@tekyzinc/stt-component 0.1.0 → 0.2.1

package/README.md

@@ -1,6 +1,6 @@
 # STT-Component
 
-![Version](https://img.shields.io/badge/version-0.1.0-blue)
+![Version](https://img.shields.io/badge/version-0.2.1-blue)
 
 A framework-agnostic, browser-first speech-to-text package with real-time streaming transcription and mid-recording Whisper correction, powered by [@huggingface/transformers](https://github.com/huggingface/transformers.js).
 
@@ -83,8 +83,8 @@ Unsubscribe a specific listener.
 
 | Event | Callback Signature | Description |
 |-------|-------------------|-------------|
-| `transcript` | `(text: string) => void` | Streaming interim text during recording |
-| `correction` | `(text: string) => void` | Whisper-corrected text replacing interim text |
+| `transcript` | `(text: string) => void` | Real-time streaming text via Web Speech API (display in *italics*) |
+| `correction` | `(text: string) => void` | Whisper-corrected text replacing interim text (display in normal style) |
 | `error` | `(error: STTError) => void` | Actionable error (`{ code: string, message: string }`) |
 | `status` | `(state: STTState) => void` | Engine state changes |
 
@@ -96,6 +96,7 @@ Unsubscribe a specific listener.
 | `MODEL_LOAD_FAILED` | Whisper model download or initialization failed |
 | `TRANSCRIPTION_FAILED` | Whisper inference failed (recording continues) |
 | `WORKER_ERROR` | Web Worker encountered an error |
+| `STREAMING_ERROR` | Web Speech API streaming error |
 
 #### Engine States (`STTStatus`)
 
@@ -120,8 +121,11 @@ All fields are optional. Defaults shown in the table.
 | `language` | `string` | `'en'` | Transcription language |
 | `dtype` | `string` | `'q4'` | Model quantization dtype |
 | `correction.enabled` | `boolean` | `true` | Enable mid-recording Whisper correction |
+| `correction.provider` | `'whisper'` | `'whisper'` | Correction engine provider |
 | `correction.pauseThreshold` | `number` (ms) | `3000` | Silence duration before triggering correction |
 | `correction.forcedInterval` | `number` (ms) | `5000` | Maximum interval between forced corrections |
+| `streaming.enabled` | `boolean` | `true` | Enable real-time streaming transcript via Web Speech API |
+| `streaming.provider` | `'web-speech-api'` | `'web-speech-api'` | Streaming provider (Chrome/Edge) |
 | `chunking.chunkLengthS` | `number` (seconds) | `30` | Chunk length for Whisper processing |
 | `chunking.strideLengthS` | `number` (seconds) | `5` | Stride length for overlapping chunks |
 
@@ -262,6 +266,9 @@ import type {
   STTModelSize,
   STTBackend,
   STTStatus,
+  STTCorrectionProvider,
+  STTStreamingProvider,
+  STTStreamingConfig,
 } from '@tekyzinc/stt-component';
 
 // Utilities (advanced usage)
@@ -271,6 +278,7 @@ import {
   TypedEventEmitter,
   WorkerManager,
   CorrectionOrchestrator,
+  SpeechStreamingManager,
 } from '@tekyzinc/stt-component';
 ```
 

package/dist/index.cjs

@@ -23,6 +23,7 @@ __export(index_exports, {
   CorrectionOrchestrator: () => CorrectionOrchestrator,
   DEFAULT_STT_CONFIG: () => DEFAULT_STT_CONFIG,
   STTEngine: () => STTEngine,
+  SpeechStreamingManager: () => SpeechStreamingManager,
   TypedEventEmitter: () => TypedEventEmitter,
   WorkerManager: () => WorkerManager,
   resampleAudio: () => resampleAudio,
@@ -41,12 +42,17 @@ var DEFAULT_STT_CONFIG = {
   dtype: "q4",
   correction: {
     enabled: true,
+    provider: "whisper",
     pauseThreshold: 3e3,
     forcedInterval: 5e3
   },
   chunking: {
     chunkLengthS: 30,
     strideLengthS: 5
+  },
+  streaming: {
+    enabled: true,
+    provider: "web-speech-api"
   }
 };
 function resolveConfig(config) {
@@ -57,12 +63,17 @@ function resolveConfig(config) {
     dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,
     correction: {
       enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,
+      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,
       pauseThreshold: config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,
       forcedInterval: config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval
