@tekyzinc/stt-component 0.2.5 → 0.3.0

package/dist/index.cjs

@@ -26,8 +26,10 @@ __export(index_exports, {
   SpeechStreamingManager: () => SpeechStreamingManager,
   TypedEventEmitter: () => TypedEventEmitter,
   WorkerManager: () => WorkerManager,
+  pauseCapture: () => pauseCapture,
   resampleAudio: () => resampleAudio,
   resolveConfig: () => resolveConfig,
+  resumeCapture: () => resumeCapture,
   snapshotAudio: () => snapshotAudio,
   startCapture: () => startCapture,
   stopCapture: () => stopCapture
@@ -43,8 +45,8 @@ var DEFAULT_STT_CONFIG = {
   correction: {
     enabled: true,
     provider: "whisper",
-    pauseThreshold: 3e3,
-    forcedInterval: 5e3
+    pauseThreshold: 1e3,
+    forcedInterval: 3e3
   },
   chunking: {
     chunkLengthS: 30,
@@ -133,7 +135,19 @@ async function startCapture() {
   source.connect(processor);
   processor.connect(silencer);
   silencer.connect(audioCtx.destination);
-  return { audioCtx, stream, samples, _processor: processor };
+  return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };
+}
+async function pauseCapture(capture) {
+  capture._source.disconnect();
+  const currentSamples = [...capture.samples];
+  capture.samples.length = 0;
+  return resampleAudio(currentSamples, capture.audioCtx.sampleRate);
+}
+async function resumeCapture(capture) {
+  if (capture.audioCtx.state === "suspended") {
+    await capture.audioCtx.resume();
+  }
+  capture._source.connect(capture._processor);
 }
 function snapshotAudio(capture) {
   return [...capture.samples];
@@ -179,8 +193,17 @@ var import_meta = {};
 var WorkerManager = class extends TypedEventEmitter {
   worker = null;
   transcribeResolve = null;
+  currentTranscribePromise = null;
   modelReadyResolve = null;
   modelReadyReject = null;
+  /** True while a transcription job is running in the worker. */
+  get isTranscribing() {
+    return this.transcribeResolve !== null;
+  }
+  /** Await the current in-flight transcription without starting a new one. */
+  awaitCurrentTranscription() {
+    return this.currentTranscribePromise ?? Promise.resolve("");
+  }
   /** Spawn the Web Worker. Must be called before loadModel/transcribe. */
   spawn(workerUrl) {
     if (this.worker) return;
@@ -216,10 +239,11 @@ var WorkerManager = class extends TypedEventEmitter {
   async transcribe(audio) {
     if (!this.worker) throw new Error("Worker not spawned");
     if (audio.length === 0) return "";
-    return new Promise((resolve) => {
+    this.currentTranscribePromise = new Promise((resolve) => {
       this.transcribeResolve = resolve;
       this.worker.postMessage({ type: "transcribe", audio }, [audio.buffer]);
     });
+    return this.currentTranscribePromise;
   }
   /** Cancel any in-flight transcription. */
   cancel() {
@@ -439,8 +463,12 @@ var SpeechStreamingManager = class {
    * SpeechRecognition has claimed the microphone (onaudiostart) or after
    * a 300ms fallback — whichever comes first. The engine should await
    * this before calling getUserMedia to avoid dual-mic conflicts.
+   *
+   * When skipMicWait is true (warm restart — mic already active), returns
+   * immediately after calling recognition.start() without waiting for
+   * onaudiostart.
    */
-  start(language) {
+  start(language, skipMicWait = false) {
     const SR = getSpeechRecognition();
     if (!SR) {
       this.log("[SSM] SpeechRecognition not available in this environment");
@@ -550,6 +578,10 @@ var SpeechStreamingManager = class {
       );
       return Promise.resolve();
     }
+    if (skipMicWait) {
+      this.log("[SSM] skipMicWait \u2014 warm restart, returning immediately");
+      return Promise.resolve();
+    }
     return micClaimPromise;
   }
   clearNoResultTimer() {
@@ -597,6 +629,8 @@ var STTEngine = class extends TypedEventEmitter {
   capture = null;
   state;
   workerUrl;
+  /** Prevents performCorrection from emitting while stop() is consuming the in-flight result. */
+  _stopping = false;
   /**
    * Create a new STT engine instance.
    * @param config - Optional configuration overrides (model, backend, language, etc.).
@@ -642,14 +676,22 @@ var STTEngine = class extends TypedEventEmitter {
       throw new Error(`Cannot start: engine is "${this.state.status}", expected "ready"`);
     }
     try {
+      const warmCapture = this.capture && this.capture.stream.getTracks().every((t) => t.readyState === "live");
       this.emitDebug(
-        `[STT] start() \u2014 streaming: ${this.config.streaming.enabled}, lang: "${this.config.language}"`
+        `[STT] start() \u2014 streaming: ${this.config.streaming.enabled}, lang: "${this.config.language}", warm: ${!!warmCapture}`
       );
       if (this.config.streaming.enabled) {
-        await this.speechStreaming.start(this.config.language);
-        this.emitDebug("[STT] Speech API mic claim complete \u2014 starting getUserMedia");
+        await this.speechStreaming.start(this.config.language, !!warmCapture);
+        if (!warmCapture) {
+          this.emitDebug("[STT] Speech API mic claim complete \u2014 starting getUserMedia");
+        }
+      }
+      if (warmCapture) {
+        await resumeCapture(this.capture);
+        this.emitDebug("[STT] warm mic resumed \u2014 skipped getUserMedia");
+      } else {
+        this.capture = await startCapture();
       }
-      this.capture = await startCapture();
       this.updateStatus("recording");
       this.correctionOrchestrator.start();
     } catch (err) {
@@ -659,16 +701,49 @@ var STTEngine = class extends TypedEventEmitter {
       );
     }
   }
-  /** Stop recording, run final transcription, return text. */
+  /** Stop recording, run final transcription, return text.
+   *  Mic and AudioContext stay alive for fast restart — call destroy() to fully release. */
   async stop() {
     if (!this.capture) return "";
+    this._stopping = true;
     this.correctionOrchestrator.stop();
     this.speechStreaming.stop();
-    this.workerManager.cancel();
     this.updateStatus("processing");
+    if (this.workerManager.isTranscribing) {
+      try {
+        const [audio, inFlightText] = await Promise.all([
+          pauseCapture(this.capture),
+          this.workerManager.awaitCurrentTranscription()
+        ]);
+        this._stopping = false;
+        const text = inFlightText.trim();
+        if (text) {
+          this.emit("correction", text);
+          this.updateStatus("ready");
+          return text;
+        }
+        if (audio.length > 0) {
+          const freshText = await this.workerManager.transcribe(audio);
+          this.emit("correction", freshText);
+          this.updateStatus("ready");
+          return freshText;
+        }
+        this.updateStatus("ready");
+        return "";
+      } catch (err) {
+        this._stopping = false;
+        this.emitError(
+          "TRANSCRIPTION_FAILED",
+          err instanceof Error ? err.message : "Final transcription failed."
+        );
+        this.updateStatus("ready");
+        return "";
+      }
+    }
+    this.workerManager.cancel();
+    this._stopping = false;
     try {
-      const audio = await stopCapture(this.capture);
-      this.capture = null;
+      const audio = await pauseCapture(this.capture);
       if (audio.length === 0) {
         this.updateStatus("ready");
         return "";
@@ -691,6 +766,10 @@ var STTEngine = class extends TypedEventEmitter {
     this.correctionOrchestrator.stop();
     this.speechStreaming.destroy();
     if (this.capture) {
+      try {
+        this.capture._processor.disconnect();
+      } catch {
+      }
       for (const track of this.capture.stream.getTracks()) {
         track.stop();
       }
@@ -719,7 +798,7 @@ var STTEngine = class extends TypedEventEmitter {
       const audio = await resampleAudio(samples, nativeSr);
       if (audio.length === 0) return;
       const text = await this.workerManager.transcribe(audio);
-      if (text.trim() && this.capture) {
+      if (text.trim() && this.capture && !this._stopping) {
         this.emit("correction", text);
       }
     } catch (err) {
@@ -777,8 +856,10 @@ var STTEngine = class extends TypedEventEmitter {
   SpeechStreamingManager,
   TypedEventEmitter,
   WorkerManager,
+  pauseCapture,
   resampleAudio,
   resolveConfig,
+  resumeCapture,
   snapshotAudio,
   startCapture,
   stopCapture

package/dist/index.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/index.ts","../src/types.ts","../src/event-emitter.ts","../src/audio-capture.ts","../src/worker-manager.ts","../src/correction-orchestrator.ts","../src/speech-streaming.ts","../src/stt-engine.ts"],"sourcesContent":["// Public types\r\nexport type {\r\n  STTModelSize,\r\n  STTBackend,\r\n  STTStatus,\r\n  STTCorrectionProvider,\r\n  STTStreamingProvider,\r\n  STTCorrectionConfig,\r\n  STTStreamingConfig,\r\n  STTChunkingConfig,\r\n  STTConfig,\r\n  ResolvedSTTConfig,\r\n  STTState,\r\n  STTError,\r\n  STTEvents,\r\n  AudioCaptureHandle,\r\n} from './types.js';\r\n\r\n// Public values\r\nexport { DEFAULT_STT_CONFIG, resolveConfig } from './types.js';\r\n\r\n// Event emitter\r\nexport { TypedEventEmitter } from './event-emitter.js';\r\n\r\n// Audio capture\r\nexport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\r\n\r\n// Worker manager\r\nexport { WorkerManager } from './worker-manager.js';\r\nexport type { WorkerManagerEvents } from './worker-manager.js';\r\n\r\n// Correction orchestrator\r\nexport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\n\r\n// Speech streaming\r\nexport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n// STT Engine (main public API)\r\nexport { STTEngine } from './stt-engine.js';\r\n","/** Supported Whisper model sizes. */\r\nexport type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';\r\n\r\n/** Supported compute backends. */\r\nexport type STTBackend = 'webgpu' | 'wasm' | 'auto';\r\n\r\n/** Engine lifecycle states. */\r\nexport type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';\r\n\r\n/** Supported correction engine providers. */\r\nexport type STTCorrectionProvider = 'whisper';\r\n\r\n/** Supported real-time streaming providers. */\r\nexport type STTStreamingProvider = 'web-speech-api';\r\n\r\n/** Correction engine configuration. */\r\nexport interface STTCorrectionConfig {\r\n  /** Enable mid-recording correction. Default: true */\r\n  enabled?: boolean;\r\n  /** Correction engine provider. Default: 'whisper' */\r\n  provider?: STTCorrectionProvider;\r\n  /** Silence duration (ms) before triggering correction. Default: 3000 */\r\n  pauseThreshold?: number;\r\n  /** Maximum interval (ms) between forced corrections. Default: 5000 */\r\n  forcedInterval?: number;\r\n}\r\n\r\n/** Real-time streaming preview configuration. */\r\nexport interface STTStreamingConfig {\r\n  /** Enable real-time streaming transcript. Default: true */\r\n  enabled?: boolean;\r\n  /** Streaming provider. Default: 'web-speech-api' */\r\n  provider?: STTStreamingProvider;\r\n}\r\n\r\n/** Audio chunking configuration for long-form audio. */\r\nexport interface STTChunkingConfig {\r\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\r\n  chunkLengthS?: number;\r\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\r\n  strideLengthS?: number;\r\n}\r\n\r\n/** Full engine configuration. All fields optional — sensible defaults applied. */\r\nexport interface STTConfig {\r\n  /** Whisper model size. Default: 'tiny' */\r\n  model?: STTModelSize;\r\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\r\n  backend?: STTBackend;\r\n  /** Transcription language. Default: 'en' */\r\n  language?: string;\r\n  /** Model quantization dtype. Default: 'q4' */\r\n  dtype?: string;\r\n  /** Mid-recording correction settings. */\r\n  correction?: STTCorrectionConfig;\r\n  /** Audio chunking settings for long-form audio. */\r\n  chunking?: STTChunkingConfig;\r\n  /** Web Speech API streaming preview settings. */\r\n  streaming?: STTStreamingConfig;\r\n}\r\n\r\n/** Resolved configuration with all defaults applied. */\r\nexport interface ResolvedSTTConfig {\r\n  model: STTModelSize;\r\n  backend: STTBackend;\r\n  language: string;\r\n  dtype: string;\r\n  correction: Required<STTCorrectionConfig>;\r\n  chunking: Required<STTChunkingConfig>;\r\n  streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/** Engine state exposed to consumers via status events. */\r\nexport interface STTState {\r\n  status: STTStatus;\r\n  isModelLoaded: boolean;\r\n  /** Model download progress (0–100). */\r\n  loadProgress: number;\r\n  /** Active compute backend, or null if not yet determined. */\r\n  backend: 'webgpu' | 'wasm' | null;\r\n  error: string | null;\r\n}\r\n\r\n/** Structured error emitted via the 'error' event. */\r\nexport interface STTError {\r\n  code: string;\r\n  message: string;\r\n}\r\n\r\n/** Event map for the typed event emitter. */\r\nexport type STTEvents = {\r\n  /** Streaming interim text during recording. */\r\n  transcript: (text: string) => void;\r\n  /** Whisper-corrected text replacing interim text. */\r\n  correction: (text: string) => void;\r\n  /** Actionable error (mic denied, model fail, transcription fail). */\r\n  error: (error: STTError) => void;\r\n  /** Engine state change. */\r\n  status: (state: STTState) => void;\r\n  /** Diagnostic log for debugging (subscribe to capture all internal events). */\r\n  debug: (message: string) => void;\r\n};\r\n\r\n/** Handle returned by audio capture — used internally. */\r\nexport interface AudioCaptureHandle {\r\n  audioCtx: AudioContext;\r\n  stream: MediaStream;\r\n  samples: Float32Array[];\r\n  /** Retain reference to prevent GC from stopping audio processing. */\r\n  _processor: ScriptProcessorNode;\r\n}\r\n\r\n/** Message sent from main thread to Whisper worker. */\r\nexport interface WorkerMessage {\r\n  type: 'load' | 'transcribe' | 'cancel';\r\n  audio?: Float32Array;\r\n  config?: {\r\n    model: string;\r\n    backend: STTBackend;\r\n    language: string;\r\n    dtype: string;\r\n    chunkLengthS: number;\r\n    strideLengthS: number;\r\n  };\r\n}\r\n\r\n/** Response sent from Whisper worker to main thread. */\r\nexport interface WorkerResponse {\r\n  type: 'progress' | 'ready' | 'result' | 'error';\r\n  data?: unknown;\r\n}\r\n\r\n/** Default configuration values. */\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n  model: 'tiny',\r\n  backend: 'auto',\r\n  language: 'en',\r\n  dtype: 'q4',\r\n  correction: {\r\n    enabled: true,\r\n    provider: 'whisper',\r\n    pauseThreshold: 3_000,\r\n    forcedInterval: 5_000,\r\n  },\r\n  chunking: {\r\n    chunkLengthS: 30,\r\n    strideLengthS: 5,\r\n  },\r\n  streaming: {\r\n    enabled: true,\r\n    provider: 'web-speech-api',\r\n  },\r\n};\r\n\r\n/** Merge user config with defaults to produce resolved config. */\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n  return {\r\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n    correction: {\r\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n      pauseThreshold:\r\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n      forcedInterval:\r\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n    },\r\n    chunking: {\r\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n    },\r\n    streaming: {\r\n      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n    },\r\n  };\r\n}\r\n","/**\n * A generic, typed event emitter.\n *\n * Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n */\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n  private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n  /** Subscribe to an event. */\n  on<K extends keyof T>(event: K, listener: T[K]): void {\n    let set = this.listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.listeners.set(event, set);\n    }\n    set.add(listener as T[keyof T]);\n  }\n\n  /** Unsubscribe a specific listener. No-op if not registered. */\n  off<K extends keyof T>(event: K, listener: T[K]): void {\n    this.listeners.get(event)?.delete(listener as T[keyof T]);\n  }\n\n  /** Emit an event, calling all registered listeners in insertion order. */\n  emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n    const set = this.listeners.get(event);\n    if (!set) return;\n    for (const listener of set) {\n      (listener as (...a: Parameters<T[K]>) => void)(...args);\n    }\n  }\n\n  /** Remove all listeners, optionally for a single event. */\n  removeAllListeners(event?: keyof T): void {\n    if (event !== undefined) {\n      this.listeners.delete(event);\n    } else {\n      this.listeners.clear();\n    }\n  }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/**\r\n * Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n */\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n  const stream = await navigator.mediaDevices.getUserMedia({\r\n    audio: { channelCount: 1 },\r\n  });\r\n  const audioCtx = new AudioContext();\r\n\r\n  // Chrome may suspend AudioContext — must resume within user gesture\r\n  if (audioCtx.state === 'suspended') {\r\n    await audioCtx.resume();\r\n  }\r\n\r\n  const source = audioCtx.createMediaStreamSource(stream);\r\n  const samples: Float32Array[] = [];\r\n\r\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n  };\r\n\r\n  // Connect through a silent gain node so mic audio doesn't play back\r\n  const silencer = audioCtx.createGain();\r\n  silencer.gain.value = 0;\r\n  source.connect(processor);\r\n  processor.connect(silencer);\r\n  silencer.connect(audioCtx.destination);\r\n\r\n  return { audioCtx, stream, samples, _processor: processor };\r\n}\r\n\r\n/**\r\n * Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n */\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n  return [...capture.samples];\r\n}\r\n\r\n/**\r\n * Concatenate sample chunks and resample to 16kHz for Whisper.\r\n */\r\nexport async function resampleAudio(\r\n  samples: Float32Array[],\r\n  nativeSr: number,\r\n): Promise<Float32Array> {\r\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n  if (totalLength === 0) return new Float32Array(0);\r\n\r\n  const fullAudio = new Float32Array(totalLength);\r\n  let offset = 0;\r\n  for (const s of samples) {\r\n    fullAudio.set(s, offset);\r\n    offset += s.length;\r\n  }\r\n\r\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n  const duration = fullAudio.length / nativeSr;\r\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\r\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n  buffer.getChannelData(0).set(fullAudio);\r\n  const src = offline.createBufferSource();\r\n  src.buffer = buffer;\r\n  src.connect(offline.destination);\r\n  src.start(0);\r\n  const resampled = await offline.startRendering();\r\n  return resampled.getChannelData(0);\r\n}\r\n\r\n/**\r\n * Stop capturing and return resampled audio at 16kHz.\r\n */\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n  const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n  // Disconnect processor to stop capturing\r\n  try {\r\n    _processor.disconnect();\r\n  } catch {\r\n    /* already disconnected */\r\n  }\r\n\r\n  // Stop microphone tracks\r\n  for (const track of stream.getTracks()) {\r\n    track.stop();\r\n  }\r\n\r\n  const nativeSr = audioCtx.sampleRate;\r\n  await audioCtx.close();\r\n\r\n  return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/** Events emitted by the WorkerManager. */\nexport type WorkerManagerEvents = {\n  progress: (percent: number) => void;\n  ready: () => void;\n  result: (text: string) => void;\n  error: (message: string) => void;\n};\n\n/**\n * Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n */\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n  private worker: Worker | null = null;\n  private transcribeResolve: ((text: string) => void) | null = null;\n  private modelReadyResolve: (() => void) | null = null;\n  private modelReadyReject: ((err: Error) => void) | null = null;\n\n  /** Spawn the Web Worker. Must be called before loadModel/transcribe. */\n  spawn(workerUrl?: URL): void {\n    if (this.worker) return;\n\n    const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n    this.worker = new Worker(url, { type: 'module' });\n    this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n      this.handleMessage(e.data);\n    };\n    this.worker.onerror = (e: ErrorEvent) => {\n      this.emit('error', e.message ?? 