +5,7 @@
  * parses tool calls from OpenAI responses.
  */
 import type { ChatCompletionTool, ChatCompletionToolChoiceOption } from 'openai/resources/chat/completions';
-import type { ToolDefinition, ToolCallRequest, ToolChoice } from '@llmrtc/llmrtc-core';
+import type { ToolDefinition, ToolCallRequest, ToolChoice, StopReason } from '@llmrtc/llmrtc-core';
 /**
  * Convert provider-agnostic tool definitions to OpenAI format
  */
@@ -55,4 +55,4 @@ export declare function finalizeToolCalls(accumulators: Map<number, StreamingToo
 /**
  * Map stop reason from OpenAI to provider-agnostic format
  */
-export declare function mapStopReasonFromOpenAI(finishReason: string | null | undefined): 'end_turn' | 'tool_use' | 'max_tokens' | 'stop_sequence' | undefined;
+export declare function mapStopReasonFromOpenAI(finishReason: string | null | undefined): StopReason | undefined;

package/dist/tool-adapter.js

@@ -124,7 +124,7 @@ export function mapStopReasonFromOpenAI(finishReason) {
         case 'length':
             return 'max_tokens';
         case 'content_filter':
-            return 'stop_sequence';
+            return 'content_filter';
         default:
             return undefined;
     }

package/dist/tool-adapter.js.map

@@ -1 +1 @@
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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@llmrtc/llmrtc-provider-openai",
-  "version": "1.1.0",
+  "version": "1.2.0",
   "description": "OpenAI/compatible LLM + Whisper adapters for @llmrtc/LLMRTC",
   "homepage": "https://www.llmrtc.org",
   "license": "Apache-2.0",
@@ -32,8 +32,9 @@
     "prepublishOnly": "npm run build"
   },
   "dependencies": {
-    "@llmrtc/llmrtc-core": "^1.1.0",
-    "openai": "^4.72.0"
+    "@llmrtc/llmrtc-core": "^1.2.0",
+    "openai": "^4.89.0",
+    "ws": "^8.16.0"
   },
   "exports": {
     ".": {
@@ -46,5 +47,8 @@
   "sideEffects": false,
   "engines": {
     "node": ">=20"
+  },
+  "devDependencies": {
+    "@types/ws": "^8.5.10"
   }
 }