     },
     chunking: {
       chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,
       strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS
+    },
+    streaming: {
+      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,
+      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider
     }
   };
 }
@@ -318,11 +329,152 @@ var CorrectionOrchestrator = class {
   }
 };
 
+// src/speech-streaming.ts
+function getSpeechRecognition() {
+  if (typeof globalThis === "undefined") return null;
+  const w = globalThis;
+  return w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null;
+}
+var NO_RESULT_TIMEOUT_MS = 5e3;
+var SpeechStreamingManager = class {
+  recognition = null;
+  accumulated = "";
+  active = false;
+  receivedResult = false;
+  noResultTimer = null;
+  onTranscript = null;
+  onPause = null;
+  onError = null;
+  /** Check if the Web Speech API is available in this environment. */
+  static isSupported() {
+    return getSpeechRecognition() !== null;
+  }
+  /** Set callback for streaming transcript updates (interim + final text). */
+  setOnTranscript(fn) {
+    this.onTranscript = fn;
+  }
+  /** Set callback for speech pause detection (Speech API onend). */
+  setOnPause(fn) {
+    this.onPause = fn;
+  }
+  /** Set callback for errors. */
+  setOnError(fn) {
+    this.onError = fn;
+  }
+  /** Start streaming recognition. No-op if Speech API unavailable. */
+  start(language) {
+    const SR = getSpeechRecognition();
+    if (!SR) return;
+    this.accumulated = "";
+    this.active = true;
+    this.receivedResult = false;
+    const recognition = new SR();
+    recognition.continuous = true;
+    recognition.interimResults = true;
+    recognition.lang = language;
+    let lastFinalIndex = -1;
+    let lastFinalText = "";
+    this.clearNoResultTimer();
+    this.noResultTimer = setTimeout(() => {
+      if (this.active && !this.receivedResult) {
+        this.onError?.(
+          "Speech streaming started but received no results. Mic may be blocked by another audio capture."
+        );
+      }
+    }, NO_RESULT_TIMEOUT_MS);
+    recognition.onresult = (e) => {
+      if (this.recognition !== recognition) return;
+      this.receivedResult = true;
+      this.clearNoResultTimer();
+      let final_ = "";
+      let interim = "";
+      for (let i = e.resultIndex; i < e.results.length; i++) {
+        const t = e.results[i][0].transcript;
+        if (e.results[i].isFinal) {
+          if (i > lastFinalIndex) {
+            final_ += t;
+            lastFinalIndex = i;
+          }
+        } else {
+          interim += t;
+        }
+      }
+      if (final_ && final_.trim() !== lastFinalText) {
+        lastFinalText = final_.trim();
+        this.accumulated = this.accumulated ? this.accumulated + " " + final_.trim() : final_.trim();
+        this.onTranscript?.(this.accumulated);
+      } else if (interim) {
+        const trimmed = interim.trimStart();
+        const full = this.accumulated ? this.accumulated + " " + trimmed : trimmed;
+        this.onTranscript?.(full);
+      }
+    };
+    recognition.onerror = (e) => {
+      if (this.recognition !== recognition) return;
+      this.onError?.(e.error);
+    };
+    recognition.onend = () => {
+      if (this.recognition !== recognition) return;
+      if (this.active) {
+        this.onPause?.();
+        try {
+          recognition.start();
+        } catch {
+          this.recognition = null;
+        }
+      } else {
+        this.recognition = null;
+      }
+    };
+    this.recognition = recognition;
+    try {
+      recognition.start();
+    } catch {
+      this.recognition = null;
+      this.active = false;
+    }
+  }
+  clearNoResultTimer() {
+    if (this.noResultTimer) {
+      clearTimeout(this.noResultTimer);
+      this.noResultTimer = null;
+    }
+  }
+  /** Stop streaming recognition and return accumulated text. */
+  stop() {
+    this.active = false;
+    this.clearNoResultTimer();