'Worker error');\n    };\n  }\n\n  /** Load the Whisper model in the worker. Resolves when ready. */\n  async loadModel(config: ResolvedSTTConfig): Promise<void> {\n    if (!this.worker) throw new Error('Worker not spawned');\n\n    return new Promise<void>((resolve, reject) => {\n      this.modelReadyResolve = resolve;\n      this.modelReadyReject = reject;\n      this.worker!.postMessage({\n        type: 'load',\n        config: {\n          model: config.model,\n          backend: config.backend,\n          language: config.language,\n          dtype: config.dtype,\n          chunkLengthS: config.chunking.chunkLengthS,\n          strideLengthS: config.chunking.strideLengthS,\n        },\n      });\n    });\n  }\n\n  /** Send audio to the worker for transcription. Resolves with text. */\n  async transcribe(audio: Float32Array): Promise<string> {\n    if (!this.worker) throw new Error('Worker not spawned');\n    if (audio.length === 0) return '';\n\n    return new Promise<string>((resolve) => {\n      this.transcribeResolve = resolve;\n      this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n    });\n  }\n\n  /** Cancel any in-flight transcription. */\n  cancel(): void {\n    this.worker?.postMessage({ type: 'cancel' });\n    if (this.transcribeResolve) {\n      this.transcribeResolve('');\n      this.transcribeResolve = null;\n    }\n  }\n\n  /** Terminate the worker and release resources. */\n  destroy(): void {\n    this.cancel();\n    this.worker?.terminate();\n    this.worker = null;\n    this.removeAllListeners();\n  }\n\n  private handleMessage(msg: WorkerResponse): void {\n    switch (msg.type) {\n      case 'progress':\n        this.emit('progress', msg.data as number);\n        break;\n      case 'ready':\n        this.emit('ready');\n        this.modelReadyResolve?.();\n        this.modelReadyResolve = null;\n        this.modelReadyReject = null;\n        break;\n      case 'result':\n        this.emit('result', msg.data as string);\n        this.transcribeResolve?.(msg.data as string);\n        this.transcribeResolve = null;\n        break;\n      case 'error': {\n        const errMsg = msg.data as string;\n        this.emit('error', errMsg);\n        // Reject model load if still pending\n        if (this.modelReadyReject) {\n          this.modelReadyReject(new Error(errMsg));\n          this.modelReadyResolve = null;\n          this.modelReadyReject = null;\n        }\n        // Resolve transcribe with empty string on error\n        if (this.transcribeResolve) {\n          this.transcribeResolve('');\n          this.transcribeResolve = null;\n        }\n        break;\n      }\n    }\n  }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/**\n * Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n */\nexport class CorrectionOrchestrator {\n  private forcedTimer: ReturnType<typeof setInterval> | null = null;\n  private lastCorrectionTime = 0;\n  private correctionFn: (() => void) | null = null;\n  private config: ResolvedSTTConfig['correction'];\n\n  /** Create a new correction orchestrator with the given timing config. */\n  constructor(config: ResolvedSTTConfig['correction']) {\n    this.config = config;\n  }\n\n  /** Set the function to call when a correction is triggered. */\n  setCorrectionFn(fn: () => void): void {\n    this.correctionFn = fn;\n  }\n\n  /** Start the correction orchestrator (begin forced interval timer). */\n  start(): void {\n    if (!this.config.enabled) return;\n\n    this.lastCorrectionTime = Date.now();\n    this.startForcedTimer();\n  }\n\n  /** Stop the orchestrator (clear all timers). */\n  stop(): void {\n    this.stopForcedTimer();\n  }\n\n  /** Called when a speech pause is detected. Triggers correction if cooldown elapsed. */\n  onPauseDetected(): void {\n    if (!this.config.enabled) return;\n\n    const now = Date.now();\n    if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n    this.triggerCorrection();\n  }\n\n  /** Force a correction now (resets timer). */\n  forceCorrection(): void {\n    this.triggerCorrection();\n  }\n\n  private triggerCorrection(): void {\n    this.lastCorrectionTime = Date.now();\n    this.correctionFn?.();\n    // Reset forced timer after any correction\n    this.restartForcedTimer();\n  }\n\n  private startForcedTimer(): void {\n    this.stopForcedTimer();\n    this.forcedTimer = setInterval(() => {\n      this.triggerCorrection();\n    }, this.config.forcedInterval);\n  }\n\n  private stopForcedTimer(): void {\n    if (this.forcedTimer) {\n      clearInterval(this.forcedTimer);\n      this.forcedTimer = null;\n    }\n  }\n\n  private restartForcedTimer(): void {\n    if (this.forcedTimer) {\n      this.startForcedTimer();\n    }\n  }\n}\n","/* ─── Web Speech API types ──────────────────────────────── */\r\n\r\ninterface SpeechRecognitionEvent {\r\n  results: SpeechRecognitionResultList;\r\n  resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n  error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n  continuous: boolean;\r\n  interimResults: boolean;\r\n  lang: string;\r\n  onaudiostart: (() => void) | null;\r\n  onresult: ((e: SpeechRecognitionEvent) => void) | null;\r\n  onerror: ((e: SpeechRecognitionErrorEvent) => void) | null;\r\n  onend: (() => void) | null;\r\n  start: () => void;\r\n  stop: () => void;\r\n  abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor | null {\r\n  if (typeof globalThis === 'undefined') return null;\r\n  const w = globalThis as unknown as Record<string, unknown>;\r\n  return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor | null;\r\n}\r\n\r\n/* ─── Language mapping ──────────────────────────────────── */\r\n\r\n/** Map Whisper language codes to BCP-47 locale tags for the Speech API. */\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n  en: 'en-US',\r\n  english: 'en-US',\r\n  zh: 'zh-CN',\r\n  chinese: 'zh-CN',\r\n  de: 'de-DE',\r\n  german: 'de-DE',\r\n  es: 'es-ES',\r\n  spanish: 'es-ES',\r\n  ru: 'ru-RU',\r\n  russian: 'ru-RU',\r\n  ko: 'ko-KR',\r\n  korean: 'ko-KR',\r\n  fr: 'fr-FR',\r\n  french: 'fr-FR',\r\n  ja: 'ja-JP',\r\n  japanese: 'ja-JP',\r\n  pt: 'pt-BR',\r\n  portuguese: 'pt-BR',\r\n  tr: 'tr-TR',\r\n  turkish: 'tr-TR',\r\n  pl: 'pl-PL',\r\n  polish: 'pl-PL',\r\n  nl: 'nl-NL',\r\n  dutch: 'nl-NL',\r\n  ar: 'ar-SA',\r\n  arabic: 'ar-SA',\r\n  sv: 'sv-SE',\r\n  swedish: 'sv-SE',\r\n  it: 'it-IT',\r\n  italian: 'it-IT',\r\n  id: 'id-ID',\r\n  indonesian: 'id-ID',\r\n  hi: 'hi-IN',\r\n  hindi: 'hi-IN',\r\n  fi: 'fi-FI',\r\n  finnish: 'fi-FI',\r\n  vi: 'vi-VN',\r\n  vietnamese: 'vi-VN',\r\n  he: 'he-IL',\r\n  hebrew: 'he-IL',\r\n  uk: 'uk-UA',\r\n  ukrainian: 'uk-UA',\r\n  el: 'el-GR',\r\n  greek: 'el-GR',\r\n  ms: 'ms-MY',\r\n  malay: 'ms-MY',\r\n  cs: 'cs-CZ',\r\n  czech: 'cs-CZ',\r\n  ro: 'ro-RO',\r\n  romanian: 'ro-RO',\r\n  da: 'da-DK',\r\n  danish: 'da-DK',\r\n  hu: 'hu-HU',\r\n  hungarian: 'hu-HU',\r\n  no: 'nb-NO',\r\n  norwegian: 'nb-NO',\r\n  th: 'th-TH',\r\n  thai: 'th-TH',\r\n};\r\n\r\n/**\r\n * Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n */\r\nfunction toBCP47(language: string): string {\r\n  if (language.includes('-')) return language;\r\n  return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/* ─── SpeechStreamingManager ────────────────────────────── */\r\n\r\n/**\r\n * Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n */\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n\r\nexport class SpeechStreamingManager {\r\n  private recognition: SpeechRecognitionInstance | null = null;\r\n  private accumulated = '';\r\n  private active = false;\r\n  private receivedResult = false;\r\n  private noResultTimer: ReturnType<typeof setTimeout> | null = null;\r\n  private onTranscript: ((text: string) => void) | null = null;\r\n  private onPause: (() => void) | null = null;\r\n  private onError: ((message: string) => void) | null = null;\r\n  private onDebug: ((message: string) => void) | null = null;\r\n\r\n  /** Check if the Web Speech API is available in this environment. */\r\n  static isSupported(): boolean {\r\n    return getSpeechRecognition() !== null;\r\n  }\r\n\r\n  /** Set callback for streaming transcript updates (interim + final text). */\r\n  setOnTranscript(fn: (text: string) => void): void {\r\n    this.onTranscript = fn;\r\n  }\r\n\r\n  /** Set callback for speech pause detection (Speech API onend). */\r\n  setOnPause(fn: () => void): void {\r\n    this.onPause = fn;\r\n  }\r\n\r\n  /** Set callback for errors. */\r\n  setOnError(fn: (message: string) => void): void {\r\n    this.onError = fn;\r\n  }\r\n\r\n  /** Set callback for diagnostic debug messages. */\r\n  setOnDebug(fn: (message: string) => void): void {\r\n    this.onDebug = fn;\r\n  }\r\n\r\n  private log(message: string): void {\r\n    this.onDebug?.(message);\r\n    console.warn(message);\r\n  }\r\n\r\n  /**\r\n   * Start streaming recognition. Returns a Promise that resolves once\r\n   * SpeechRecognition has claimed the microphone (onaudiostart) or after\r\n   * a 300ms fallback — whichever comes first. The engine should await\r\n   * this before calling getUserMedia to avoid dual-mic conflicts.\r\n   */\r\n  start(language: string): Promise<void> {\r\n    const SR = getSpeechRecognition();\r\n    if (!SR) {\r\n      this.log('[SSM] SpeechRecognition not available in this environment');\r\n      return Promise.resolve();\r\n    }\r\n\r\n    const bcp47 = toBCP47(language);\r\n    this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n    this.accumulated = '';\r\n    this.active = true;\r\n    this.receivedResult = false;\r\n\r\n    const recognition = new SR();\r\n    recognition.continuous = true;\r\n    recognition.interimResults = true;\r\n    recognition.lang = bcp47;\r\n\r\n    let lastFinalIndex = -1;\r\n    let lastFinalText = '';\r\n\r\n    // Promise resolves when SR claims mic (onaudiostart) or after fallback.\r\n    // This ensures getUserMedia doesn't compete for the mic.\r\n    let micReady = false;\r\n    const micClaimPromise = new Promise<void>((resolve) => {\r\n      recognition.onaudiostart = () => {\r\n        if (this.recognition !== recognition) return;\r\n        this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n        if (!micReady) {\r\n          micReady = true;\r\n          resolve();\r\n        }\r\n      };\r\n      // Fallback: resolve after 300ms even if onaudiostart never fires\r\n      setTimeout(() => {\r\n        if (!micReady) {\r\n          micReady = true;\r\n          this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n          resolve();\r\n        }\r\n      }, 300);\r\n    });\r\n\r\n    // Detect silent failure: if no onresult fires within timeout, emit error\r\n    this.clearNoResultTimer();\r\n    this.noResultTimer = setTimeout(() => {\r\n      if (this.active && !this.receivedResult) {\r\n        this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n        this.onError?.(\r\n          'Speech streaming started but received no results. ' +\r\n            'Mic may be blocked by another audio capture.',\r\n        );\r\n      }\r\n    }, NO_RESULT_TIMEOUT_MS);\r\n\r\n    recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.receivedResult = true;\r\n      this.clearNoResultTimer();\r\n\r\n      let final_ = '';\r\n      let interim = '';\r\n      for (let i = e.resultIndex; i < e.results.length; i++) {\r\n        const t = e.results[i][0].transcript;\r\n        if (e.results[i].isFinal) {\r\n          if (i > lastFinalIndex) {\r\n            final_ += t;\r\n            lastFinalIndex = i;\r\n          }\r\n        } else {\r\n          interim += t;\r\n        }\r\n      }\r\n\r\n      this.log(\r\n        `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n      );\r\n\r\n      if (final_ && final_.trim() !== lastFinalText) {\r\n        lastFinalText = final_.trim();\r\n        this.accumulated = this.accumulated\r\n          ? this.accumulated + ' ' + final_.trim()\r\n          : final_.trim();\r\n        this.onTranscript?.(this.accumulated);\r\n      } else if (interim) {\r\n        const trimmed = interim.trimStart();\r\n        const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n        this.onTranscript?.(full);\r\n      }\r\n    };\r\n\r\n    recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onerror — ${e.error}`);\r\n      this.onError?.(e.error);\r\n    };\r\n\r\n    recognition.onend = () => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onend — active: ${this.active}, receivedResult: ${this.receivedResult}`);\r\n\r\n      if (this.active) {\r\n        // Speech API paused — trigger correction\r\n        this.onPause?.();\r\n        // Restart for continued streaming\r\n        try {\r\n          recognition.start();\r\n          this.log('[SSM] restarted after pause');\r\n        } catch (err) {\r\n          this.log(`[SSM] restart THREW: ${err}`);\r\n          this.recognition = null;\r\n          this.onError?.('Speech recognition failed to restart after pause.');\r\n        }\r\n      } else {\r\n        this.recognition = null;\r\n      }\r\n    };\r\n\r\n    this.recognition = recognition;\r\n    try {\r\n      recognition.start();\r\n      this.log('[SSM] recognition.start() succeeded');\r\n    } catch (err) {\r\n      this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n      this.recognition = null;\r\n      this.active = false;\r\n      this.clearNoResultTimer();\r\n      this.onError?.(\r\n        `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n      );\r\n      return Promise.resolve(); // Resolve so engine can proceed to getUserMedia\r\n    }\r\n\r\n    return micClaimPromise;\r\n  }\r\n\r\n  private clearNoResultTimer(): void {\r\n    if (this.noResultTimer) {\r\n      clearTimeout(this.noResultTimer);\r\n      this.noResultTimer = null;\r\n    }\r\n  }\r\n\r\n  /** Stop streaming recognition and return accumulated text. */\r\n  stop(): string {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.stop();\r\n    }\r\n    const result = this.accumulated;\r\n    this.accumulated = '';\r\n    return result;\r\n  }\r\n\r\n  /** Abort immediately without returning text. */\r\n  destroy(): void {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.abort();\r\n    }\r\n    this.accumulated = '';\r\n    this.onTranscript = null;\r\n    this.onPause = null;\r\n    this.onError = null;\r\n    this.onDebug = null;\r\n  }\r\n}\r\n","import type {\r\n  STTConfig,\r\n  STTState,\r\n  STTEvents,\r\n  STTStatus,\r\n  ResolvedSTTConfig,\r\n  AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/**\r\n * Main STT engine — the public API for speech-to-text with Whisper correction.\r\n *\r\n * Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n */\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n  private config: ResolvedSTTConfig;\r\n  private workerManager: WorkerManager;\r\n  private correctionOrchestrator: CorrectionOrchestrator;\r\n  private speechStreaming: SpeechStreamingManager;\r\n  private capture: AudioCaptureHandle | null = null;\r\n  private state: STTState;\r\n  private workerUrl?: URL;\r\n\r\n  /**\r\n   * Create a new STT engine instance.\r\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n   */\r\n  constructor(config?: STTConfig, workerUrl?: URL) {\r\n    super();\r\n    this.config = resolveConfig(config);\r\n    this.workerManager = new WorkerManager();\r\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n    this.speechStreaming = new SpeechStreamingManager();\r\n    this.workerUrl = workerUrl;\r\n\r\n    this.state = {\r\n      status: 'idle',\r\n      isModelLoaded: false,\r\n      loadProgress: 0,\r\n      backend: null,\r\n      error: null,\r\n    };\r\n\r\n    this.correctionOrchestrator.setCorrectionFn(() => {\r\n      this.performCorrection();\r\n    });\r\n\r\n    this.setupWorkerListeners();\r\n    this.setupStreamingCallbacks();\r\n  }\r\n\r\n  /** Initialize the engine: spawn worker and load model. */\r\n  async init(): Promise<void> {\r\n    this.updateStatus('loading');\r\n    this.workerManager.spawn(this.workerUrl);\r\n\r\n    try {\r\n      await this.workerManager.loadModel(this.config);\r\n      this.state.isModelLoaded = true;\r\n      this.updateStatus('ready');\r\n    } catch (err) {\r\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n      this.updateStatus('idle');\r\n      throw err;\r\n    }\r\n  }\r\n\r\n  /** Start recording audio and enable correction cycles. */\r\n  async start(): Promise<void> {\r\n    if (this.state.status !== 'ready') {\r\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n    }\r\n\r\n    try {\r\n      // Start Speech API BEFORE getUserMedia and wait for it to claim\r\n      // the mic (onaudiostart or 300ms fallback). Without this wait,\r\n      // getUserMedia opens a competing audio capture and Chrome's\r\n      // SpeechRecognition receives no audio (silent failure).\r\n      this.emitDebug(\r\n        `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\"`,\r\n      );\r\n      if (this.config.streaming.enabled) {\r\n        await this.speechStreaming.start(this.config.language);\r\n        this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n      }\r\n      this.capture = await startCapture();\r\n      this.updateStatus('recording');\r\n      this.correctionOrchestrator.start();\r\n    } catch (err) {\r\n      this.emitError(\r\n        'MIC_DENIED',\r\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n      );\r\n    }\r\n  }\r\n\r\n  /** Stop recording, run final transcription, return text. */\r\n  async stop(): Promise<string> {\r\n    if (!this.capture) return '';\r\n\r\n    this.correctionOrchestrator.stop();\r\n    this.speechStreaming.stop();\r\n    this.workerManager.cancel();\r\n\r\n    this.updateStatus('processing');\r\n\r\n    try {\r\n      const audio = await stopCapture(this.capture);\r\n      this.capture = null;\r\n\r\n      if (audio.length === 0) {\r\n        this.updateStatus('ready');\r\n        return '';\r\n      }\r\n\r\n      const text = await this.workerManager.transcribe(audio);\r\n      this.emit('correction', text);\r\n      this.updateStatus('ready');\r\n      return text;\r\n    } catch (err) {\r\n      this.emitError(\r\n        'TRANSCRIPTION_FAILED',\r\n        err instanceof Error ? err.message : 'Final transcription failed.',\r\n      );\r\n      this.updateStatus('ready');\r\n      return '';\r\n    }\r\n  }\r\n\r\n  /** Destroy the engine: terminate worker, release all resources. */\r\n  destroy(): void {\r\n    this.correctionOrchestrator.stop();\r\n    this.speechStreaming.destroy();\r\n\r\n    if (this.capture) {\r\n      for (const track of this.capture.stream.getTracks()) {\r\n        track.stop();\r\n      }\r\n      this.capture.audioCtx.close().catch(() => {});\r\n      this.capture = null;\r\n    }\r\n\r\n    this.workerManager.destroy();\r\n    this.updateStatus('idle');\r\n    this.removeAllListeners();\r\n  }\r\n\r\n  /** Get current engine state. */\r\n  getState(): Readonly<STTState> {\r\n    return { ...this.state };\r\n  }\r\n\r\n  /** Notify the correction orchestrator of a speech pause. */\r\n  notifyPause(): void {\r\n    this.correctionOrchestrator.onPauseDetected();\r\n  }\r\n\r\n  private async performCorrection(): Promise<void> {\r\n    if (!this.capture || !this.state.isModelLoaded) return;\r\n\r\n    this.workerManager.cancel();\r\n\r\n    try {\r\n      const samples = snapshotAudio(this.capture);\r\n      const nativeSr = this.capture.audioCtx.sampleRate;\r\n      const audio = await resampleAudio(samples, nativeSr);\r\n\r\n      if (audio.length === 0) return;\r\n\r\n      const text = await this.workerManager.transcribe(audio);\r\n      if (text.trim() && this.capture) {\r\n        this.emit('correction', text);\r\n      }\r\n    } catch (err) {\r\n      this.emitError(\r\n        'TRANSCRIPTION_FAILED',\r\n        err instanceof Error ? err.message : 'Correction transcription failed.',\r\n      );\r\n      // Recording continues — error is non-fatal\r\n    }\r\n  }\r\n\r\n  private setupStreamingCallbacks(): void {\r\n    this.speechStreaming.setOnDebug((message) => {\r\n      this.emit('debug', message);\r\n    });\r\n\r\n    this.speechStreaming.setOnTranscript((text) => {\r\n      this.emitDebug(`[STT] transcript callback — \"${text}\"`);\r\n      this.emit('transcript', text);\r\n    });\r\n\r\n    this.speechStreaming.setOnPause(() => {\r\n      this.emitDebug('[STT] pause callback — triggering correction');\r\n      this.correctionOrchestrator.onPauseDetected();\r\n    });\r\n\r\n    this.speechStreaming.setOnError((message) => {\r\n      this.emitDebug(`[STT] streaming error — \"${message}\"`);\r\n      this.emitError('STREAMING_ERROR', message);\r\n    });\r\n  }\r\n\r\n  private setupWorkerListeners(): void {\r\n    this.workerManager.on('progress', (percent) => {\r\n      this.state.loadProgress = percent;\r\n      this.emit('status', { ...this.state });\r\n    });\r\n\r\n    this.workerManager.on('error', (message) => {\r\n      this.emitError('WORKER_ERROR', message);\r\n    });\r\n  }\r\n\r\n  private updateStatus(status: STTStatus): void {\r\n    this.state.status = status;\r\n    this.state.error = null;\r\n    this.emit('status', { ...this.state });\r\n  }\r\n\r\n  private emitError(code: string, message: string): void {\r\n    this.state.error = message;\r\n    this.emit('error', { code, message 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Default: true */\r\n  enabled?: boolean;\r\n  /** Correction engine provider. Default: 'whisper' */\r\n  provider?: STTCorrectionProvider;\r\n  /** Silence duration (ms) before triggering correction. Default: 1000 */\r\n  pauseThreshold?: number;\r\n  /** Maximum interval (ms) between forced corrections. Default: 3000 */\r\n  forcedInterval?: number;\r\n}\r\n\r\n/** Real-time streaming preview configuration. */\r\nexport interface STTStreamingConfig {\r\n  /** Enable real-time streaming transcript. Default: true */\r\n  enabled?: boolean;\r\n  /** Streaming provider. Default: 'web-speech-api' */\r\n  provider?: STTStreamingProvider;\r\n}\r\n\r\n/** Audio chunking configuration for long-form audio. */\r\nexport interface STTChunkingConfig {\r\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\r\n  chunkLengthS?: number;\r\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\r\n  strideLengthS?: number;\r\n}\r\n\r\n/** Full engine configuration. All fields optional — sensible defaults applied. */\r\nexport interface STTConfig {\r\n  /** Whisper model size. Default: 'tiny' */\r\n  model?: STTModelSize;\r\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\r\n  backend?: STTBackend;\r\n  /** Transcription language. Default: 'en' */\r\n  language?: string;\r\n  /** Model quantization dtype. Default: 'q4' */\r\n  dtype?: string;\r\n  /** Mid-recording correction settings. */\r\n  correction?: STTCorrectionConfig;\r\n  /** Audio chunking settings for long-form audio. */\r\n  chunking?: STTChunkingConfig;\r\n  /** Web Speech API streaming preview settings. */\r\n  streaming?: STTStreamingConfig;\r\n}\r\n\r\n/** Resolved configuration with all defaults applied. */\r\nexport interface ResolvedSTTConfig {\r\n  model: STTModelSize;\r\n  backend: STTBackend;\r\n  language: string;\r\n  dtype: string;\r\n  correction: Required<STTCorrectionConfig>;\r\n  chunking: Required<STTChunkingConfig>;\r\n  streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/** Engine state exposed to consumers via status events. */\r\nexport interface STTState {\r\n  status: STTStatus;\r\n  isModelLoaded: boolean;\r\n  /** Model download progress (0–100). */\r\n  loadProgress: number;\r\n  /** Active compute backend, or null if not yet determined. */\r\n  backend: 'webgpu' | 'wasm' | null;\r\n  error: string | null;\r\n}\r\n\r\n/** Structured error emitted via the 'error' event. */\r\nexport interface STTError {\r\n  code: string;\r\n  message: string;\r\n}\r\n\r\n/** Event map for the typed event emitter. */\r\nexport type STTEvents = {\r\n  /** Streaming interim text during recording. */\r\n  transcript: (text: string) => void;\r\n  /** Whisper-corrected text replacing interim text. */\r\n  correction: (text: string) => void;\r\n  /** Actionable error (mic denied, model fail, transcription fail). */\r\n  error: (error: STTError) => void;\r\n  /** Engine state change. */\r\n  status: (state: STTState) => void;\r\n  /** Diagnostic log for debugging (subscribe to capture all internal events). */\r\n  debug: (message: string) => void;\r\n};\r\n\r\n/** Handle returned by audio capture — used internally. */\r\nexport interface AudioCaptureHandle {\r\n  audioCtx: AudioContext;\r\n  stream: MediaStream;\r\n  samples: Float32Array[];\r\n  /** Retain reference to prevent GC from stopping audio processing. */\r\n  _processor: ScriptProcessorNode;\r\n  /** Source node for disconnect/reconnect on pause/resume. */\r\n  _source: MediaStreamAudioSourceNode;\r\n  /** Gain node (silent) to prevent mic playback. */\r\n  _silencer: GainNode;\r\n}\r\n\r\n/** Message sent from main thread to Whisper worker. */\r\nexport interface WorkerMessage {\r\n  type: 'load' | 'transcribe' | 'cancel';\r\n  audio?: Float32Array;\r\n  config?: {\r\n    model: string;\r\n    backend: STTBackend;\r\n    language: string;\r\n    dtype: string;\r\n    chunkLengthS: number;\r\n    strideLengthS: number;\r\n  };\r\n}\r\n\r\n/** Response sent from Whisper worker to main thread. */\r\nexport interface WorkerResponse {\r\n  type: 'progress' | 'ready' | 'result' | 'error';\r\n  data?: unknown;\r\n}\r\n\r\n/** Default configuration values. */\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n  model: 'tiny',\r\n  backend: 'auto',\r\n  language: 'en',\r\n  dtype: 'q4',\r\n  correction: {\r\n    enabled: true,\r\n    provider: 'whisper',\r\n    pauseThreshold: 1_000,\r\n    forcedInterval: 3_000,\r\n  },\r\n  chunking: {\r\n    chunkLengthS: 30,\r\n    strideLengthS: 5,\r\n  },\r\n  streaming: {\r\n    enabled: true,\r\n    provider: 'web-speech-api',\r\n  },\r\n};\r\n\r\n/** Merge user config with defaults to produce resolved config. */\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n  return {\r\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n    correction: {\r\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n      pauseThreshold:\r\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n      forcedInterval:\r\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n    },\r\n    chunking: {\r\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n    },\r\n    streaming: {\r\n      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n    },\r\n  };\r\n}\r\n","/**\n * A generic, typed event emitter.\n *\n * Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n */\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n  private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n  /** Subscribe to an event. */\n  on<K extends keyof T>(event: K, listener: T[K]): void {\n    let set = this.listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.listeners.set(event, set);\n    }\n    set.add(listener as T[keyof T]);\n  }\n\n  /** Unsubscribe a specific listener. No-op if not registered. */\n  off<K extends keyof T>(event: K, listener: T[K]): void {\n    this.listeners.get(event)?.delete(listener as T[keyof T]);\n  }\n\n  /** Emit an event, calling all registered listeners in insertion order. */\n  emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n    const set = this.listeners.get(event);\n    if (!set) return;\n    for (const listener of set) {\n      (listener as (...a: Parameters<T[K]>) => void)(...args);\n    }\n  }\n\n  /** Remove all listeners, optionally for a single event. */\n  removeAllListeners(event?: keyof T): void {\n    if (event !== undefined) {\n      this.listeners.delete(event);\n    } else {\n      this.listeners.clear();\n    }\n  }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/**\r\n * Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n */\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n  const stream = await navigator.mediaDevices.getUserMedia({\r\n    audio: { channelCount: 1 },\r\n  });\r\n  const audioCtx = new AudioContext();\r\n\r\n  // Chrome may suspend AudioContext — must resume within user gesture\r\n  if (audioCtx.state === 'suspended') {\r\n    await audioCtx.resume();\r\n  }\r\n\r\n  const source = audioCtx.createMediaStreamSource(stream);\r\n  const samples: Float32Array[] = [];\r\n\r\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n  };\r\n\r\n  // Connect through a silent gain node so mic audio doesn't play back\r\n  const silencer = audioCtx.createGain();\r\n  silencer.gain.value = 0;\r\n  source.connect(processor);\r\n  processor.connect(silencer);\r\n  silencer.connect(audioCtx.destination);\r\n\r\n  return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };\r\n}\r\n\r\n/**\r\n * Pause capture without releasing mic or AudioContext.\r\n * Disconnects the audio source so no new samples are collected.\r\n * Returns resampled audio from the recording period.\r\n * Call resumeCapture() to start collecting again.\r\n */\r\nexport async function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n  capture._source.disconnect();\r\n  const currentSamples = [...capture.samples];\r\n  capture.samples.length = 0;\r\n  return resampleAudio(currentSamples, capture.audioCtx.sampleRate);\r\n}\r\n\r\n/**\r\n * Resume a paused capture. Reconnects the audio source to the processor.\r\n * AudioContext is resumed if suspended.\r\n */\r\nexport async function resumeCapture(capture: AudioCaptureHandle): Promise<void> {\r\n  if (capture.audioCtx.state === 'suspended') {\r\n    await capture.audioCtx.resume();\r\n  }\r\n  capture._source.connect(capture._processor);\r\n}\r\n\r\n/**\r\n * Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n */\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n  return [...capture.samples];\r\n}\r\n\r\n/**\r\n * Concatenate sample chunks and resample to 16kHz for Whisper.\r\n */\r\nexport async function resampleAudio(\r\n  samples: Float32Array[],\r\n  nativeSr: number,\r\n): Promise<Float32Array> {\r\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n  if (totalLength === 0) return new Float32Array(0);\r\n\r\n  const fullAudio = new Float32Array(totalLength);\r\n  let offset = 0;\r\n  for (const s of samples) {\r\n    fullAudio.set(s, offset);\r\n    offset += s.length;\r\n  }\r\n\r\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n  const duration = fullAudio.length / nativeSr;\r\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\r\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n  buffer.getChannelData(0).set(fullAudio);\r\n  const src = offline.createBufferSource();\r\n  src.buffer = buffer;\r\n  src.connect(offline.destination);\r\n  src.start(0);\r\n  const resampled = await offline.startRendering();\r\n  return resampled.getChannelData(0);\r\n}\r\n\r\n/**\r\n * Stop capturing and return resampled audio at 16kHz.\r\n */\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n  const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n  // Disconnect processor to stop capturing\r\n  try {\r\n    _processor.disconnect();\r\n  } catch {\r\n    /* already disconnected */\r\n  }\r\n\r\n  // Stop microphone tracks\r\n  for (const track of stream.getTracks()) {\r\n    track.stop();\r\n  }\r\n\r\n  const nativeSr = audioCtx.sampleRate;\r\n  await audioCtx.close();\r\n\r\n  return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/** Events emitted by the WorkerManager. */\nexport type WorkerManagerEvents = {\n  progress: (percent: number) => void;\n  ready: () => void;\n  result: (text: string) => void;\n  error: (message: string) => void;\n};\n\n/**\n * Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n */\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n  private worker: Worker | null = null;\n  private transcribeResolve: ((text: string) => void) | null = null;\n  private currentTranscribePromise: Promise<string> | null = null;\n  private modelReadyResolve: (() => void) | null = null;\n  private modelReadyReject: ((err: Error) => void) | null = null;\n\n  /** True while a transcription job is running in the worker. */\n  get isTranscribing(): boolean {\n    return this.transcribeResolve !== null;\n  }\n\n  /** Await the current in-flight transcription without starting a new one. */\n  awaitCurrentTranscription(): Promise<string> {\n    return this.currentTranscribePromise ?? Promise.resolve('');\n  }\n\n  /** Spawn the Web Worker. Must be called before loadModel/transcribe. */\n  spawn(workerUrl?: URL): void {\n    if (this.worker) return;\n\n    const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n    this.worker = new Worker(url, { type: 'module' });\n    this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n      this.handleMessage(e.data);\n    };\n    this.worker.onerror = (e: ErrorEvent) => {\n      this.emit('error', e.message ?? 'Worker error');\n    };\n  }\n\n  /** Load the Whisper model in the worker. Resolves when ready. */\n  async loadModel(config: ResolvedSTTConfig): Promise<void> {\n    if (!this.worker) throw new Error('Worker not spawned');\n\n    return new Promise<void>((resolve, reject) => {\n      this.modelReadyResolve = resolve;\n      this.modelReadyReject = reject;\n      this.worker!.postMessage({\n        type: 'load',\n        config: {\n          model: config.model,\n          backend: config.backend,\n          language: config.language,\n          dtype: config.dtype,\n          chunkLengthS: config.chunking.chunkLengthS,\n          strideLengthS: config.chunking.strideLengthS,\n        },\n      });\n    });\n  }\n\n  /** Send audio to the worker for transcription. Resolves with text. */\n  async transcribe(audio: Float32Array): Promise<string> {\n    if (!this.worker) throw new Error('Worker not spawned');\n    if (audio.length === 0) return '';\n\n    this.currentTranscribePromise = new Promise<string>((resolve) => {\n      this.transcribeResolve = resolve;\n      this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n    });\n    return this.currentTranscribePromise;\n  }\n\n  /** Cancel any in-flight transcription. */\n  cancel(): void {\n    this.worker?.postMessage({ type: 'cancel' });\n    if (this.transcribeResolve) {\n      this.transcribeResolve('');\n      this.transcribeResolve = null;\n    }\n  }\n\n  /** Terminate the worker and release resources. */\n  destroy(): void {\n    this.cancel();\n    this.worker?.terminate();\n    this.worker = null;\n    this.removeAllListeners();\n  }\n\n  private handleMessage(msg: WorkerResponse): void {\n    switch (msg.type) {\n      case 'progress':\n        this.emit('progress', msg.data as number);\n        break;\n      case 'ready':\n        this.emit('ready');\n        this.modelReadyResolve?.();\n        this.modelReadyResolve = null;\n        this.modelReadyReject = null;\n        break;\n      case 'result':\n        this.emit('result', msg.data as string);\n        this.transcribeResolve?.(msg.data as string);\n        this.transcribeResolve = null;\n        break;\n      case 'error': {\n        const errMsg = msg.data as string;\n        this.emit('error', errMsg);\n        // Reject model load if still pending\n        if (this.modelReadyReject) {\n          this.modelReadyReject(new Error(errMsg));\n          this.modelReadyResolve = null;\n          this.modelReadyReject = null;\n        }\n        // Resolve transcribe with empty string on error\n        if (this.transcribeResolve) {\n          this.transcribeResolve('');\n          this.transcribeResolve = null;\n        }\n        break;\n      }\n    }\n  }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/**\n * Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n */\nexport class CorrectionOrchestrator {\n  private forcedTimer: ReturnType<typeof setInterval> | null = null;\n  private lastCorrectionTime = 0;\n  private correctionFn: (() => void) | null = null;\n  private config: ResolvedSTTConfig['correction'];\n\n  /** Create a new correction orchestrator with the given timing config. */\n  constructor(config: ResolvedSTTConfig['correction']) {\n    this.config = config;\n  }\n\n  /** Set the function to call when a correction is triggered. */\n  setCorrectionFn(fn: () => void): void {\n    this.correctionFn = fn;\n  }\n\n  /** Start the correction orchestrator (begin forced interval timer). */\n  start(): void {\n    if (!this.config.enabled) return;\n\n    this.lastCorrectionTime = Date.now();\n    this.startForcedTimer();\n  }\n\n  /** Stop the orchestrator (clear all timers). */\n  stop(): void {\n    this.stopForcedTimer();\n  }\n\n  /** Called when a speech pause is detected. Triggers correction if cooldown elapsed. */\n  onPauseDetected(): void {\n    if (!this.config.enabled) return;\n\n    const now = Date.now();\n    if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n    this.triggerCorrection();\n  }\n\n  /** Force a correction now (resets timer). */\n  forceCorrection(): void {\n    this.triggerCorrection();\n  }\n\n  private triggerCorrection(): void {\n    this.lastCorrectionTime = Date.now();\n    this.correctionFn?.();\n    // Reset forced timer after any correction\n    this.restartForcedTimer();\n  }\n\n  private startForcedTimer(): void {\n    this.stopForcedTimer();\n    this.forcedTimer = setInterval(() => {\n      this.triggerCorrection();\n    }, this.config.forcedInterval);\n  }\n\n  private stopForcedTimer(): void {\n    if (this.forcedTimer) {\n      clearInterval(this.forcedTimer);\n      this.forcedTimer = null;\n    }\n  }\n\n  private restartForcedTimer(): void {\n    if (this.forcedTimer) {\n      this.startForcedTimer();\n    }\n  }\n}\n","/* ─── Web Speech API types ──────────────────────────────── */\r\n\r\ninterface SpeechRecognitionEvent {\r\n  results: SpeechRecognitionResultList;\r\n  resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n  error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n  continuous: boolean;\r\n  interimResults: boolean;\r\n  lang: string;\r\n  onaudiostart: (() => void) | null;\r\n  onresult: ((e: SpeechRecognitionEvent) => void) | null;\r\n  onerror: ((e: SpeechRecognitionErrorEvent) => void) | null;\r\n  onend: (() => void) | null;\r\n  start: () => void;\r\n  stop: () => void;\r\n  abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor | null {\r\n  if (typeof globalThis === 'undefined') return null;\r\n  const w = globalThis as unknown as Record<string, unknown>;\r\n  return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor | null;\r\n}\r\n\r\n/* ─── Language mapping ──────────────────────────────────── */\r\n\r\n/** Map Whisper language codes to BCP-47 locale tags for the Speech API. */\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n  en: 'en-US',\r\n  english: 'en-US',\r\n  zh: 'zh-CN',\r\n  chinese: 'zh-CN',\r\n  de: 'de-DE',\r\n  german: 'de-DE',\r\n  es: 'es-ES',\r\n  spanish: 'es-ES',\r\n  ru: 'ru-RU',\r\n  russian: 'ru-RU',\r\n  ko: 'ko-KR',\r\n  korean: 'ko-KR',\r\n  fr: 'fr-FR',\r\n  french: 'fr-FR',\r\n  ja: 'ja-JP',\r\n  japanese: 'ja-JP',\r\n  pt: 'pt-BR',\r\n  portuguese: 'pt-BR',\r\n  tr: 'tr-TR',\r\n  turkish: 'tr-TR',\r\n  pl: 'pl-PL',\r\n  polish: 'pl-PL',\r\n  nl: 'nl-NL',\r\n  dutch: 'nl-NL',\r\n  ar: 'ar-SA',\r\n  arabic: 'ar-SA',\r\n  sv: 'sv-SE',\r\n  swedish: 'sv-SE',\r\n  it: 'it-IT',\r\n  italian: 'it-IT',\r\n  id: 'id-ID',\r\n  indonesian: 'id-ID',\r\n  hi: 'hi-IN',\r\n  hindi: 'hi-IN',\r\n  fi: 'fi-FI',\r\n  finnish: 'fi-FI',\r\n  vi: 'vi-VN',\r\n  vietnamese: 'vi-VN',\r\n  he: 'he-IL',\r\n  hebrew: 'he-IL',\r\n  uk: 'uk-UA',\r\n  ukrainian: 'uk-UA',\r\n  el: 'el-GR',\r\n  greek: 'el-GR',\r\n  ms: 'ms-MY',\r\n  malay: 'ms-MY',\r\n  cs: 'cs-CZ',\r\n  czech: 'cs-CZ',\r\n  ro: 'ro-RO',\r\n  romanian: 'ro-RO',\r\n  da: 'da-DK',\r\n  danish: 'da-DK',\r\n  hu: 'hu-HU',\r\n  hungarian: 'hu-HU',\r\n  no: 'nb-NO',\r\n  norwegian: 'nb-NO',\r\n  th: 'th-TH',\r\n  thai: 'th-TH',\r\n};\r\n\r\n/**\r\n * Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n */\r\nfunction toBCP47(language: string): string {\r\n  if (language.includes('-')) return language;\r\n  return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/* ─── SpeechStreamingManager ────────────────────────────── */\r\n\r\n/**\r\n * Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n */\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n\r\nexport class SpeechStreamingManager {\r\n  private recognition: SpeechRecognitionInstance | null = null;\r\n  private accumulated = '';\r\n  private active = false;\r\n  private receivedResult = false;\r\n  private noResultTimer: ReturnType<typeof setTimeout> | null = null;\r\n  private onTranscript: ((text: string) => void) | null = null;\r\n  private onPause: (() => void) | null = null;\r\n  private onError: ((message: string) => void) | null = null;\r\n  private onDebug: ((message: string) => void) | null = null;\r\n\r\n  /** Check if the Web Speech API is available in this environment. */\r\n  static isSupported(): boolean {\r\n    return getSpeechRecognition() !