+    if (this.recognition) {
+      const rec = this.recognition;
+      this.recognition = null;
+      rec.stop();
+    }
+    const result = this.accumulated;
+    this.accumulated = "";
+    return result;
+  }
+  /** Abort immediately without returning text. */
+  destroy() {
+    this.active = false;
+    this.clearNoResultTimer();
+    if (this.recognition) {
+      const rec = this.recognition;
+      this.recognition = null;
+      rec.abort();
+    }
+    this.accumulated = "";
+    this.onTranscript = null;
+    this.onPause = null;
+    this.onError = null;
+  }
+};
+
 // src/stt-engine.ts
 var STTEngine = class extends TypedEventEmitter {
   config;
   workerManager;
   correctionOrchestrator;
+  speechStreaming;
   capture = null;
   state;
   workerUrl;
@@ -336,6 +488,7 @@ var STTEngine = class extends TypedEventEmitter {
     this.config = resolveConfig(config);
     this.workerManager = new WorkerManager();
     this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);
+    this.speechStreaming = new SpeechStreamingManager();
     this.workerUrl = workerUrl;
     this.state = {
       status: "idle",
@@ -348,6 +501,7 @@ var STTEngine = class extends TypedEventEmitter {
       this.performCorrection();
     });
     this.setupWorkerListeners();
+    this.setupStreamingCallbacks();
   }
   /** Initialize the engine: spawn worker and load model. */
   async init() {
@@ -369,6 +523,9 @@ var STTEngine = class extends TypedEventEmitter {
       throw new Error(`Cannot start: engine is "${this.state.status}", expected "ready"`);
     }
     try {
+      if (this.config.streaming.enabled) {
+        this.speechStreaming.start(this.config.language);
+      }
       this.capture = await startCapture();
       this.updateStatus("recording");
       this.correctionOrchestrator.start();
@@ -383,6 +540,7 @@ var STTEngine = class extends TypedEventEmitter {
   async stop() {
     if (!this.capture) return "";
     this.correctionOrchestrator.stop();
+    this.speechStreaming.stop();
     this.workerManager.cancel();
     this.updateStatus("processing");
     try {
@@ -408,6 +566,7 @@ var STTEngine = class extends TypedEventEmitter {
   /** Destroy the engine: terminate worker, release all resources. */
   destroy() {
     this.correctionOrchestrator.stop();
+    this.speechStreaming.destroy();
     if (this.capture) {
       for (const track of this.capture.stream.getTracks()) {
         track.stop();
@@ -447,6 +606,17 @@ var STTEngine = class extends TypedEventEmitter {
       );
     }
   }
+  setupStreamingCallbacks() {
+    this.speechStreaming.setOnTranscript((text) => {
+      this.emit("transcript", text);
+    });
+    this.speechStreaming.setOnPause(() => {
+      this.correctionOrchestrator.onPauseDetected();
+    });
+    this.speechStreaming.setOnError((message) => {
+      this.emitError("STREAMING_ERROR", message);
+    });
+  }
   setupWorkerListeners() {
     this.workerManager.on("progress", (percent) => {
       this.state.loadProgress = percent;
@@ -471,6 +641,7 @@ var STTEngine = class extends TypedEventEmitter {
   CorrectionOrchestrator,
   DEFAULT_STT_CONFIG,
   STTEngine,
+  SpeechStreamingManager,
   TypedEventEmitter,
   WorkerManager,
   resampleAudio,

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/stt-engine.ts"],"sourcesContent":["// Public types\nexport type {\n  STTModelSize,\n  STTBackend,\n  STTStatus,\n  STTCorrectionConfig,\n  STTChunkingConfig,\n  STTConfig,\n  ResolvedSTTConfig,\n  STTState,\n  STTError,\n  STTEvents,\n  AudioCaptureHandle,\n} from './types.js';\n\n// Public values\nexport { DEFAULT_STT_CONFIG, resolveConfig } from './types.js';\n\n// Event emitter\nexport { TypedEventEmitter } from './event-emitter.js';\n\n// Audio capture\nexport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\n\n// Worker manager\nexport { WorkerManager } from './worker-manager.js';\nexport type { WorkerManagerEvents } from './worker-manager.js';\n\n// Correction orchestrator\nexport { CorrectionOrchestrator } from './correction-orchestrator.js';\n\n// STT Engine (main public API)\nexport { STTEngine } from './stt-engine.js';\n","/** Supported Whisper model sizes. */\nexport type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';\n\n/** Supported compute backends. */\nexport type STTBackend = 'webgpu' | 'wasm' | 'auto';\n\n/** Engine lifecycle states. */\nexport type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';\n\n/** Correction timing configuration. */\nexport interface STTCorrectionConfig {\n  /** Enable mid-recording Whisper correction. Default: true */\n  enabled?: boolean;\n  /** Silence duration (ms) before triggering correction. Default: 3000 */\n  pauseThreshold?: number;\n  /** Maximum interval (ms) between forced corrections. Default: 5000 */\n  forcedInterval?: number;\n}\n\n/** Audio chunking configuration for long-form audio. */\nexport interface STTChunkingConfig {\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\n  chunkLengthS?: number;\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\n  strideLengthS?: number;\n}\n\n/** Full engine configuration. All fields optional — sensible defaults applied. */\nexport interface STTConfig {\n  /** Whisper model size. Default: 'tiny' */\n  model?: STTModelSize;\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\n  backend?: STTBackend;\n  /** Transcription language. Default: 'en' */\n  language?: string;\n  /** Model quantization dtype. Default: 'q4' */\n  dtype?: string;\n  /** Mid-recording correction settings. */\n  correction?: STTCorrectionConfig;\n  /** Audio chunking settings for long-form audio. */\n  chunking?: STTChunkingConfig;\n}\n\n/** Resolved configuration with all defaults applied. */\nexport interface ResolvedSTTConfig {\n  model: STTModelSize;\n  backend: STTBackend;\n  language: string;\n  dtype: string;\n  correction: Required<STTCorrectionConfig>;\n  chunking: Required<STTChunkingConfig>;\n}\n\n/** Engine state exposed to consumers via status events. */\nexport interface STTState {\n  status: STTStatus;\n  isModelLoaded: boolean;\n  /** Model download progress (0–100). */\n  loadProgress: number;\n  /** Active compute backend, or null if not yet determined. */\n  backend: 'webgpu' | 'wasm' | null;\n  error: string | null;\n}\n\n/** Structured error emitted via the 'error' event. */\nexport interface STTError {\n  code: string;\n  message: string;\n}\n\n/** Event map for the typed event emitter. */\nexport type STTEvents = {\n  /** Streaming interim text during recording. */\n  transcript: (text: string) => void;\n  /** Whisper-corrected text replacing interim text. */\n  correction: (text: string) => void;\n  /** Actionable error (mic denied, model fail, transcription fail). */\n  error: (error: STTError) => void;\n  /** Engine state change. */\n  status: (state: STTState) => void;\n};\n\n/** Handle returned by audio capture — used internally. */\nexport interface AudioCaptureHandle {\n  audioCtx: AudioContext;\n  stream: MediaStream;\n  samples: Float32Array[];\n  /** Retain reference to prevent GC from stopping audio processing. */\n  _processor: ScriptProcessorNode;\n}\n\n/** Message sent from main thread to Whisper worker. */\nexport interface WorkerMessage {\n  type: 'load' | 'transcribe' | 'cancel';\n  audio?: Float32Array;\n  config?: {\n    model: string;\n    backend: STTBackend;\n    language: string;\n    dtype: string;\n    chunkLengthS: number;\n    strideLengthS: number;\n  };\n}\n\n/** Response sent from Whisper worker to main thread. */\nexport interface WorkerResponse {\n  type: 'progress' | 'ready' | 'result' | 'error';\n  data?: unknown;\n}\n\n/** Default configuration values. */\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\n  model: 'tiny',\n  backend: 'auto',\n  language: 'en',\n  dtype: 'q4',\n  correction: {\n    enabled: true,\n    pauseThreshold: 3_000,\n    forcedInterval: 5_000,\n  },\n  chunking: {\n    chunkLengthS: 30,\n    strideLengthS: 5,\n  },\n};\n\n/** Merge user config with defaults to produce resolved config. */\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\n  return {\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\n    correction: {\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\n      pauseThreshold:\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\n      forcedInterval:\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\n    },\n    chunking: {\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\n    },\n  };\n}\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';\n\nconst TARGET_SAMPLE_RATE = 16_000;\n\n/**\n * Start capturing raw PCM audio from the microphone.