== null;\r\n  }\r\n\r\n  /** Set callback for streaming transcript updates (interim + final text). */\r\n  setOnTranscript(fn: (text: string) => void): void {\r\n    this.onTranscript = fn;\r\n  }\r\n\r\n  /** Set callback for speech pause detection (Speech API onend). */\r\n  setOnPause(fn: () => void): void {\r\n    this.onPause = fn;\r\n  }\r\n\r\n  /** Set callback for errors. */\r\n  setOnError(fn: (message: string) => void): void {\r\n    this.onError = fn;\r\n  }\r\n\r\n  /** Set callback for diagnostic debug messages. */\r\n  setOnDebug(fn: (message: string) => void): void {\r\n    this.onDebug = fn;\r\n  }\r\n\r\n  private log(message: string): void {\r\n    this.onDebug?.(message);\r\n    console.warn(message);\r\n  }\r\n\r\n  /**\r\n   * Start streaming recognition. Returns a Promise that resolves once\r\n   * SpeechRecognition has claimed the microphone (onaudiostart) or after\r\n   * a 300ms fallback — whichever comes first. The engine should await\r\n   * this before calling getUserMedia to avoid dual-mic conflicts.\r\n   *\r\n   * When skipMicWait is true (warm restart — mic already active), returns\r\n   * immediately after calling recognition.start() without waiting for\r\n   * onaudiostart.\r\n   */\r\n  start(language: string, skipMicWait = false): Promise<void> {\r\n    const SR = getSpeechRecognition();\r\n    if (!SR) {\r\n      this.log('[SSM] SpeechRecognition not available in this environment');\r\n      return Promise.resolve();\r\n    }\r\n\r\n    const bcp47 = toBCP47(language);\r\n    this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n    this.accumulated = '';\r\n    this.active = true;\r\n    this.receivedResult = false;\r\n\r\n    const recognition = new SR();\r\n    recognition.continuous = true;\r\n    recognition.interimResults = true;\r\n    recognition.lang = bcp47;\r\n\r\n    let lastFinalIndex = -1;\r\n    let lastFinalText = '';\r\n\r\n    // Promise resolves when SR claims mic (onaudiostart) or after fallback.\r\n    // This ensures getUserMedia doesn't compete for the mic.\r\n    let micReady = false;\r\n    const micClaimPromise = new Promise<void>((resolve) => {\r\n      recognition.onaudiostart = () => {\r\n        if (this.recognition !== recognition) return;\r\n        this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n        if (!micReady) {\r\n          micReady = true;\r\n          resolve();\r\n        }\r\n      };\r\n      // Fallback: resolve after 300ms even if onaudiostart never fires\r\n      setTimeout(() => {\r\n        if (!micReady) {\r\n          micReady = true;\r\n          this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n          resolve();\r\n        }\r\n      }, 300);\r\n    });\r\n\r\n    // Detect silent failure: if no onresult fires within timeout, emit error\r\n    this.clearNoResultTimer();\r\n    this.noResultTimer = setTimeout(() => {\r\n      if (this.active && !this.receivedResult) {\r\n        this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n        this.onError?.(\r\n          'Speech streaming started but received no results. ' +\r\n            'Mic may be blocked by another audio capture.',\r\n        );\r\n      }\r\n    }, NO_RESULT_TIMEOUT_MS);\r\n\r\n    recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.receivedResult = true;\r\n      this.clearNoResultTimer();\r\n\r\n      let final_ = '';\r\n      let interim = '';\r\n      for (let i = e.resultIndex; i < e.results.length; i++) {\r\n        const t = e.results[i][0].transcript;\r\n        if (e.results[i].isFinal) {\r\n          if (i > lastFinalIndex) {\r\n            final_ += t;\r\n            lastFinalIndex = i;\r\n          }\r\n        } else {\r\n          interim += t;\r\n        }\r\n      }\r\n\r\n      this.log(\r\n        `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n      );\r\n\r\n      if (final_ && final_.trim() !== lastFinalText) {\r\n        lastFinalText = final_.trim();\r\n        this.accumulated = this.accumulated\r\n          ? this.accumulated + ' ' + final_.trim()\r\n          : final_.trim();\r\n        this.onTranscript?.(this.accumulated);\r\n      } else if (interim) {\r\n        const trimmed = interim.trimStart();\r\n        const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n        this.onTranscript?.(full);\r\n      }\r\n    };\r\n\r\n    recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onerror — ${e.error}`);\r\n      this.onError?.(e.error);\r\n    };\r\n\r\n    recognition.onend = () => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onend — active: ${this.active}, receivedResult: ${this.receivedResult}`);\r\n\r\n      if (this.active) {\r\n        // Speech API paused — trigger correction\r\n        this.onPause?.();\r\n        // Restart for continued streaming\r\n        try {\r\n          recognition.start();\r\n          this.log('[SSM] restarted after pause');\r\n        } catch (err) {\r\n          this.log(`[SSM] restart THREW: ${err}`);\r\n          this.recognition = null;\r\n          this.onError?.('Speech recognition failed to restart after pause.');\r\n        }\r\n      } else {\r\n        this.recognition = null;\r\n      }\r\n    };\r\n\r\n    this.recognition = recognition;\r\n    try {\r\n      recognition.start();\r\n      this.log('[SSM] recognition.start() succeeded');\r\n    } catch (err) {\r\n      this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n      this.recognition = null;\r\n      this.active = false;\r\n      this.clearNoResultTimer();\r\n      this.onError?.(\r\n        `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n      );\r\n      return Promise.resolve(); // Resolve so engine can proceed to getUserMedia\r\n    }\r\n\r\n    if (skipMicWait) {\r\n      this.log('[SSM] skipMicWait — warm restart, returning immediately');\r\n      return Promise.resolve();\r\n    }\r\n\r\n    return micClaimPromise;\r\n  }\r\n\r\n  private clearNoResultTimer(): void {\r\n    if (this.noResultTimer) {\r\n      clearTimeout(this.noResultTimer);\r\n      this.noResultTimer = null;\r\n    }\r\n  }\r\n\r\n  /** Stop streaming recognition and return accumulated text. */\r\n  stop(): string {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.stop();\r\n    }\r\n    const result = this.accumulated;\r\n    this.accumulated = '';\r\n    return result;\r\n  }\r\n\r\n  /** Abort immediately without returning text. */\r\n  destroy(): void {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.abort();\r\n    }\r\n    this.accumulated = '';\r\n    this.onTranscript = null;\r\n    this.onPause = null;\r\n    this.onError = null;\r\n    this.onDebug = null;\r\n  }\r\n}\r\n","import type {\r\n  STTConfig,\r\n  STTState,\r\n  STTEvents,\r\n  STTStatus,\r\n  ResolvedSTTConfig,\r\n  AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, pauseCapture, resumeCapture, snapshotAudio, resampleAudio } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/**\r\n * Main STT engine — the public API for speech-to-text with Whisper correction.\r\n *\r\n * Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n */\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n  private config: ResolvedSTTConfig;\r\n  private workerManager: WorkerManager;\r\n  private correctionOrchestrator: CorrectionOrchestrator;\r\n  private speechStreaming: SpeechStreamingManager;\r\n  private capture: AudioCaptureHandle | null = null;\r\n  private state: STTState;\r\n  private workerUrl?: URL;\r\n  /** Prevents performCorrection from emitting while stop() is consuming the in-flight result. */\r\n  private _stopping = false;\r\n\r\n  /**\r\n   * Create a new STT engine instance.\r\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n   */\r\n  constructor(config?: STTConfig, workerUrl?: URL) {\r\n    super();\r\n    this.config = resolveConfig(config);\r\n    this.workerManager = new WorkerManager();\r\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n    this.speechStreaming = new SpeechStreamingManager();\r\n    this.workerUrl = workerUrl;\r\n\r\n    this.state = {\r\n      status: 'idle',\r\n      isModelLoaded: false,\r\n      loadProgress: 0,\r\n      backend: null,\r\n      error: null,\r\n    };\r\n\r\n    this.correctionOrchestrator.setCorrectionFn(() => {\r\n      this.performCorrection();\r\n    });\r\n\r\n    this.setupWorkerListeners();\r\n    this.setupStreamingCallbacks();\r\n  }\r\n\r\n  /** Initialize the engine: spawn worker and load model. */\r\n  async init(): Promise<void> {\r\n    this.updateStatus('loading');\r\n    this.workerManager.spawn(this.workerUrl);\r\n\r\n    try {\r\n      await this.workerManager.loadModel(this.config);\r\n      this.state.isModelLoaded = true;\r\n      this.updateStatus('ready');\r\n    } catch (err) {\r\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n      this.updateStatus('idle');\r\n      throw err;\r\n    }\r\n  }\r\n\r\n  /** Start recording audio and enable correction cycles. */\r\n  async start(): Promise<void> {\r\n    if (this.state.status !== 'ready') {\r\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n    }\r\n\r\n    try {\r\n      // Check if mic is already warm from a previous recording session.\r\n      // When warm, skip getUserMedia and the 300ms SR mic-claim wait.\r\n      const warmCapture =\r\n        this.capture &&\r\n        this.capture.stream.getTracks().every((t) => t.readyState === 'live');\r\n\r\n      this.emitDebug(\r\n        `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\", warm: ${!!warmCapture}`,\r\n      );\r\n\r\n      if (this.config.streaming.enabled) {\r\n        // On warm restart, skip the mic-claim wait — no getUserMedia race.\r\n        await this.speechStreaming.start(this.config.language, !!warmCapture);\r\n        if (!warmCapture) {\r\n          this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n        }\r\n      }\r\n\r\n      if (warmCapture) {\r\n        await resumeCapture(this.capture!);\r\n        this.emitDebug('[STT] warm mic resumed — skipped getUserMedia');\r\n      } else {\r\n        // First start or stale capture: full mic init with getUserMedia.\r\n        this.capture = await startCapture();\r\n      }\r\n\r\n      this.updateStatus('recording');\r\n      this.correctionOrchestrator.start();\r\n    } catch (err) {\r\n      this.emitError(\r\n        'MIC_DENIED',\r\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n      );\r\n    }\r\n  }\r\n\r\n  /** Stop recording, run final transcription, return text.\r\n   *  Mic and AudioContext stay alive for fast restart — call destroy() to fully release. */\r\n  async stop(): Promise<string> {\r\n    if (!this.capture) return '';\r\n\r\n    // Prevent any in-flight performCorrection from emitting — stop() will own the final emit.\r\n    this._stopping = true;\r\n    this.correctionOrchestrator.stop();\r\n    this.speechStreaming.stop();\r\n\r\n    this.updateStatus('processing');\r\n\r\n    // If a correction job is already running, reuse it: Whisper can't be interrupted mid-inference\r\n    // anyway, so cancelling just wastes the work and causes a second full run.\r\n    if (this.workerManager.isTranscribing) {\r\n      try {\r\n        // Resample audio and await the in-flight result in parallel.\r\n        const [audio, inFlightText] = await Promise.all([\r\n          pauseCapture(this.capture),\r\n          this.workerManager.awaitCurrentTranscription(),\r\n        ]);\r\n        this._stopping = false;\r\n\r\n        const text = inFlightText.trim();\r\n        if (text) {\r\n          this.emit('correction', text);\r\n          this.updateStatus('ready');\r\n          return text;\r\n        }\r\n\r\n        // In-flight returned empty (cancelled or error) — fall through to fresh transcription.\r\n        if (audio.length > 0) {\r\n          const freshText = await this.workerManager.transcribe(audio);\r\n          this.emit('correction', freshText);\r\n          this.updateStatus('ready');\r\n          return freshText;\r\n        }\r\n\r\n        this.updateStatus('ready');\r\n        return '';\r\n      } catch (err) {\r\n        this._stopping = false;\r\n        this.emitError(\r\n          'TRANSCRIPTION_FAILED',\r\n          err instanceof Error ? err.message : 'Final transcription failed.',\r\n        );\r\n        this.updateStatus('ready');\r\n        return '';\r\n      }\r\n    }\r\n\r\n    // No job in-flight — cancel is a no-op, run fresh transcription.\r\n    this.workerManager.cancel();\r\n    this._stopping = false;\r\n\r\n    try {\r\n      // Soft pause — keeps stream and AudioContext alive for fast restart.\r\n      const audio = await pauseCapture(this.capture);\r\n\r\n      if (audio.length === 0) {\r\n        this.updateStatus('ready');\r\n        return '';\r\n      }\r\n\r\n      const text = await this.workerManager.transcribe(audio);\r\n      this.emit('correction', text);\r\n      this.updateStatus('ready');\r\n      return text;\r\n    } catch (err) {\r\n      this.emitError(\r\n        'TRANSCRIPTION_FAILED',\r\n        err instanceof Error ? err.message : 'Final transcription failed.',\r\n      );\r\n      this.updateStatus('ready');\r\n      return '';\r\n    }\r\n  }\r\n\r\n  /** Destroy the engine: terminate worker, release all resources. */\r\n  destroy(): void {\r\n    this.correctionOrchestrator.stop();\r\n    this.speechStreaming.destroy();\r\n\r\n    if (this.capture) {\r\n      try {\r\n        this.capture._processor.disconnect();\r\n      } catch {\r\n        /* already disconnected */\r\n      }\r\n      for (const track of this.capture.stream.getTracks()) {\r\n        track.stop();\r\n      }\r\n      this.capture.audioCtx.close().catch(() => {});\r\n      this.capture = null;\r\n    }\r\n\r\n    this.workerManager.destroy();\r\n    this.updateStatus('idle');\r\n    this.removeAllListeners();\r\n  }\r\n\r\n  /** Get current engine state. */\r\n  getState(): Readonly<STTState> {\r\n    return { ...this.state };\r\n  }\r\n\r\n  /** Notify the correction orchestrator of a speech pause. */\r\n  notifyPause(): void {\r\n    this.correctionOrchestrator.onPauseDetected();\r\n  }\r\n\r\n  private async performCorrection(): Promise<void> {\r\n    if (!this.capture || !this.state.isModelLoaded) return;\r\n\r\n    this.workerManager.cancel();\r\n\r\n    try {\r\n      const samples = snapshotAudio(this.capture);\r\n      const nativeSr = this.capture.audioCtx.sampleRate;\r\n      const audio = await resampleAudio(samples, nativeSr);\r\n\r\n      if (audio.length === 0) return;\r\n\r\n      const text = await this.workerManager.transcribe(audio);\r\n      if (text.trim() && this.capture && !this._stopping) {\r\n        this.emit('correction', text);\r\n      }\r\n    } catch (err) {\r\n      this.emitError(\r\n        'TRANSCRIPTION_FAILED',\r\n        err instanceof Error ? err.message : 'Correction transcription failed.',\r\n      );\r\n      // Recording continues — error is non-fatal\r\n    }\r\n  }\r\n\r\n  private setupStreamingCallbacks(): void {\r\n    this.speechStreaming.setOnDebug((message) => {\r\n      this.emit('debug', message);\r\n    });\r\n\r\n    this.speechStreaming.setOnTranscript((text) => {\r\n      this.emitDebug(`[STT] transcript callback — \"${text}\"`);\r\n      this.emit('transcript', text);\r\n    });\r\n\r\n    this.speechStreaming.setOnPause(() => {\r\n      this.emitDebug('[STT] pause callback — triggering correction');\r\n      this.correctionOrchestrator.onPauseDetected();\r\n    });\r\n\r\n    this.speechStreaming.setOnError((message) => {\r\n      this.emitDebug(`[STT] streaming error — \"${message}\"`);\r\n      this.emitError('STREAMING_ERROR', message);\r\n    });\r\n  }\r\n\r\n  private setupWorkerListeners(): void {\r\n    this.workerManager.on('progress', (percent) => {\r\n      this.state.loadProgress = percent;\r\n      this.emit('status', { ...this.state });\r\n    });\r\n\r\n    this.workerManager.on('error', (message) => {\r\n      this.emitError('WORKER_ERROR', message);\r\n    });\r\n  }\r\n\r\n  private updateStatus(status: STTStatus): void {\r\n    this.state.status = status;\r\n    this.state.error = null;\r\n    this.emit('status', { ...this.state });\r\n  }\r\n\r\n  private emitError(code: string, message: string): void {\r\n    this.state.error = message;\r\n    this.emit('error', { code, message });\r\n  }\r\n\r\n  private emitDebug(message: string): void {\r\n    console.warn(message);\r\n    this.emit('debug', message);\r\n  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package/dist/index.d.cts

@@ -14,9 +14,9 @@ interface STTCorrectionConfig {
     enabled?: boolean;
     /** Correction engine provider. Default: 'whisper' */
     provider?: STTCorrectionProvider;
-    /** Silence duration (ms) before triggering correction. Default: 3000 */
+    /** Silence duration (ms) before triggering correction. Default: 1000 */
     pauseThreshold?: number;
-    /** Maximum interval (ms) between forced corrections. Default: 5000 */
+    /** Maximum interval (ms) between forced corrections. Default: 3000 */
     forcedInterval?: number;
 }
 /** Real-time streaming preview configuration. */
@@ -95,6 +95,10 @@ interface AudioCaptureHandle {
     samples: Float32Array[];
     /** Retain reference to prevent GC from stopping audio processing. */
     _processor: ScriptProcessorNode;
+    /** Source node for disconnect/reconnect on pause/resume. */
+    _source: MediaStreamAudioSourceNode;
+    /** Gain node (silent) to prevent mic playback. */
+    _silencer: GainNode;
 }
 /** Default configuration values. */
 declare const DEFAULT_STT_CONFIG: ResolvedSTTConfig;
@@ -124,6 +128,18 @@ declare class TypedEventEmitter<T extends Record<string, (...args: any[]) => voi
  * Uses ScriptProcessorNode to collect Float32Array samples directly.
  */
 declare function startCapture(): Promise<AudioCaptureHandle>;
+/**
+ * Pause capture without releasing mic or AudioContext.
+ * Disconnects the audio source so no new samples are collected.
+ * Returns resampled audio from the recording period.
+ * Call resumeCapture() to start collecting again.
+ */
+declare function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array>;
+/**
+ * Resume a paused capture. Reconnects the audio source to the processor.
+ * AudioContext is resumed if suspended.
+ */
+declare function resumeCapture(capture: AudioCaptureHandle): Promise<void>;
 /**
  * Copy current audio buffer without stopping capture.
  * Returns a shallow copy of the samples array (each chunk is shared, not cloned).
@@ -152,8 +168,13 @@ type WorkerManagerEvents = {
 declare class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {
     private worker;
     private transcribeResolve;
+    private currentTranscribePromise;
     private modelReadyResolve;
     private modelReadyReject;
+    /** True while a transcription job is running in the worker. */
+    get isTranscribing(): boolean;
+    /** Await the current in-flight transcription without starting a new one. */
+    awaitCurrentTranscription(): Promise<string>;
     /** Spawn the Web Worker. Must be called before loadModel/transcribe. */
     spawn(workerUrl?: URL): void;
     /** Load the Whisper model in the worker. Resolves when ready. */
@@ -220,8 +241,12 @@ declare class SpeechStreamingManager {
      * SpeechRecognition has claimed the microphone (onaudiostart) or after
      * a 300ms fallback — whichever comes first. The engine should await
      * this before calling getUserMedia to avoid dual-mic conflicts.
+     *
+     * When skipMicWait is true (warm restart — mic already active), returns
+     * immediately after calling recognition.start() without waiting for
+     * onaudiostart.
      */
-    start(language: string): Promise<void>;
+    start(language: string, skipMicWait?: boolean): Promise<void>;
     private clearNoResultTimer;
     /** Stop streaming recognition and return accumulated text. */
     stop(): string;
@@ -250,6 +275,8 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     private capture;
     private state;
     private workerUrl?;
+    /** Prevents performCorrection from emitting while stop() is consuming the in-flight result. */
+    private _stopping;
     /**
      * Create a new STT engine instance.
      * @param config - Optional configuration overrides (model, backend, language, etc.).
@@ -260,7 +287,8 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     init(): Promise<void>;
     /** Start recording audio and enable correction cycles. */
     start(): Promise<void>;
-    /** Stop recording, run final transcription, return text. */
+    /** Stop recording, run final transcription, return text.
+     *  Mic and AudioContext stay alive for fast restart — call destroy() to fully release. */
     stop(): Promise<string>;
     /** Destroy the engine: terminate worker, release all resources. */
     destroy(): void;
@@ -276,4 +304,4 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     private emitDebug;
 }
 
-export { type AudioCaptureHandle, CorrectionOrchestrator, DEFAULT_STT_CONFIG, type ResolvedSTTConfig, type STTBackend, type STTChunkingConfig, type STTConfig, type STTCorrectionConfig, type STTCorrectionProvider, STTEngine, type STTError, type STTEvents, type STTModelSize, type STTState, type STTStatus, type STTStreamingConfig, type STTStreamingProvider, SpeechStreamingManager, TypedEventEmitter, WorkerManager, type WorkerManagerEvents, resampleAudio, resolveConfig, snapshotAudio, startCapture, stopCapture };
+export { type AudioCaptureHandle, CorrectionOrchestrator, DEFAULT_STT_CONFIG, type ResolvedSTTConfig, type STTBackend, type STTChunkingConfig, type STTConfig, type STTCorrectionConfig, type STTCorrectionProvider, STTEngine, type STTError, type STTEvents, type STTModelSize, type STTState, type STTStatus, type STTStreamingConfig, type STTStreamingProvider, SpeechStreamingManager, TypedEventEmitter, WorkerManager, type WorkerManagerEvents, pauseCapture, resampleAudio, resolveConfig, resumeCapture, snapshotAudio, startCapture, stopCapture };

package/dist/index.d.ts

@@ -14,9 +14,9 @@ interface STTCorrectionConfig {
     enabled?: boolean;
     /** Correction engine provider. Default: 'whisper' */
     provider?: STTCorrectionProvider;
-    /** Silence duration (ms) before triggering correction. Default: 3000 */
+    /** Silence duration (ms) before triggering correction. Default: 1000 */
     pauseThreshold?: number;
-    /** Maximum interval (ms) between forced corrections. Default: 5000 */
+    /** Maximum interval (ms) between forced corrections. Default: 3000 */
     forcedInterval?: number;
 }
 /** Real-time streaming preview configuration. */
@@ -95,6 +95,10 @@ interface AudioCaptureHandle {
     samples: Float32Array[];
     /** Retain reference to prevent GC from stopping audio processing. */
     _processor: ScriptProcessorNode;
+    /** Source node for disconnect/reconnect on pause/resume. */
+    _source: MediaStreamAudioSourceNode;
+    /** Gain node (silent) to prevent mic playback. */
+    _silencer: GainNode;
 }
 /** Default configuration values. */
 declare const DEFAULT_STT_CONFIG: ResolvedSTTConfig;
@@ -124,6 +128,18 @@ declare class TypedEventEmitter<T extends Record<string, (...args: any[]) => voi
  * Uses ScriptProcessorNode to collect Float32Array samples directly.
  */
 declare function startCapture(): Promise<AudioCaptureHandle>;
+/**
+ * Pause capture without releasing mic or AudioContext.
+ * Disconnects the audio source so no new samples are collected.
+ * Returns resampled audio from the recording period.
+ * Call resumeCapture() to start collecting again.
+ */
+declare function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array>;
+/**
+ * Resume a paused capture. Reconnects the audio source to the processor.
+ * AudioContext is resumed if suspended.
+ */
+declare function resumeCapture(capture: AudioCaptureHandle): Promise<void>;
 /**
  * Copy current audio buffer without stopping capture.
  * Returns a shallow copy of the samples array (each chunk is shared, not cloned).
@@ -152,8 +168,13 @@ type WorkerManagerEvents = {
 declare class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {
     private worker;
     private transcribeResolve;
+    private currentTranscribePromise;
     private modelReadyResolve;
     private modelReadyReject;
+    /** True while a transcription job is running in the worker. */
+    get isTranscribing(): boolean;
+    /** Await the current in-flight transcription without starting a new one. */
+    awaitCurrentTranscription(): Promise<string>;
     /** Spawn the Web Worker. Must be called before loadModel/transcribe. */
     spawn(workerUrl?: URL): void;
     /** Load the Whisper model in the worker. Resolves when ready. */
@@ -220,8 +241,12 @@ declare class SpeechStreamingManager {
      * SpeechRecognition has claimed the microphone (onaudiostart) or after
      * a 300ms fallback — whichever comes first. The engine should await
      * this before calling getUserMedia to avoid dual-mic conflicts.
+     *
+     * When skipMicWait is true (warm restart — mic already active), returns
+     * immediately after calling recognition.start() without waiting for
+     * onaudiostart.
      */
-    start(language: string): Promise<void>;
+    start(language: string, skipMicWait?: boolean): Promise<void>;
     private clearNoResultTimer;
     /** Stop streaming recognition and return accumulated text. */
     stop(): string;
@@ -250,6 +275,8 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     private capture;
     private state;
     private workerUrl?;
+    /** Prevents performCorrection from emitting while stop() is consuming the in-flight result. */
+    private _stopping;
     /**
      * Create a new STT engine instance.
      * @param config - Optional configuration overrides (model, backend, language, etc.).
@@ -260,7 +287,8 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     init(): Promise<void>;
     /** Start recording audio and enable correction cycles. */
     start(): Promise<void>;
-    /** Stop recording, run final transcription, return text. */
+    /** Stop recording, run final transcription, return text.
+     *  Mic and AudioContext stay alive for fast restart — call destroy() to fully release. */
     stop(): Promise<string>;
     /** Destroy the engine: terminate worker, release all resources. */
     destroy(): void;
@@ -276,4 +304,4 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     private emitDebug;
 }
 
-export { type AudioCaptureHandle, CorrectionOrchestrator, DEFAULT_STT_CONFIG, type ResolvedSTTConfig, type STTBackend, type STTChunkingConfig, type STTConfig, type STTCorrectionConfig, type STTCorrectionProvider, STTEngine, type STTError, type STTEvents, type STTModelSize, type STTState, type STTStatus, type STTStreamingConfig, type STTStreamingProvider, SpeechStreamingManager, TypedEventEmitter, WorkerManager, type WorkerManagerEvents, resampleAudio, resolveConfig, snapshotAudio, startCapture, stopCapture };
+export { type AudioCaptureHandle, CorrectionOrchestrator, DEFAULT_STT_CONFIG, type ResolvedSTTConfig, type STTBackend, type STTChunkingConfig, type STTConfig, type STTCorrectionConfig, type STTCorrectionProvider, STTEngine, type STTError, type STTEvents, type STTModelSize, type STTState, type STTStatus, type STTStreamingConfig, type STTStreamingProvider, SpeechStreamingManager, TypedEventEmitter, WorkerManager, type WorkerManagerEvents, pauseCapture, resampleAudio, resolveConfig, resumeCapture, snapshotAudio, startCapture, stopCapture };

package/dist/index.js

@@ -7,8 +7,8 @@ var DEFAULT_STT_CONFIG = {
   correction: {
     enabled: true,
     provider: "whisper",
-    pauseThreshold: 3e3,
-    forcedInterval: 5e3
+    pauseThreshold: 1e3,
+    forcedInterval: 3e3
   },
   chunking: {
     chunkLengthS: 30,
@@ -97,7 +97,19 @@ async function startCapture() {
   source.connect(processor);
   processor.connect(silencer);
   silencer.connect(audioCtx.destination);
-  return { audioCtx, stream, samples, _processor: processor };
+  return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };
+}
+async function pauseCapture(capture) {
+  capture._source.disconnect();
+  const currentSamples = [...capture.samples];
+  capture.samples.length = 0;
+  return resampleAudio(currentSamples, capture.audioCtx.sampleRate);
+}
+async function resumeCapture(capture) {
+  if (capture.audioCtx.state === "suspended") {
+    await capture.audioCtx.resume();
+  }
+  capture._source.connect(capture._processor);
 }
 function snapshotAudio(capture) {
   return [...capture.samples];
@@ -142,8 +154,17 @@ async function stopCapture(capture) {
 var WorkerManager = class extends TypedEventEmitter {
   worker = null;
   transcribeResolve = null;
+  currentTranscribePromise = null;
   modelReadyResolve = null;
   modelReadyReject = null;
+  /** True while a transcription job is running in the worker. */
+  get isTranscribing() {
+    return this.transcribeResolve !== null;
+  }
+  /** Await the current in-flight transcription without starting a new one. */
+  awaitCurrentTranscription() {
+    return this.currentTranscribePromise ?? Promise.resolve("");
+  }
   /** Spawn the Web Worker. Must be called before loadModel/transcribe. */
   spawn(workerUrl) {
     if (this.worker) return;
@@ -179,10 +200,11 @@ var WorkerManager = class extends TypedEventEmitter {
   async transcribe(audio) {
     if (!this.worker) throw new Error("Worker not spawned");
     if (audio.length === 0) return "";
-    return new Promise((resolve) => {
+    this.currentTranscribePromise = new Promise((resolve) => {
       this.transcribeResolve = resolve;
       this.worker.postMessage({ type: "transcribe", audio }, [audio.buffer]);
     });
+    return this.currentTranscribePromise;
   }
   /** Cancel any in-flight transcription. */
   cancel() {
@@ -402,8 +424,12 @@ var SpeechStreamingManager = class {
    * SpeechRecognition has claimed the microphone (onaudiostart) or after
    * a 300ms fallback — whichever comes first. The engine should await
    * this before calling getUserMedia to avoid dual-mic conflicts.
+   *
+   * When skipMicWait is true (warm restart — mic already active), returns
+   * immediately after calling recognition.start() without waiting for
+   * onaudiostart.
    */
-  start(language) {
+  start(language, skipMicWait = false) {
     const SR = getSpeechRecognition();
     if (!SR) {
       this.log("[SSM] SpeechRecognition not available in this environment");
@@ -513,6 +539,10 @@ var SpeechStreamingManager = class {
       );
       return Promise.resolve();
     }
+    if (skipMicWait) {
+      this.log("[SSM] skipMicWait \u2014 warm restart, returning immediately");
+      return Promise.resolve();
+    }
     return micClaimPromise;
   }
   clearNoResultTimer() {
@@ -560,6 +590,8 @@ var STTEngine = class extends TypedEventEmitter {
   capture = null;
   state;
   workerUrl;
+  /** Prevents performCorrection from emitting while stop() is consuming the in-flight result. */
+  _stopping = false;
   /**
    * Create a new STT engine instance.
    * @param config - Optional configuration overrides (model, backend, language, etc.).
@@ -605,14 +637,22 @@ var STTEngine = class extends TypedEventEmitter {
       throw new Error(`Cannot start: engine is "${this.state.status}", expected "ready"`);
     }
     try {
+      const warmCapture = this.capture && this.capture.stream.getTracks().every((t) => t.readyState === "live");
       this.emitDebug(
-        `[STT] start() \u2014 streaming: ${this.config.streaming.enabled}, lang: "${this.config.language}"`
+        `[STT] start() \u2014 streaming: ${this.config.streaming.enabled}, lang: "${this.config.language}", warm: ${!!warmCapture}`
       );
       if (this.config.streaming.enabled) {
-        await this.speechStreaming.start(this.config.language);
-        this.emitDebug("[STT] Speech API mic claim complete \u2014 starting getUserMedia");
+        await this.speechStreaming.start(this.config.language, !!warmCapture);
+        if (!warmCapture) {
+          this.emitDebug("[STT] Speech API mic claim complete \u2014 starting getUserMedia");
+        }
+      }
+      if (warmCapture) {
+        await resumeCapture(this.capture);
+        this.emitDebug("[STT] warm mic resumed \u2014 skipped getUserMedia");
+      } else {
+        this.capture = await startCapture();
       }
-      this.capture = await startCapture();
       this.updateStatus("recording");
       this.correctionOrchestrator.start();
     } catch (err) {
@@ -622,16 +662,49 @@ var STTEngine = class extends TypedEventEmitter {
       );
     }
   }
-  /** Stop recording, run final transcription, return text. */
+  /** Stop recording, run final transcription, return text.
+   *  Mic and AudioContext stay alive for fast restart — call destroy() to fully release. */
   async stop() {
     if (!this.capture) return "";
+    this._stopping = true;
     this.correctionOrchestrator.stop();
     this.speechStreaming.stop();
-    this.workerManager.cancel();
     this.updateStatus("processing");
+    if (this.workerManager.isTranscribing) {
+      try {
+        const [audio, inFlightText] = await Promise.all([
+          pauseCapture(this.capture),
+          this.workerManager.awaitCurrentTranscription()
+        ]);
+        this._stopping = false;
+        const text = inFlightText.trim();
+        if (text) {
+          this.emit("correction", text);
+          this.updateStatus("ready");
+          return text;
+        }
+        if (audio.length > 0) {
+          const freshText = await this.workerManager.transcribe(audio);
+          this.emit("correction", freshText);
+          this.updateStatus("ready");
+          return freshText;
+        }
+        this.updateStatus("ready");
+        return "";
+      } catch (err) {
+        this._stopping = false;
+        this.emitError(
+          "TRANSCRIPTION_FAILED",
+          err instanceof Error ? err.message : "Final transcription failed."