\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\n */\nexport async function startCapture(): Promise<AudioCaptureHandle> {\n  const stream = await navigator.mediaDevices.getUserMedia({\n    audio: { channelCount: 1 },\n  });\n  const audioCtx = new AudioContext();\n\n  // Chrome may suspend AudioContext — must resume within user gesture\n  if (audioCtx.state === 'suspended') {\n    await audioCtx.resume();\n  }\n\n  const source = audioCtx.createMediaStreamSource(stream);\n  const samples: Float32Array[] = [];\n\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\n  };\n\n  // Connect through a silent gain node so mic audio doesn't play back\n  const silencer = audioCtx.createGain();\n  silencer.gain.value = 0;\n  source.connect(processor);\n  processor.connect(silencer);\n  silencer.connect(audioCtx.destination);\n\n  return { audioCtx, stream, samples, _processor: processor };\n}\n\n/**\n * Copy current audio buffer without stopping capture.\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\n */\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\n  return [...capture.samples];\n}\n\n/**\n * Concatenate sample chunks and resample to 16kHz for Whisper.\n */\nexport async function resampleAudio(\n  samples: Float32Array[],\n  nativeSr: number,\n): Promise<Float32Array> {\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\n  if (totalLength === 0) return new Float32Array(0);\n\n  const fullAudio = new Float32Array(totalLength);\n  let offset = 0;\n  for (const s of samples) {\n    fullAudio.set(s, offset);\n    offset += s.length;\n  }\n\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\n\n  const duration = fullAudio.length / nativeSr;\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\n  buffer.getChannelData(0).set(fullAudio);\n  const src = offline.createBufferSource();\n  src.buffer = buffer;\n  src.connect(offline.destination);\n  src.start(0);\n  const resampled = await offline.startRendering();\n  return resampled.getChannelData(0);\n}\n\n/**\n * Stop capturing and return resampled audio at 16kHz.\n */\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\n  const { audioCtx, stream, samples, _processor } = capture;\n\n  // Disconnect processor to stop capturing\n  try {\n    _processor.disconnect();\n  } catch {\n    /* already disconnected */\n  }\n\n  // Stop microphone tracks\n  for (const track of stream.getTracks()) {\n    track.stop();\n  }\n\n  const nativeSr = audioCtx.sampleRate;\n  await audioCtx.close();\n\n  return resampleAudio(samples, nativeSr);\n}\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","import type {\n  STTConfig,\n  STTState,\n  STTEvents,\n  STTStatus,\n  ResolvedSTTConfig,\n  AudioCaptureHandle,\n} from './types.js';\nimport { resolveConfig } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\nimport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\nimport { WorkerManager } from './worker-manager.js';\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\n\n/**\n * Main STT engine — the public API for speech-to-text with Whisper correction.\n *\n * Usage:\n * ```typescript\n * const engine = new STTEngine({ model: 'tiny' });\n * engine.on('transcript', (text) => console.log(text));\n * engine.on('correction', (text) => console.log('corrected:', text));\n * await engine.init();\n * await engine.start();\n * const finalText = await engine.stop();\n * ```\n */\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\n  private config: ResolvedSTTConfig;\n  private workerManager: WorkerManager;\n  private correctionOrchestrator: CorrectionOrchestrator;\n  private capture: AudioCaptureHandle | null = null;\n  private state: STTState;\n  private workerUrl?: URL;\n\n  /**\n   * Create a new STT engine instance.