+        );
+        this.updateStatus("ready");
+        return "";
+      }
+    }
+    this.workerManager.cancel();
+    this._stopping = false;
     try {
-      const audio = await stopCapture(this.capture);
-      this.capture = null;
+      const audio = await pauseCapture(this.capture);
       if (audio.length === 0) {
         this.updateStatus("ready");
         return "";
@@ -654,6 +727,10 @@ var STTEngine = class extends TypedEventEmitter {
     this.correctionOrchestrator.stop();
     this.speechStreaming.destroy();
     if (this.capture) {
+      try {
+        this.capture._processor.disconnect();
+      } catch {
+      }
       for (const track of this.capture.stream.getTracks()) {
         track.stop();
       }
@@ -682,7 +759,7 @@ var STTEngine = class extends TypedEventEmitter {
       const audio = await resampleAudio(samples, nativeSr);
       if (audio.length === 0) return;
       const text = await this.workerManager.transcribe(audio);
-      if (text.trim() && this.capture) {
+      if (text.trim() && this.capture && !this._stopping) {
         this.emit("correction", text);
       }
     } catch (err) {
@@ -739,8 +816,10 @@ export {
   SpeechStreamingManager,
   TypedEventEmitter,
   WorkerManager,
+  pauseCapture,
   resampleAudio,
   resolveConfig,
+  resumeCapture,
   snapshotAudio,
   startCapture,
   stopCapture

package/dist/index.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/types.ts","../src/event-emitter.ts","../src/audio-capture.ts","../src/worker-manager.ts","../src/correction-orchestrator.ts","../src/speech-streaming.ts","../src/stt-engine.ts"],"sourcesContent":["/** Supported Whisper model sizes. */\r\nexport type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';\r\n\r\n/** Supported compute backends. */\r\nexport type STTBackend = 'webgpu' | 'wasm' | 'auto';\r\n\r\n/** Engine lifecycle states. */\r\nexport type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';\r\n\r\n/** Supported correction engine providers. */\r\nexport type STTCorrectionProvider = 'whisper';\r\n\r\n/** Supported real-time streaming providers. */\r\nexport type STTStreamingProvider = 'web-speech-api';\r\n\r\n/** Correction engine configuration. */\r\nexport interface STTCorrectionConfig {\r\n  /** Enable mid-recording correction. Default: true */\r\n  enabled?: boolean;\r\n  /** Correction engine provider. Default: 'whisper' */\r\n  provider?: STTCorrectionProvider;\r\n  /** Silence duration (ms) before triggering correction. Default: 3000 */\r\n  pauseThreshold?: number;\r\n  /** Maximum interval (ms) between forced corrections. Default: 5000 */\r\n  forcedInterval?: number;\r\n}\r\n\r\n/** Real-time streaming preview configuration. */\r\nexport interface STTStreamingConfig {\r\n  /** Enable real-time streaming transcript. Default: true */\r\n  enabled?: boolean;\r\n  /** Streaming provider. Default: 'web-speech-api' */\r\n  provider?: STTStreamingProvider;\r\n}\r\n\r\n/** Audio chunking configuration for long-form audio. */\r\nexport interface STTChunkingConfig {\r\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\r\n  chunkLengthS?: number;\r\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\r\n  strideLengthS?: number;\r\n}\r\n\r\n/** Full engine configuration. All fields optional — sensible defaults applied. */\r\nexport interface STTConfig {\r\n  /** Whisper model size. Default: 'tiny' */\r\n  model?: STTModelSize;\r\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\r\n  backend?: STTBackend;\r\n  /** Transcription language. Default: 'en' */\r\n  language?: string;\r\n  /** Model quantization dtype. Default: 'q4' */\r\n  dtype?: string;\r\n  /** Mid-recording correction settings. */\r\n  correction?: STTCorrectionConfig;\r\n  /** Audio chunking settings for long-form audio. */\r\n  chunking?: STTChunkingConfig;\r\n  /** Web Speech API streaming preview settings. */\r\n  streaming?: STTStreamingConfig;\r\n}\r\n\r\n/** Resolved configuration with all defaults applied. */\r\nexport interface ResolvedSTTConfig {\r\n  model: STTModelSize;\r\n  backend: STTBackend;\r\n  language: string;\r\n  dtype: string;\r\n  correction: Required<STTCorrectionConfig>;\r\n  chunking: Required<STTChunkingConfig>;\r\n  streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/** Engine state exposed to consumers via status events. */\r\nexport interface STTState {\r\n  status: STTStatus;\r\n  isModelLoaded: boolean;\r\n  /** Model download progress (0–100). */\r\n  loadProgress: number;\r\n  /** Active compute backend, or null if not yet determined. */\r\n  backend: 'webgpu' | 'wasm' | null;\r\n  error: string | null;\r\n}\r\n\r\n/** Structured error emitted via the 'error' event. */\r\nexport interface STTError {\r\n  code: string;\r\n  message: string;\r\n}\r\n\r\n/** Event map for the typed event emitter. */\r\nexport type STTEvents = {\r\n  /** Streaming interim text during recording. */\r\n  transcript: (text: string) => void;\r\n  /** Whisper-corrected text replacing interim text. */\r\n  correction: (text: string) => void;\r\n  /** Actionable error (mic denied, model fail, transcription fail). */\r\n  error: (error: STTError) => void;\r\n  /** Engine state change. */\r\n  status: (state: STTState) => void;\r\n  /** Diagnostic log for debugging (subscribe to capture all internal events). */\r\n  debug: (message: string) => void;\r\n};\r\n\r\n/** Handle returned by audio capture — used internally. */\r\nexport interface AudioCaptureHandle {\r\n  audioCtx: AudioContext;\r\n  stream: MediaStream;\r\n  samples: Float32Array[];\r\n  /** Retain reference to prevent GC from stopping audio processing. */\r\n  _processor: ScriptProcessorNode;\r\n}\r\n\r\n/** Message sent from main thread to Whisper worker. */\r\nexport interface WorkerMessage {\r\n  type: 'load' | 'transcribe' | 'cancel';\r\n  audio?: Float32Array;\r\n  config?: {\r\n    model: string;\r\n    backend: STTBackend;\r\n    language: string;\r\n    dtype: string;\r\n    chunkLengthS: number;\r\n    strideLengthS: number;\r\n  };\r\n}\r\n\r\n/** Response sent from Whisper worker to main thread. */\r\nexport interface WorkerResponse {\r\n  type: 'progress' | 'ready' | 'result' | 'error';\r\n  data?: unknown;\r\n}\r\n\r\n/** Default configuration values. */\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n  model: 'tiny',\r\n  backend: 'auto',\r\n  language: 'en',\r\n  dtype: 'q4',\r\n  correction: {\r\n    enabled: true,\r\n    provider: 'whisper',\r\n    pauseThreshold: 3_000,\r\n    forcedInterval: 5_000,\r\n  },\r\n  chunking: {\r\n    chunkLengthS: 30,\r\n    strideLengthS: 5,\r\n  },\r\n  streaming: {\r\n    enabled: true,\r\n    provider: 'web-speech-api',\r\n  },\r\n};\r\n\r\n/** Merge user config with defaults to produce resolved config. */\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n  return {\r\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n    correction: {\r\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n      pauseThreshold:\r\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n      forcedInterval:\r\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n    },\r\n    chunking: {\r\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n    },\r\n    streaming: {\r\n      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n    },\r\n  };\r\n}\r\n","/**\n * A generic, typed event emitter.\n *\n * Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n */\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n  private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n  /** Subscribe to an event. */\n  on<K extends keyof T>(event: K, listener: T[K]): void {\n    let set = this.listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.listeners.set(event, set);\n    }\n    set.add(listener as T[keyof T]);\n  }\n\n  /** Unsubscribe a specific listener. No-op if not registered. */\n  off<K extends keyof T>(event: K, listener: T[K]): void {\n    this.listeners.get(event)?.delete(listener as T[keyof T]);\n  }\n\n  /** Emit an event, calling all registered listeners in insertion order. */\n  emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n    const set = this.listeners.get(event);\n    if (!set) return;\n    for (const listener of set) {\n      (listener as (...a: Parameters<T[K]>) => void)(...args);\n    }\n  }\n\n  /** Remove all listeners, optionally for a single event. */\n  removeAllListeners(event?: keyof T): void {\n    if (event !== undefined) {\n      this.listeners.delete(event);\n    } else {\n      this.listeners.clear();\n    }\n  }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/**\r\n * Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n */\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n  const stream = await navigator.mediaDevices.getUserMedia({\r\n    audio: { channelCount: 1 },\r\n  });\r\n  const audioCtx = new AudioContext();\r\n\r\n  // Chrome may suspend AudioContext — must resume within user gesture\r\n  if (audioCtx.state === 'suspended') {\r\n    await audioCtx.resume();\r\n  }\r\n\r\n  const source = audioCtx.createMediaStreamSource(stream);\r\n  const samples: Float32Array[] = [];\r\n\r\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n  };\r\n\r\n  // Connect through a silent gain node so mic audio doesn't play back\r\n  const silencer = audioCtx.createGain();\r\n  silencer.gain.value = 0;\r\n  source.connect(processor);\r\n  processor.connect(silencer);\r\n  silencer.connect(audioCtx.destination);\r\n\r\n  return { audioCtx, stream, samples, _processor: processor };\r\n}\r\n\r\n/**\r\n * Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n */\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n  return [...capture.samples];\r\n}\r\n\r\n/**\r\n * Concatenate sample chunks and resample to 16kHz for Whisper.\r\n */\r\nexport async function resampleAudio(\r\n  samples: Float32Array[],\r\n  nativeSr: number,\r\n): Promise<Float32Array> {\r\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n  if (totalLength === 0) return new Float32Array(0);\r\n\r\n  const fullAudio = new Float32Array(totalLength);\r\n  let offset = 0;\r\n  for (const s of samples) {\r\n    fullAudio.set(s, offset);\r\n    offset += s.length;\r\n  }\r\n\r\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n  const duration = fullAudio.length / nativeSr;\r\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\r\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n  buffer.getChannelData(0).set(fullAudio);\r\n  const src = offline.createBufferSource();\r\n  src.buffer = buffer;\r\n  src.connect(offline.destination);\r\n  src.start(0);\r\n  const resampled = await offline.startRendering();\r\n  return resampled.getChannelData(0);\r\n}\r\n\r\n/**\r\n * Stop capturing and return resampled audio at 16kHz.\r\n */\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n  const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n  // Disconnect processor to stop capturing\r\n  try {\r\n    _processor.disconnect();\r\n  } catch {\r\n    /* already disconnected */\r\n  }\r\n\r\n  // Stop microphone tracks\r\n  for (const track of stream.getTracks()) {\r\n    track.stop();\r\n  }\r\n\r\n  const nativeSr = audioCtx.sampleRate;\r\n  await audioCtx.close();\r\n\r\n  return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/** Events emitted by the WorkerManager. */\nexport type WorkerManagerEvents = {\n  progress: (percent: number) => void;\n  ready: () => void;\n  result: (text: string) => void;\n  error: (message: string) => void;\n};\n\n/**\n * Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n */\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n  private worker: Worker | null = null;\n  private transcribeResolve: ((text: string) => void) | null = null;\n  private modelReadyResolve: (() => void) | null = null;\n  private modelReadyReject: ((err: Error) => void) | null = null;\n\n  /** Spawn the Web Worker. Must be called before loadModel/transcribe. */\n  spawn(workerUrl?: URL): void {\n    if (this.worker) return;\n\n    const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n    this.worker = new Worker(url, { type: 'module' });\n    this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n      this.handleMessage(e.data);\n    };\n    this.worker.onerror = (e: ErrorEvent) => {\n      this.emit('error', e.message ?? 'Worker error');\n    };\n  }\n\n  /** Load the Whisper model in the worker. Resolves when ready. */\n  async loadModel(config: ResolvedSTTConfig): Promise<void> {\n    if (!this.worker) throw new Error('Worker not spawned');\n\n    return new Promise<void>((resolve, reject) => {\n      this.modelReadyResolve = resolve;\n      this.modelReadyReject = reject;\n      this.worker!.postMessage({\n        type: 'load',\n        config: {\n          model: config.model,\n          backend: config.backend,\n          language: config.language,\n          dtype: config.dtype,\n          chunkLengthS: config.chunking.chunkLengthS,\n          strideLengthS: config.chunking.strideLengthS,\n        },\n      });\n    });\n  }\n\n  /** Send audio to the worker for transcription. Resolves with text. */\n  async transcribe(audio: Float32Array): Promise<string> {\n    if (!this.worker) throw new Error('Worker not spawned');\n    if (audio.length === 0) return '';\n\n    return new Promise<string>((resolve) => {\n      this.transcribeResolve = resolve;\n      this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n    });\n  }\n\n  /** Cancel any in-flight transcription. */\n  cancel(): void {\n    this.worker?.postMessage({ type: 'cancel' });\n    if (this.transcribeResolve) {\n      this.transcribeResolve('');\n      this.transcribeResolve = null;\n    }\n  }\n\n  /** Terminate the worker and release resources. */\n  destroy(): void {\n    this.cancel();\n    this.worker?.terminate();\n    this.worker = null;\n    this.removeAllListeners();\n  }\n\n  private handleMessage(msg: WorkerResponse): void {\n    switch (msg.type) {\n      case 'progress':\n        this.emit('progress', msg.data as number);\n        break;\n      case 'ready':\n        this.emit('ready');\n        this.modelReadyResolve?.();\n        this.modelReadyResolve = null;\n        this.modelReadyReject = null;\n        break;\n      case 'result':\n        this.emit('result', msg.data as string);\n        this.transcribeResolve?.(msg.data as string);\n        this.transcribeResolve = null;\n        break;\n      case 'error': {\n        const errMsg = msg.data as string;\n        this.emit('error', errMsg);\n        // Reject model load if still pending\n        if (this.modelReadyReject) {\n          this.modelReadyReject(new Error(errMsg));\n          this.modelReadyResolve = null;\n          this.modelReadyReject = null;\n        }\n        // Resolve transcribe with empty string on error\n        if (this.transcribeResolve) {\n          this.transcribeResolve('');\n          this.transcribeResolve = null;\n        }\n        break;\n      }\n    }\n  }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/**\n * Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n */\nexport class CorrectionOrchestrator {\n  private forcedTimer: ReturnType<typeof setInterval> | null = null;\n  private lastCorrectionTime = 0;\n  private correctionFn: (() => void) | null = null;\n  private config: ResolvedSTTConfig['correction'];\n\n  /** Create a new correction orchestrator with the given timing config. */\n  constructor(config: ResolvedSTTConfig['correction']) {\n    this.config = config;\n  }\n\n  /** Set the function to call when a correction is triggered. */\n  setCorrectionFn(fn: () => void): void {\n    this.correctionFn = fn;\n  }\n\n  /** Start the correction orchestrator (begin forced interval timer). */\n  start(): void {\n    if (!this.config.enabled) return;\n\n    this.lastCorrectionTime = Date.now();\n    this.startForcedTimer();\n  }\n\n  /** Stop the orchestrator (clear all timers). */\n  stop(): void {\n    this.stopForcedTimer();\n  }\n\n  /** Called when a speech pause is detected. Triggers correction if cooldown elapsed. */\n  onPauseDetected(): void {\n    if (!this.config.enabled) return;\n\n    const now = Date.now();\n    if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n    this.triggerCorrection();\n  }\n\n  /** Force a correction now (resets timer). */\n  forceCorrection(): void {\n    this.triggerCorrection();\n  }\n\n  private triggerCorrection(): void {\n    this.lastCorrectionTime = Date.now();\n    this.correctionFn?.();\n    // Reset forced timer after any correction\n    this.restartForcedTimer();\n  }\n\n  private startForcedTimer(): void {\n    this.stopForcedTimer();\n    this.forcedTimer = setInterval(() => {\n      this.triggerCorrection();\n    }, this.config.forcedInterval);\n  }\n\n  private stopForcedTimer(): void {\n    if (this.forcedTimer) {\n      clearInterval(this.forcedTimer);\n      this.forcedTimer = null;\n    }\n  }\n\n  private restartForcedTimer(): void {\n    if (this.forcedTimer) {\n      this.startForcedTimer();\n    }\n  }\n}\n","/* ─── Web Speech API types ──────────────────────────────── */\r\n\r\ninterface SpeechRecognitionEvent {\r\n  results: SpeechRecognitionResultList;\r\n  resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n  error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n  continuous: boolean;\r\n  interimResults: boolean;\r\n  lang: string;\r\n  onaudiostart: (() => void) | null;\r\n  onresult: ((e: SpeechRecognitionEvent) => void) | null;\r\n  onerror: ((e: SpeechRecognitionErrorEvent) => void) | null;\r\n  onend: (() => void) | null;\r\n  start: () => void;\r\n  stop: () => void;\r\n  abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor | null {\r\n  if (typeof globalThis === 'undefined') return null;\r\n  const w = globalThis as unknown as Record<string, unknown>;\r\n  return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor | null;\r\n}\r\n\r\n/* ─── Language mapping ──────────────────────────────────── */\r\n\r\n/** Map Whisper language codes to BCP-47 locale tags for the Speech API. */\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n  en: 'en-US',\r\n  english: 'en-US',\r\n  zh: 'zh-CN',\r\n  chinese: 'zh-CN',\r\n  de: 'de-DE',\r\n  german: 'de-DE',\r\n  es: 'es-ES',\r\n  spanish: 'es-ES',\r\n  ru: 'ru-RU',\r\n  russian: 'ru-RU',\r\n  ko: 'ko-KR',\r\n  korean: 'ko-KR',\r\n  fr: 'fr-FR',\r\n  french: 'fr-FR',\r\n  ja: 'ja-JP',\r\n  japanese: 'ja-JP',\r\n  pt: 'pt-BR',\r\n  portuguese: 'pt-BR',\r\n  tr: 'tr-TR',\r\n  turkish: 'tr-TR',\r\n  pl: 'pl-PL',\r\n  polish: 'pl-PL',\r\n  nl: 'nl-NL',\r\n  dutch: 'nl-NL',\r\n  ar: 'ar-SA',\r\n  arabic: 'ar-SA',\r\n  sv: 'sv-SE',\r\n  swedish: 'sv-SE',\r\n  it: 'it-IT',\r\n  italian: 'it-IT',\r\n  id: 'id-ID',\r\n  indonesian: 'id-ID',\r\n  hi: 'hi-IN',\r\n  hindi: 'hi-IN',\r\n  fi: 'fi-FI',\r\n  finnish: 'fi-FI',\r\n  vi: 'vi-VN',\r\n  vietnamese: 'vi-VN',\r\n  he: 'he-IL',\r\n  hebrew: 'he-IL',\r\n  uk: 'uk-UA',\r\n  ukrainian: 'uk-UA',\r\n  el: 'el-GR',\r\n  greek: 'el-GR',\r\n  ms: 'ms-MY',\r\n  malay: 'ms-MY',\r\n  cs: 'cs-CZ',\r\n  czech: 'cs-CZ',\r\n  ro: 'ro-RO',\r\n  romanian: 'ro-RO',\r\n  da: 'da-DK',\r\n  danish: 'da-DK',\r\n  hu: 'hu-HU',\r\n  hungarian: 'hu-HU',\r\n  no: 'nb-NO',\r\n  norwegian: 'nb-NO',\r\n  th: 'th-TH',\r\n  thai: 'th-TH',\r\n};\r\n\r\n/**\r\n * Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n */\r\nfunction toBCP47(language: string): string {\r\n  if (language.includes('-')) return language;\r\n  return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/* ─── SpeechStreamingManager ────────────────────────────── */\r\n\r\n/**\r\n * Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n */\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n\r\nexport class SpeechStreamingManager {\r\n  private recognition: SpeechRecognitionInstance | null = null;\r\n  private accumulated = '';\r\n  private active = false;\r\n  private receivedResult = false;\r\n  private noResultTimer: ReturnType<typeof setTimeout> | null = null;\r\n  private onTranscript: ((text: string) => void) | null = null;\r\n  private onPause: (() => void) | null = null;\r\n  private onError: ((message: string) => void) | null = null;\r\n  private onDebug: ((message: string) => void) | null = null;\r\n\r\n  /** Check if the Web Speech API is available in this environment. */\r\n  static isSupported(): boolean {\r\n    return getSpeechRecognition() !== null;\r\n  }\r\n\r\n  /** Set callback for streaming transcript updates (interim + final text). */\r\n  setOnTranscript(fn: (text: string) => void): void {\r\n    this.onTranscript = fn;\r\n  }\r\n\r\n  /** Set callback for speech pause detection (Speech API onend). */\r\n  setOnPause(fn: () => void): void {\r\n    this.onPause = fn;\r\n  }\r\n\r\n  /** Set callback for errors. */\r\n  setOnError(fn: (message: string) => void): void {\r\n    this.onError = fn;\r\n  }\r\n\r\n  /** Set callback for diagnostic debug messages. */\r\n  setOnDebug(fn: (message: string) => void): void {\r\n    this.onDebug = fn;\r\n  }\r\n\r\n  private log(message: string): void {\r\n    this.onDebug?.(message);\r\n    console.warn(message);\r\n  }\r\n\r\n  /**\r\n   * Start streaming recognition. Returns a Promise that resolves once\r\n   * SpeechRecognition has claimed the microphone (onaudiostart) or after\r\n   * a 300ms fallback — whichever comes first. The engine should await\r\n   * this before calling getUserMedia to avoid dual-mic conflicts.\r\n   */\r\n  start(language: string): Promise<void> {\r\n    const SR = getSpeechRecognition();\r\n    if (!SR) {\r\n      this.log('[SSM] SpeechRecognition not available in this environment');\r\n      return Promise.resolve();\r\n    }\r\n\r\n    const bcp47 = toBCP47(language);\r\n    this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n    this.accumulated = '';\r\n    this.active = true;\r\n    this.receivedResult = false;\r\n\r\n    const recognition = new SR();\r\n    recognition.continuous = true;\r\n    recognition.interimResults = true;\r\n    recognition.lang = bcp47;\r\n\r\n    let lastFinalIndex = -1;\r\n    let lastFinalText = '';\r\n\r\n    // Promise resolves when SR claims mic (onaudiostart) or after fallback.\r\n    // This ensures getUserMedia doesn't compete for the mic.\r\n    let micReady = false;\r\n    const micClaimPromise = new Promise<void>((resolve) => {\r\n      recognition.onaudiostart = () => {\r\n        if (this.recognition !== recognition) return;\r\n        this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n        if (!micReady) {\r\n          micReady = true;\r\n          resolve();\r\n        }\r\n      };\r\n      // Fallback: resolve after 300ms even if onaudiostart never fires\r\n      setTimeout(() => {\r\n        if (!micReady) {\r\n          micReady = true;\r\n          this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n          resolve();\r\n        }\r\n      }, 300);\r\n    });\r\n\r\n    // Detect silent failure: if no onresult fires within timeout, emit error\r\n    this.clearNoResultTimer();\r\n    this.noResultTimer = setTimeout(() => {\r\n      if (this.active && !this.receivedResult) {\r\n        this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n        this.onError?.(\r\n          'Speech streaming started but received no results. ' +\r\n            'Mic may be blocked by another audio capture.',\r\n        );\r\n      }\r\n    }, NO_RESULT_TIMEOUT_MS);\r\n\r\n    recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.receivedResult = true;\r\n      this.clearNoResultTimer();\r\n\r\n      let final_ = '';\r\n      let interim = '';\r\n      for (let i = e.resultIndex; i < e.results.length; i++) {\r\n        const t = e.results[i][0].transcript;\r\n        if (e.results[i].isFinal) {\r\n          if (i > lastFinalIndex) {\r\n            final_ += t;\r\n            lastFinalIndex = i;\r\n          }\r\n        } else {\r\n          interim += t;\r\n        }\r\n      }\r\n\r\n      this.log(\r\n        `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n      );\r\n\r\n      if (final_ && final_.trim() !== lastFinalText) {\r\n        lastFinalText = final_.trim();\r\n        this.accumulated = this.accumulated\r\n          ? this.accumulated + ' ' + final_.trim()\r\n          : final_.trim();\r\n        this.onTranscript?.(this.accumulated);\r\n      } else if (interim) {\r\n        const trimmed = interim.trimStart();\r\n        const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n        this.onTranscript?.(full);\r\n      }\r\n    };\r\n\r\n    recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onerror — ${e.error}`);\r\n      this.onError?.(e.error);\r\n    };\r\n\r\n    recognition.onend = () => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onend — active: ${this.active}, receivedResult: ${this.receivedResult}`);\r\n\r\n      if (this.active) {\r\n        // Speech API paused — trigger correction\r\n        this.onPause?.();\r\n        // Restart for continued streaming\r\n        try {\r\n          recognition.start();\r\n          this.log('[SSM] restarted after pause');\r\n        } catch (err) {\r\n          this.log(`[SSM] restart THREW: ${err}`);\r\n          this.recognition = null;\r\n          this.onError?.('Speech recognition failed to restart after pause.');\r\n        }\r\n      } else {\r\n        this.recognition = null;\r\n      }\r\n    };\r\n\r\n    this.recognition = recognition;\r\n    try {\r\n      recognition.start();\r\n      this.log('[SSM] recognition.start() succeeded');\r\n    } catch (err) {\r\n      this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n      this.recognition = null;\r\n      this.active = false;\r\n      this.clearNoResultTimer();\r\n      this.onError?.(\r\n        `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n      );\r\n      return Promise.resolve(); // Resolve so engine can proceed to getUserMedia\r\n    }\r\n\r\n    return micClaimPromise;\r\n  }\r\n\r\n  private clearNoResultTimer(): void {\r\n    if (this.noResultTimer) {\r\n      clearTimeout(this.noResultTimer);\r\n      this.noResultTimer = null;\r\n    }\r\n  }\r\n\r\n  /** Stop streaming recognition and return accumulated text. */\r\n  stop(): string {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.stop();\r\n    }\r\n    const result = this.accumulated;\r\n    this.accumulated = '';\r\n    return result;\r\n  }\r\n\r\n  /** Abort immediately without returning text. */\r\n  destroy(): void {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.abort();\r\n    }\r\n    this.accumulated = '';\r\n    this.onTranscript = null;\r\n    this.onPause = null;\r\n    this.onError = null;\r\n    this.onDebug = null;\r\n  }\r\n}\r\n","import type {\r\n  STTConfig,\r\n  STTState,\r\n  STTEvents,\r\n  STTStatus,\r\n  ResolvedSTTConfig,\r\n  AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/**\r\n * Main STT engine — the public API for speech-to-text with Whisper correction.\r\n *\r\n * Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n */\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n  private config: ResolvedSTTConfig;\r\n  private workerManager: WorkerManager;\r\n  private correctionOrchestrator: CorrectionOrchestrator;\r\n  private speechStreaming: SpeechStreamingManager;\r\n  private capture: AudioCaptureHandle | null = null;\r\n  private state: STTState;\r\n  private workerUrl?: URL;\r\n\r\n  /**\r\n   * Create a new STT engine instance.\r\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n   */\r\n  constructor(config?: STTConfig, workerUrl?: URL) {\r\n    super();\r\n    this.config = resolveConfig(config);\r\n    this.workerManager = new WorkerManager();\r\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n    this.speechStreaming = new SpeechStreamingManager();\r\n    this.workerUrl = workerUrl;\r\n\r\n    this.state = {\r\n      status: 'idle',\r\n      isModelLoaded: false,\r\n      loadProgress: 0,\r\n      backend: null,\r\n      error: null,\r\n    };\r\n\r\n    this.correctionOrchestrator.setCorrectionFn(() => {\r\n      this.performCorrection();\r\n    });\r\n\r\n    this.setupWorkerListeners();\r\n    this.setupStreamingCallbacks();\r\n  }\r\n\r\n  /** Initialize the engine: spawn worker and load model. */\r\n  async init(): Promise<void> {\r\n    this.updateStatus('loading');\r\n    this.workerManager.spawn(this.workerUrl);\r\n\r\n    try {\r\n      await this.workerManager.loadModel(this.config);\r\n      this.state.isModelLoaded = true;\r\n      this.updateStatus('ready');\r\n    } catch (err) {\r\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n      this.updateStatus('idle');\r\n      throw err;\r\n    }\r\n  }\r\n\r\n  /** Start recording audio and enable correction cycles. */\r\n  async start(): Promise<void> {\r\n    if (this.state.status !== 'ready') {\r\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n    }\r\n\r\n    try {\r\n      // Start Speech API BEFORE getUserMedia and wait for it to claim\r\n      // the mic (onaudiostart or 300ms fallback). Without this wait,\r\n      // getUserMedia opens a competing audio capture and Chrome's\r\n      // SpeechRecognition receives no audio (silent failure).\r\n      this.emitDebug(\r\n        `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\"`,\r\n      );\r\n      if (this.config.streaming.enabled) {\r\n        await this.speechStreaming.start(this.config.language);\r\n        this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n      }\r\n      this.capture = await startCapture();\r\n      this.updateStatus('recording');\r\n      this.correctionOrchestrator.start();\r\n    } catch (err) {\r\n      this.emitError(\r\n        'MIC_DENIED',\r\n        err instanceof Error ? err.message : 'Microphone access denied. 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Default: 'whisper' */\r\n  provider?: STTCorrectionProvider;\r\n  /** Silence duration (ms) before triggering correction. Default: 1000 */\r\n  pauseThreshold?: number;\r\n  /** Maximum interval (ms) between forced corrections. Default: 3000 */\r\n  forcedInterval?: number;\r\n}\r\n\r\n/** Real-time streaming preview configuration. */\r\nexport interface STTStreamingConfig {\r\n  /** Enable real-time streaming transcript. Default: true */\r\n  enabled?: boolean;\r\n  /** Streaming provider. Default: 'web-speech-api' */\r\n  provider?: STTStreamingProvider;\r\n}\r\n\r\n/** Audio chunking configuration for long-form audio. */\r\nexport interface STTChunkingConfig {\r\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\r\n  chunkLengthS?: number;\r\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\r\n  strideLengthS?: number;\r\n}\r\n\r\n/** Full engine configuration. All fields optional — sensible defaults applied. */\r\nexport interface STTConfig {\r\n  /** Whisper model size. Default: 'tiny' */\r\n  model?: STTModelSize;\r\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\r\n  backend?: STTBackend;\r\n  /** Transcription language. Default: 'en' */\r\n  language?: string;\r\n  /** Model quantization dtype. Default: 'q4' */\r\n  dtype?: string;\r\n  /** Mid-recording correction settings. */\r\n  correction?: STTCorrectionConfig;\r\n  /** Audio chunking settings for long-form audio. */\r\n  chunking?: STTChunkingConfig;\r\n  /** Web Speech API streaming preview settings. */\r\n  streaming?: STTStreamingConfig;\r\n}\r\n\r\n/** Resolved configuration with all defaults applied. */\r\nexport interface ResolvedSTTConfig {\r\n  model: STTModelSize;\r\n  backend: STTBackend;\r\n  language: string;\r\n  dtype: string;\r\n  correction: Required<STTCorrectionConfig>;\r\n  chunking: Required<STTChunkingConfig>;\r\n  streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/** Engine state exposed to consumers via status events. */\r\nexport interface STTState {\r\n  status: STTStatus;\r\n  isModelLoaded: boolean;\r\n  /** Model download progress (0–100). */\r\n  loadProgress: number;\r\n  /** Active compute backend, or null if not yet determined. */\r\n  backend: 'webgpu' | 'wasm' | null;\r\n  error: string | null;\r\n}\r\n\r\n/** Structured error emitted via the 'error' event. */\r\nexport interface STTError {\r\n  code: string;\r\n  message: string;\r\n}\r\n\r\n/** Event map for the typed event emitter. */\r\nexport type STTEvents = {\r\n  /** Streaming interim text during recording. */\r\n  transcript: (text: string) => void;\r\n  /** Whisper-corrected text replacing interim text. */\r\n  correction: (text: string) => void;\r\n  /** Actionable error (mic denied, model fail, transcription fail). */\r\n  error: (error: STTError) => void;\r\n  /** Engine state change. */\r\n  status: (state: STTState) => void;\r\n  /** Diagnostic log for debugging (subscribe to capture all internal events). */\r\n  debug: (message: string) => void;\r\n};\r\n\r\n/** Handle returned by audio capture — used internally. */\r\nexport interface AudioCaptureHandle {\r\n  audioCtx: AudioContext;\r\n  stream: MediaStream;\r\n  samples: Float32Array[];\r\n  /** Retain reference to prevent GC from stopping audio processing. */\r\n  _processor: ScriptProcessorNode;\r\n  /** Source node for disconnect/reconnect on pause/resume. */\r\n  _source: MediaStreamAudioSourceNode;\r\n  /** Gain node (silent) to prevent mic playback. */\r\n  _silencer: GainNode;\r\n}\r\n\r\n/** Message sent from main thread to Whisper worker. */\r\nexport interface WorkerMessage {\r\n  type: 'load' | 'transcribe' | 'cancel';\r\n  audio?: Float32Array;\r\n  config?: {\r\n    model: string;\r\n    backend: STTBackend;\r\n    language: string;\r\n    dtype: string;\r\n    chunkLengthS: number;\r\n    strideLengthS: number;\r\n  };\r\n}\r\n\r\n/** Response sent from Whisper worker to main thread. */\r\nexport interface WorkerResponse {\r\n  type: 'progress' | 'ready' | 'result' | 'error';\r\n  data?: unknown;\r\n}\r\n\r\n/** Default configuration values. */\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n  model: 'tiny',\r\n  backend: 'auto',\r\n  language: 'en',\r\n  dtype: 'q4',\r\n  correction: {\r\n    enabled: true,\r\n    provider: 'whisper',\r\n    pauseThreshold: 1_000,\r\n    forcedInterval: 3_000,\r\n  },\r\n  chunking: {\r\n    chunkLengthS: 30,\r\n    strideLengthS: 5,\r\n  },\r\n  streaming: {\r\n    enabled: true,\r\n    provider: 'web-speech-api',\r\n  },\r\n};\r\n\r\n/** Merge user config with defaults to produce resolved config. */\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n  return {\r\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n    correction: {\r\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n      pauseThreshold:\r\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n      forcedInterval:\r\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n    },\r\n    chunking: {\r\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n    },\r\n    streaming: {\r\n      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n    },\r\n  };\r\n}\r\n","/**\n * A generic, typed event emitter.\n *\n * Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n */\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n  private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n  /** Subscribe to an event. */\n  on<K extends keyof T>(event: K, listener: T[K]): void {\n    let set = this.listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.listeners.set(event, set);\n    }\n    set.add(listener as T[keyof T]);\n  }\n\n  /** Unsubscribe a specific listener. No-op if not registered. */\n  off<K extends keyof T>(event: K, listener: T[K]): void {\n    this.listeners.get(event)?.delete(listener as T[keyof T]);\n  }\n\n  /** Emit an event, calling all registered listeners in insertion order. */\n  emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n    const set = this.listeners.get(event);\n    if (!set) return;\n    for (const listener of set) {\n      (listener as (...a: Parameters<T[K]>) => void)(...args);\n    }\n  }\n\n  /** Remove all listeners, optionally for a single event. */\n  removeAllListeners(event?: keyof T): void {\n    if (event !== undefined) {\n      this.listeners.delete(event);\n    } else {\n      this.listeners.clear();\n    }\n  }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/**\r\n * Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n */\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n  const stream = await navigator.mediaDevices.getUserMedia({\r\n    audio: { channelCount: 1 },\r\n  });\r\n  const audioCtx = new AudioContext();\r\n\r\n  // Chrome may suspend AudioContext — must resume within user gesture\r\n  if (audioCtx.state === 'suspended') {\r\n    await audioCtx.resume();\r\n  }\r\n\r\n  const source = audioCtx.createMediaStreamSource(stream);\r\n  const samples: Float32Array[] = [];\r\n\r\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n  };\r\n\r\n  // Connect through a silent gain node so mic audio doesn't play back\r\n  const silencer = audioCtx.createGain();\r\n  silencer.gain.value = 0;\r\n  source.connect(processor);\r\n  processor.connect(silencer);\r\n  silencer.connect(audioCtx.destination);\r\n\r\n  return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };\r\n}\r\n\r\n/**\r\n * Pause capture without releasing mic or AudioContext.\r\n * Disconnects the audio source so no new samples are collected.\r\n * Returns resampled audio from the recording period.\r\n * Call resumeCapture() to start collecting again.\r\n */\r\nexport async function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n  capture._source.disconnect();\r\n  const currentSamples = [...capture.samples];\r\n  capture.samples.length = 0;\r\n  return resampleAudio(currentSamples, capture.audioCtx.sampleRate);\r\n}\r\n\r\n/**\r\n * Resume a paused capture. Reconnects the audio source to the processor.\r\n * AudioContext is resumed if suspended.\r\n */\r\nexport async function resumeCapture(capture: AudioCaptureHandle): Promise<void> {\r\n  if (capture.audioCtx.state === 'suspended') {\r\n    await capture.audioCtx.resume();\r\n  }\r\n  capture._source.connect(capture._processor);\r\n}\r\n\r\n/**\r\n * Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n */\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n  return [...capture.samples];\r\n}\r\n\r\n/**\r\n * Concatenate sample chunks and resample to 16kHz for Whisper.\r\n */\r\nexport async function resampleAudio(\r\n  samples: Float32Array[],\r\n  nativeSr: number,\r\n): Promise<Float32Array> {\r\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n  if (totalLength === 0) return new Float32Array(0);\r\n\r\n  const fullAudio = new Float32Array(totalLength);\r\n  let offset = 0;\r\n  for (const s of samples) {\r\n    fullAudio.set(s, offset);\r\n    offset += s.length;\r\n  }\r\n\r\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n  const duration = fullAudio.length / nativeSr;\r\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\r\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n  buffer.getChannelData(0).set(fullAudio);\r\n  const src = offline.createBufferSource();\r\n  src.buffer = buffer;\r\n  src.connect(offline.destination);\r\n  src.start(0);\r\n  const resampled = await offline.startRendering();\r\n  return resampled.getChannelData(0);\r\n}\r\n\r\n/**\r\n * Stop capturing and return resampled audio at 16kHz.\r\n */\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n  const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n  // Disconnect processor to stop capturing\r\n  try {\r\n    _processor.disconnect();\r\n  } catch {\r\n    /* already disconnected */\r\n  }\r\n\r\n  // Stop microphone tracks\r\n  for (const track of stream.getTracks()) {\r\n    track.stop();\r\n  }\r\n\r\n  const nativeSr = audioCtx.sampleRate;\r\n  await audioCtx.close();\r\n\r\n  return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/** Events emitted by the WorkerManager. */\nexport type WorkerManagerEvents = {\n  progress: (percent: number) => void;\n  ready: () => void;\n  result: (text: string) => void;\n  error: (message: string) => void;\n};\n\n/**\n * Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n */\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n  private worker: Worker | null = null;\n  private transcribeResolve: ((text: string) => void) | null = null;\n  private currentTranscribePromise: Promise<string> | null = null;\n  private modelReadyResolve: (() => void) | null = null;\n  private modelReadyReject: ((err: Error) => void) | null = null;\n\n  /** True while a transcription job is running in the worker. */\n  get isTranscribing(): boolean {\n    return this.transcribeResolve !== null;\n  }\n\n  /** Await the current in-flight transcription without starting a new one. */\n  awaitCurrentTranscription(): Promise<string> {\n    return this.currentTranscribePromise ?? Promise.resolve('');\n  }\n\n  /** Spawn the Web Worker. Must be called before loadModel/transcribe. */\n  spawn(workerUrl?: URL): void {\n    if (this.worker) return;\n\n    const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n    this.worker = new Worker(url, { type: 'module' });\n    this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n      this.handleMessage(e.data);\n    };\n    this.worker.onerror = (e: ErrorEvent) => {\n      this.emit('error', e.message ?? 'Worker error');\n    };\n  }\n\n  /** Load the Whisper model in the worker. Resolves when ready. */\n  async loadModel(config: ResolvedSTTConfig): Promise<void> {\n    if (!this.worker) throw new Error('Worker not spawned');\n\n    return new Promise<void>((resolve, reject) => {\n      this.modelReadyResolve = resolve;\n      this.modelReadyReject = reject;\n      this.worker!.postMessage({\n        type: 'load',\n        config: {\n          model: config.model,\n          backend: config.backend,\n          language: config.language,\n          dtype: config.dtype,\n          chunkLengthS: config.chunking.chunkLengthS,\n          strideLengthS: config.chunking.strideLengthS,\n        },\n      });\n    });\n  }\n\n  /** Send audio to the worker for transcription. Resolves with text. */\n  async transcribe(audio: Float32Array): Promise<string> {\n    if (!this.worker) throw new Error('Worker not spawned');\n    if (audio.length === 0) return '';\n\n    this.currentTranscribePromise = new Promise<string>((resolve) => {\n      this.transcribeResolve = resolve;\n      this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n    });\n    return this.currentTranscribePromise;\n  }\n\n  /** Cancel any in-flight transcription. */\n  cancel(): void {\n    this.worker?.postMessage({ type: 'cancel' });\n    if (this.transcribeResolve) {\n      this.transcribeResolve('');\n      this.transcribeResolve = null;\n    }\n  }\n\n  /** Terminate the worker and release resources. */\n  destroy(): void {\n    this.cancel();\n    this.worker?.terminate();\n    this.worker = null;\n    this.removeAllListeners();\n  }\n\n  private handleMessage(msg: WorkerResponse): void {\n    switch (msg.type) {\n      case 'progress':\n        this.emit('progress', msg.data as number);\n        break;\n      case 'ready':\n        this.emit('ready');\n        this.modelReadyResolve?.();\n        this.modelReadyResolve = null;\n        this.modelReadyReject = null;\n        break;\n      case 'result':\n        this.emit('result', msg.data as string);\n        this.transcribeResolve?.(msg.data as string);\n        this.transcribeResolve = null;\n        break;\n      case 'error': {\n        const errMsg = msg.data as string;\n        this.emit('error', errMsg);\n        // Reject model load if still pending\n        if (this.modelReadyReject) {\n          this.modelReadyReject(new Error(errMsg));\n          this.modelReadyResolve = null;\n          this.modelReadyReject = null;\n        }\n        // Resolve transcribe with empty string on error\n        if (this.transcribeResolve) {\n          this.transcribeResolve('');\n          this.transcribeResolve = null;\n        }\n        break;\n      }\n    }\n  }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/**\n * Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n */\nexport class CorrectionOrchestrator {\n  private forcedTimer: ReturnType<typeof setInterval> | null = null;\n  private lastCorrectionTime = 0;\n  private correctionFn: (() => void) | null = null;\n  private config: ResolvedSTTConfig['correction'];\n\n  /** Create a new correction orchestrator with the given timing config. */\n  constructor(config: ResolvedSTTConfig['correction']) {\n    this.config = config;\n  }\n\n  /** Set the function to call when a correction is triggered. */\n  setCorrectionFn(fn: () => void): void {\n    this.correctionFn = fn;\n  }\n\n  /** Start the correction orchestrator (begin forced interval timer). */\n  start(): void {\n    if (!this.config.enabled) return;\n\n    this.lastCorrectionTime = Date.now();\n    this.startForcedTimer();\n  }\n\n  /** Stop the orchestrator (clear all timers). */\n  stop(): void {\n    this.stopForcedTimer();\n  }\n\n  /** Called when a speech pause is detected. Triggers correction if cooldown elapsed. */\n  onPauseDetected(): void {\n    if (!this.config.enabled) return;\n\n    const now = Date.now();\n    if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n    this.triggerCorrection();\n  }\n\n  /** Force a correction now (resets timer). */\n  forceCorrection(): void {\n    this.triggerCorrection();\n  }\n\n  private triggerCorrection(): void {\n    this.lastCorrectionTime = Date.now();\n    this.correctionFn?.();\n    // Reset forced timer after any correction\n    this.restartForcedTimer();\n  }\n\n  private startForcedTimer(): void {\n    this.stopForcedTimer();\n    this.forcedTimer = setInterval(() => {\n      this.triggerCorrection();\n    }, this.config.forcedInterval);\n  }\n\n  private stopForcedTimer(): void {\n    if (this.forcedTimer) {\n      clearInterval(this.forcedTimer);\n      this.forcedTimer = null;\n    }\n  }\n\n  private restartForcedTimer(): void {\n    if (this.forcedTimer) {\n      this.startForcedTimer();\n    }\n  }\n}\n","/* ─── Web Speech API types ──────────────────────────────── */\r\n\r\ninterface SpeechRecognitionEvent {\r\n  results: SpeechRecognitionResultList;\r\n  resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n  error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n  continuous: boolean;\r\n  interimResults: boolean;\r\n  lang: string;\r\n  onaudiostart: (() => void) | null;\r\n  onresult: ((e: SpeechRecognitionEvent) => void) | null;\r\n  onerror: ((e: SpeechRecognitionErrorEvent) => void) | null;\r\n  onend: (() => void) | null;\r\n  start: () => void;\r\n  stop: () => void;\r\n  abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor | null {\r\n  if (typeof globalThis === 'undefined') return null;\r\n  const w = globalThis as unknown as Record<string, unknown>;\r\n  return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor | null;\r\n}\r\n\r\n/* ─── Language mapping ──────────────────────────────────── */\r\n\r\n/** Map Whisper language codes to BCP-47 locale tags for the Speech API. */\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n  en: 'en-US',\r\n  english: 'en-US',\r\n  zh: 'zh-CN',\r\n  chinese: 'zh-CN',\r\n  de: 'de-DE',\r\n  german: 'de-DE',\r\n  es: 'es-ES',\r\n  spanish: 'es-ES',\r\n  ru: 'ru-RU',\r\n  russian: 'ru-RU',\r\n  ko: 'ko-KR',\r\n  korean: 'ko-KR',\r\n  fr: 'fr-FR',\r\n  french: 'fr-FR',\r\n  ja: 'ja-JP',\r\n  japanese: 'ja-JP',\r\n  pt: 'pt-BR',\r\n  portuguese: 'pt-BR',\r\n  tr: 'tr-TR',\r\n  turkish: 'tr-TR',\r\n  pl: 'pl-PL',\r\n  polish: 'pl-PL',\r\n  nl: 'nl-NL',\r\n  dutch: 'nl-NL',\r\n  ar: 'ar-SA',\r\n  arabic: 'ar-SA',\r\n  sv: 'sv-SE',\r\n  swedish: 'sv-SE',\r\n  it: 'it-IT',\r\n  italian: 'it-IT',\r\n  id: 'id-ID',\r\n  indonesian: 'id-ID',\r\n  hi: 'hi-IN',\r\n  hindi: 'hi-IN',\r\n  fi: 'fi-FI',\r\n  finnish: 'fi-FI',\r\n  vi: 'vi-VN',\r\n  vietnamese: 'vi-VN',\r\n  he: 'he-IL',\r\n  hebrew: 'he-IL',\r\n  uk: 'uk-UA',\r\n  ukrainian: 'uk-UA',\r\n  el: 'el-GR',\r\n  greek: 'el-GR',\r\n  ms: 'ms-MY',\r\n  malay: 'ms-MY',\r\n  cs: 'cs-CZ',\r\n  czech: 'cs-CZ',\r\n  ro: 'ro-RO',\r\n  romanian: 'ro-RO',\r\n  da: 'da-DK',\r\n  danish: 'da-DK',\r\n  hu: 'hu-HU',\r\n  hungarian: 'hu-HU',\r\n  no: 'nb-NO',\r\n  norwegian: 'nb-NO',\r\n  th: 'th-TH',\r\n  thai: 'th-TH',\r\n};\r\n\r\n/**\r\n * Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n */\r\nfunction toBCP47(language: string): string {\r\n  if (language.includes('-')) return language;\r\n  return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/* ─── SpeechStreamingManager ────────────────────────────── */\r\n\r\n/**\r\n * Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n */\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n\r\nexport class SpeechStreamingManager {\r\n  private recognition: SpeechRecognitionInstance | null = null;\r\n  private accumulated = '';\r\n  private active = false;\r\n  private receivedResult = false;\r\n  private noResultTimer: ReturnType<typeof setTimeout> | null = null;\r\n  private onTranscript: ((text: string) => void) | null = null;\r\n  private onPause: (() => void) | null = null;\r\n  private onError: ((message: string) => void) | null = null;\r\n  private onDebug: ((message: string) => void) | null = null;\r\n\r\n  /** Check if the Web Speech API is available in this environment. */\r\n  static isSupported(): boolean {\r\n    return getSpeechRecognition() !== null;\r\n  }\r\n\r\n  /** Set callback for streaming transcript updates (interim + final text). */\r\n  setOnTranscript(fn: (text: string) => void): void {\r\n    this.onTranscript = fn;\r\n  }\r\n\r\n  /** Set callback for speech pause detection (Speech API onend). */\r\n  setOnPause(fn: () => void): void {\r\n    this.onPause = fn;\r\n  }\r\n\r\n  /** Set callback for errors. */\r\n  setOnError(fn: (message: string) => void): void {\r\n    this.onError = fn;\r\n  }\r\n\r\n  /** Set callback for diagnostic debug messages. */\r\n  setOnDebug(fn: (message: string) => void): void {\r\n    this.onDebug = fn;\r\n  }\r\n\r\n  private log(message: string): void {\r\n    this.onDebug?.(message);\r\n    console.warn(message);\r\n  }\r\n\r\n  /**\r\n   * Start streaming recognition. Returns a Promise that resolves once\r\n   * SpeechRecognition has claimed the microphone (onaudiostart) or after\r\n   * a 300ms fallback — whichever comes first. The engine should await\r\n   * this before calling getUserMedia to avoid dual-mic conflicts.\r\n   *\r\n   * When skipMicWait is true (warm restart — mic already active), returns\r\n   * immediately after calling recognition.start() without waiting for\r\n   * onaudiostart.\r\n   */\r\n  start(language: string, skipMicWait = false): Promise<void> {\r\n    const SR = getSpeechRecognition();\r\n    if (!SR) {\r\n      this.log('[SSM] SpeechRecognition not available in this environment');\r\n      return Promise.resolve();\r\n    }\r\n\r\n    const bcp47 = toBCP47(language);\r\n    this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n    this.accumulated = '';\r\n    this.active = true;\r\n    this.receivedResult = false;\r\n\r\n    const recognition = new SR();\r\n    recognition.continuous = true;\r\n    recognition.interimResults = true;\r\n    recognition.lang = bcp47;\r\n\r\n    let lastFinalIndex = -1;\r\n    let lastFinalText = '';\r\n\r\n    // Promise resolves when SR claims mic (onaudiostart) or after fallback.\r\n    // This ensures getUserMedia doesn't compete for the mic.\r\n    let micReady = false;\r\n    const micClaimPromise = new Promise<void>((resolve) => {\r\n      recognition.onaudiostart = () => {\r\n        if (this.recognition !== recognition) return;\r\n        this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n        if (!micReady) {\r\n          micReady = true;\r\n          resolve();\r\n        }\r\n      };\r\n      // Fallback: resolve after 300ms even if onaudiostart never fires\r\n      setTimeout(() => {\r\n        if (!micReady) {\r\n          micReady = true;\r\n          this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n          resolve();\r\n        }\r\n      }, 300);\r\n    });\r\n\r\n    // Detect silent failure: if no onresult fires within timeout, emit error\r\n    this.clearNoResultTimer();\r\n    this.noResultTimer = setTimeout(() => {\r\n      if (this.active && !this.receivedResult) {\r\n        this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n        this.onError?.(\r\n          'Speech streaming started but received no results. ' +\r\n            'Mic may be blocked by another audio capture.',\r\n        );\r\n      }\r\n    }, NO_RESULT_TIMEOUT_MS);\r\n\r\n    recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.receivedResult = true;\r\n      this.clearNoResultTimer();\r\n\r\n      let final_ = '';\r\n      let interim = '';\r\n      for (let i = e.resultIndex; i < e.results.length; i++) {\r\n        const t = e.results[i][0].transcript;\r\n        if (e.results[i].isFinal) {\r\n          if (i > lastFinalIndex) {\r\n            final_ += t;\r\n            lastFinalIndex = i;\r\n          }\r\n        } else {\r\n          interim += t;\r\n        }\r\n      }\r\n\r\n      this.log(\r\n        `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n      );\r\n\r\n      if (final_ && final_.trim() !== lastFinalText) {\r\n        lastFinalText = final_.trim();\r\n        this.accumulated = this.accumulated\r\n          ? this.accumulated + ' ' + final_.trim()\r\n          : final_.trim();\r\n        this.onTranscript?.(this.accumulated);\r\n      } else if (interim) {\r\n        const trimmed = interim.trimStart();\r\n        const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n        this.onTranscript?.(full);\r\n      }\r\n    };\r\n\r\n    recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onerror — ${e.error}`);\r\n      this.onError?.(e.error);\r\n    };\r\n\r\n    recognition.onend = () => {\r\n      if (this.recognition !== recognition) return;\r\n      this.log(`[SSM] onend — active: ${this.active}, receivedResult: ${this.receivedResult}`);\r\n\r\n      if (this.active) {\r\n        // Speech API paused — trigger correction\r\n        this.onPause?.();\r\n        // Restart for continued streaming\r\n        try {\r\n          recognition.start();\r\n          this.log('[SSM] restarted after pause');\r\n        } catch (err) {\r\n          this.log(`[SSM] restart THREW: ${err}`);\r\n          this.recognition = null;\r\n          this.onError?.('Speech recognition failed to restart after pause.');\r\n        }\r\n      } else {\r\n        this.recognition = null;\r\n      }\r\n    };\r\n\r\n    this.recognition = recognition;\r\n    try {\r\n      recognition.start();\r\n      this.log('[SSM] recognition.start() succeeded');\r\n    } catch (err) {\r\n      this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n      this.recognition = null;\r\n      this.active = false;\r\n      this.clearNoResultTimer();\r\n      this.onError?.(\r\n        `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n      );\r\n      return Promise.resolve(); // Resolve so engine can proceed to getUserMedia\r\n    }\r\n\r\n    if (skipMicWait) {\r\n      this.log('[SSM] skipMicWait — warm restart, returning immediately');\r\n      return Promise.resolve();\r\n    }\r\n\r\n    return micClaimPromise;\r\n  }\r\n\r\n  private clearNoResultTimer(): void {\r\n    if (this.noResultTimer) {\r\n      clearTimeout(this.noResultTimer);\r\n      this.noResultTimer = null;\r\n    }\r\n  }\r\n\r\n  /** Stop streaming recognition and return accumulated text. */\r\n  stop(): string {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.stop();\r\n    }\r\n    const result = this.accumulated;\r\n    this.accumulated = '';\r\n    return result;\r\n  }\r\n\r\n  /** Abort immediately without returning text. */\r\n  destroy(): void {\r\n    this.active = false;\r\n    this.clearNoResultTimer();\r\n    if (this.recognition) {\r\n      const rec = this.recognition;\r\n      this.recognition = null;\r\n      rec.abort();\r\n    }\r\n    this.accumulated = '';\r\n    this.onTranscript = null;\r\n    this.onPause = null;\r\n    this.onError = null;\r\n    this.onDebug = null;\r\n  }\r\n}\r\n","import type {\r\n  STTConfig,\r\n  STTState,\r\n  STTEvents,\r\n  STTStatus,\r\n  ResolvedSTTConfig,\r\n  AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, pauseCapture, resumeCapture, snapshotAudio, resampleAudio } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/**\r\n * Main STT engine — the public API for speech-to-text with Whisper correction.\r\n *\r\n * Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n */\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n  private config: ResolvedSTTConfig;\r\n  private workerManager: WorkerManager;\r\n  private correctionOrchestrator: CorrectionOrchestrator;\r\n  private speechStreaming: SpeechStreamingManager;\r\n  private capture: AudioCaptureHandle | null = null;\r\n  private state: STTState;\r\n  private workerUrl?: URL;\r\n  /** Prevents performCorrection from emitting while stop() is consuming the in-flight result. */\r\n  private _stopping = false;\r\n\r\n  /**\r\n   * Create a new STT engine instance.\r\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n   */\r\n  constructor(config?: STTConfig, workerUrl?: URL) {\r\n    super();\r\n    this.config = resolveConfig(config);\r\n    this.workerManager = new WorkerManager();\r\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n    this.speechStreaming = new SpeechStreamingManager();\r\n    this.workerUrl = workerUrl;\r\n\r\n    this.state = {\r\n      status: 'idle',\r\n      isModelLoaded: false,\r\n      loadProgress: 0,\r\n      backend: null,\r\n      error: null,\r\n    };\r\n\r\n    this.correctionOrchestrator.setCorrectionFn(() => {\r\n      this.performCorrection();\r\n    });\r\n\r\n    this.setupWorkerListeners();\r\n    this.setupStreamingCallbacks();\r\n  }\r\n\r\n  /** Initialize the engine: spawn worker and load model. */\r\n  async init(): Promise<void> {\r\n    this.updateStatus('loading');\r\n    this.workerManager.spawn(this.workerUrl);\r\n\r\n    try {\r\n      await this.workerManager.loadModel(this.config);\r\n      this.state.isModelLoaded = true;\r\n      this.updateStatus('ready');\r\n    } catch (err) {\r\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n      this.updateStatus('idle');\r\n      throw err;\r\n    }\r\n  }\r\n\r\n  /** Start recording audio and enable correction cycles. */\r\n  async start(): Promise<void> {\r\n    if (this.state.status !== 'ready') {\r\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n    }\r\n\r\n    try {\r\n      // Check if mic is already warm from a previous recording session.\r\n      // When warm, skip getUserMedia and the 300ms SR mic-claim wait.\r\n      const warmCapture =\r\n        this.capture &&\r\n        this.capture.stream.getTracks().every((t) => t.readyState === 'live');\r\n\r\n      this.emitDebug(\r\n        `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\", warm: ${!!warmCapture}`,\r\n      );\r\n\r\n      if (this.config.streaming.enabled) {\r\n        // On warm restart, skip the mic-claim wait — no getUserMedia race.\r\n        await this.speechStreaming.start(this.config.language, !!warmCapture);\r\n        if (!warmCapture) {\r\n          this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n        }\r\n      }\r\n\r\n      if (warmCapture) {\r\n        await resumeCapture(this.capture!);\r\n        this.emitDebug('[STT] warm mic resumed — skipped getUserMedia');\r\n      } else {\r\n        // First start or stale capture: full mic init with getUserMedia.\r\n        this.capture = await startCapture();\r\n      }\r\n\r\n      this.updateStatus('recording');\r\n      this.correctionOrchestrator.start();\r\n    } catch (err) {\r\n      this.emitError(\r\n        'MIC_DENIED',\r\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n      );\r\n    }\r\n  }\r\n\r\n  /** Stop recording, run final transcription, return text.\r\n   *  Mic and AudioContext stay alive for fast restart — call destroy() to fully release. */\r\n  async stop(): Promise<string> {\r\n    if (!this.capture) return '';\r\n\r\n    // Prevent any in-flight performCorrection from emitting — stop() will own the final emit.\r\n    this._stopping = true;\r\n    this.correctionOrchestrator.stop();\r\n    this.speechStreaming.stop();\r\n\r\n    this.updateStatus('processing');\r\n\r\n    // If a correction job is already running, reuse it: Whisper can't be interrupted mid-inference\r\n    // anyway, so cancelling just wastes the work and causes a second full run.\r\n    if (this.workerManager.isTranscribing) {\r\n      try {\r\n        // Resample audio and await the in-flight result in parallel.\r\n        const [audio, inFlightText] = await Promise.all([\r\n          pauseCapture(this.capture),\r\n          this.workerManager.awaitCurrentTranscription(),\r\n        ]);\r\n        this._stopping = false;\r\n\r\n        const text = inFlightText.trim();\r\n        if (text) {\r\n          this.emit('correction', text);\r\n          this.updateStatus('ready');\r\n          return text;\r\n        }\r\n\r\n        // In-flight returned empty (cancelled or error) — fall through to fresh transcription.\r\n        if (audio.length > 0) {\r\n          const freshText = await this.workerManager.transcribe(audio);\r\n          this.emit('correction', freshText);\r\n          this.updateStatus('ready');\r\n          return freshText;\r\n        }\r\n\r\n        this.updateStatus('ready');\r\n        return '';\r\n      } catch (err) {\r\n        this._stopping = false;\r\n        this.emitError(\r\n          'TRANSCRIPTION_FAILED',\r\n          err instanceof Error ? err.message : 'Final transcription failed.',\r\n        );\r\n        this.updateStatus('ready');\r\n        return '';\r\n      }\r\n    }\r\n\r\n    // No job in-flight — cancel is a no-op, run fresh transcription.\r\n    this.workerManager.cancel();\r\n    this._stopping = false;\r\n\r\n    try {\r\n      // Soft pause — keeps stream and AudioContext alive for fast restart.\r\n      const audio = await pauseCapture(this.capture);\r\n\r\n      if (audio.length === 0) {\r\n        this.updateStatus('ready');\r\n        return '';\r\n      }\r\n\r\n      const text = await this.workerManager.transcribe(audio);\r\n      this.emit('correction', text);\r\n      this.updateStatus('ready');\r\n      return text;\r\n    } catch (err) {\r\n      this.emitError(\r\n        'TRANSCRIPTION_FAILED',\r\n        err instanceof Error ? err.message : 'Final transcription failed.',\r\n      );\r\n      this.updateStatus('ready');\r\n      return '';\r\n    }\r\n  }\r\n\r\n  /** Destroy the engine: terminate worker, release all resources. */\r\n  destroy(): void {\r\n    this.correctionOrchestrator.stop();\r\n    this.speechStreaming.destroy();\r\n\r\n    if (this.capture) {\r\n      try {\r\n        this.capture._processor.disconnect();\r\n      } catch {\r\n        /* already disconnected */\r\n      }\r\n      for (const track of this.capture.stream.getTracks()) {\r\n        track.stop();\r\n      }\r\n      this.capture.audioCtx.close().catch(() => {});\r\n      this.capture = null;\r\n    }\r\n\r\n    this.workerManager.destroy();\r\n    this.updateStatus('idle');\r\n    this.removeAllListeners();\r\n  }\r\n\r\n  /** Get current engine state. */\r\n  getState(): Readonly<STTState> {\r\n    return { ...this.state };\r\n  }\r\n\r\n  /** Notify the correction orchestrator of a speech pause. */\r\n  notifyPause(): void {\r\n    this.correctionOrchestrator.onPauseDetected();\r\n  }\r\n\r\n  private async performCorrection(): Promise<void> {\r\n    if (!this.capture || !this.state.isModelLoaded) return;\r\n\r\n    this.workerManager.cancel();\r\n\r\n    try {\r\n      const samples = snapshotAudio(this.capture);\r\n      const nativeSr = this.capture.audioCtx.sampleRate;\r\n      const audio = await resampleAudio(samples, nativeSr);\r\n\r\n      if (audio.length === 0) return;\r\n\r\n      const text = await this.workerManager.transcribe(audio);\r\n      if (text.trim() && this.capture && !this._stopping) {\r\n        this.emit('correction', text);\r\n      }\r\n    } catch (err) {\r\n      this.emitError(\r\n        'TRANSCRIPTION_FAILED',\r\n        err instanceof Error ? err.message : 'Correction transcription failed.',\r\n      );\r\n      // Recording continues — error is non-fatal\r\n    }\r\n  }\r\n\r\n  private setupStreamingCallbacks(): void {\r\n    this.speechStreaming.setOnDebug((message) => {\r\n      this.emit('debug', message);\r\n    });\r\n\r\n    this.speechStreaming.setOnTranscript((text) => {\r\n      this.emitDebug(`[STT] transcript callback — \"${text}\"`);\r\n      this.emit('transcript', text);\r\n    });\r\n\r\n    this.speechStreaming.setOnPause(() => {\r\n      this.emitDebug('[STT] pause callback — triggering correction');\r\n      this.correctionOrchestrator.onPauseDetected();\r\n    });\r\n\r\n    this.speechStreaming.setOnError((message) => {\r\n      this.emitDebug(`[STT] streaming error — \"${message}\"`);\r\n      this.emitError('STREAMING_ERROR', message);\r\n    });\r\n  }\r\n\r\n  private setupWorkerListeners(): void {\r\n    this.workerManager.on('progress', (percent) => {\r\n      this.state.loadProgress = percent;\r\n      this.emit('status', { ...this.state });\r\n    });\r\n\r\n    this.workerManager.on('error', (message) => {\r\n      this.emitError('WORKER_ERROR', message);\r\n    });\r\n  }\r\n\r\n  private updateStatus(status: STTStatus): void {\r\n    this.state.status = status;\r\n    this.state.error = null;\r\n    this.emit('status', { ...this.state });\r\n  }\r\n\r\n  private emitError(code: string, message: string): void {\r\n    this.state.error = message;\r\n    this.emit('error', { code, message });\r\n  }\r\n\r\n  private emitDebug(message: string): void {\r\n    console.warn(message);\r\n    this.emit('debug', message);\r\n  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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@tekyzinc/stt-component",
-  "version": "0.2.5",
+  "version": "0.3.0",
   "description": "Framework-agnostic speech-to-text with real-time streaming transcription and mid-recording Whisper correction",
   "type": "module",
   "main": "dist/index.cjs",