\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\n   */\n  constructor(config?: STTConfig, workerUrl?: URL) {\n    super();\n    this.config = resolveConfig(config);\n    this.workerManager = new WorkerManager();\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\n    this.workerUrl = workerUrl;\n\n    this.state = {\n      status: 'idle',\n      isModelLoaded: false,\n      loadProgress: 0,\n      backend: null,\n      error: null,\n    };\n\n    this.correctionOrchestrator.setCorrectionFn(() => {\n      this.performCorrection();\n    });\n\n    this.setupWorkerListeners();\n  }\n\n  /** Initialize the engine: spawn worker and load model. */\n  async init(): Promise<void> {\n    this.updateStatus('loading');\n    this.workerManager.spawn(this.workerUrl);\n\n    try {\n      await this.workerManager.loadModel(this.config);\n      this.state.isModelLoaded = true;\n      this.updateStatus('ready');\n    } catch (err) {\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\n      this.updateStatus('idle');\n      throw err;\n    }\n  }\n\n  /** Start recording audio and enable correction cycles. */\n  async start(): Promise<void> {\n    if (this.state.status !== 'ready') {\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\n    }\n\n    try {\n      this.capture = await startCapture();\n      this.updateStatus('recording');\n      this.correctionOrchestrator.start();\n    } catch (err) {\n      this.emitError(\n        'MIC_DENIED',\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\n      );\n    }\n  }\n\n  /** Stop recording, run final transcription, return text. */\n  async stop(): Promise<string> {\n    if (!this.capture) return '';\n\n    this.correctionOrchestrator.stop();\n    this.workerManager.cancel();\n\n    this.updateStatus('processing');\n\n    try {\n      const audio = await stopCapture(this.capture);\n      this.capture = null;\n\n      if (audio.length === 0) {\n        this.updateStatus('ready');\n        return '';\n      }\n\n      const text = await this.workerManager.transcribe(audio);\n      this.emit('correction', text);\n      this.updateStatus('ready');\n      return text;\n    } catch (err) {\n      this.emitError(\n        'TRANSCRIPTION_FAILED',\n        err instanceof Error ? err.message : 'Final transcription failed.',\n      );\n      this.updateStatus('ready');\n      return '';\n    }\n  }\n\n  /** Destroy the engine: terminate worker, release all resources. */\n  destroy(): void {\n    this.correctionOrchestrator.stop();\n\n    if (this.capture) {\n      for (const track of this.capture.stream.getTracks()) {\n        track.stop();\n      }\n      this.capture.audioCtx.close().catch(() => {});\n      this.capture = null;\n    }\n\n    this.workerManager.destroy();\n    this.updateStatus('idle');\n    this.removeAllListeners();\n  }\n\n  /** Get current engine state. */\n  getState(): Readonly<STTState> {\n    return { ...this.state };\n  }\n\n  /** Notify the correction orchestrator of a speech pause. */\n  notifyPause(): void {\n    this.correctionOrchestrator.onPauseDetected();\n  }\n\n  private async performCorrection(): Promise<void> {\n    if (!this.capture || !this.state.isModelLoaded) return;\n\n    this.workerManager.cancel();\n\n    try {\n      const samples = snapshotAudio(this.capture);\n      const nativeSr = this.capture.audioCtx.sampleRate;\n      const audio = await resampleAudio(samples, nativeSr);\n\n      if (audio.length === 0) return;\n\n      const text = await this.workerManager.transcribe(audio);\n      if (text.trim() && this.capture) {\n        this.emit('correction', text);\n      }\n    } catch (err) {\n      this.emitError(\n        'TRANSCRIPTION_FAILED',\n        err instanceof Error ? err.message : 'Correction transcription failed.',\n      );\n      // Recording continues — error is non-fatal\n    }\n  }\n\n  private setupWorkerListeners(): void {\n    this.workerManager.on('progress', (percent) => {\n      this.state.loadProgress = percent;\n      this.emit('status', { ...this.state });\n    });\n\n    this.workerManager.on('error', (message) => {\n      this.emitError('WORKER_ERROR', message);\n    });\n  }\n\n  private updateStatus(status: STTStatus): void {\n    this.state.status = status;\n    this.state.error = null;\n    this.emit('status', { ...this.state });\n  }\n\n  private emitError(code: string, message: string): void {\n    this.state.error = message;\n    this.emit('error', { code, message });\n  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+{"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\nexport type {\n  STTModelSize,\n  STTBackend,\n  STTStatus,\n  STTCorrectionProvider,\n  STTStreamingProvider,\n  STTCorrectionConfig,\n  STTStreamingConfig,\n  STTChunkingConfig,\n  STTConfig,\n  ResolvedSTTConfig,\n  STTState,\n  STTError,\n  STTEvents,\n  AudioCaptureHandle,\n} from './types.js';\n\n// Public values\nexport { DEFAULT_STT_CONFIG, resolveConfig } from './types.js';\n\n// Event emitter\nexport { TypedEventEmitter } from './event-emitter.js';\n\n// Audio capture\nexport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\n\n// Worker manager\nexport { WorkerManager } from './worker-manager.js';\nexport type { WorkerManagerEvents } from './worker-manager.js';\n\n// Correction orchestrator\nexport { CorrectionOrchestrator } from './correction-orchestrator.js';\n\n// Speech streaming\nexport { SpeechStreamingManager } from './speech-streaming.js';\n\n// STT Engine (main public API)\nexport { STTEngine } from './stt-engine.js';\n","/** Supported Whisper model sizes. */\nexport type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';\n\n/** Supported compute backends. */\nexport type STTBackend = 'webgpu' | 'wasm' | 'auto';\n\n/** Engine lifecycle states. */\nexport type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';\n\n/** Supported correction engine providers. */\nexport type STTCorrectionProvider = 'whisper';\n\n/** Supported real-time streaming providers. */\nexport type STTStreamingProvider = 'web-speech-api';\n\n/** Correction engine configuration. */\nexport interface STTCorrectionConfig {\n  /** Enable mid-recording correction. Default: true */\n  enabled?: boolean;\n  /** Correction engine provider. Default: 'whisper' */\n  provider?: STTCorrectionProvider;\n  /** Silence duration (ms) before triggering correction. Default: 3000 */\n  pauseThreshold?: number;\n  /** Maximum interval (ms) between forced corrections. Default: 5000 */\n  forcedInterval?: number;\n}\n\n/** Real-time streaming preview configuration. */\nexport interface STTStreamingConfig {\n  /** Enable real-time streaming transcript. Default: true */\n  enabled?: boolean;\n  /** Streaming provider. Default: 'web-speech-api' */\n  provider?: STTStreamingProvider;\n}\n\n/** Audio chunking configuration for long-form audio. */\nexport interface STTChunkingConfig {\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\n  chunkLengthS?: number;\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\n  strideLengthS?: number;\n}\n\n/** Full engine configuration. All fields optional — sensible defaults applied. */\nexport interface STTConfig {\n  /** Whisper model size. Default: 'tiny' */\n  model?: STTModelSize;\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\n  backend?: STTBackend;\n  /** Transcription language. Default: 'en' */\n  language?: string;\n  /** Model quantization dtype. Default: 'q4' */\n  dtype?: string;\n  /** Mid-recording correction settings. */\n  correction?: STTCorrectionConfig;\n  /** Audio chunking settings for long-form audio. */\n  chunking?: STTChunkingConfig;\n  /** Web Speech API streaming preview settings. */\n  streaming?: STTStreamingConfig;\n}\n\n/** Resolved configuration with all defaults applied. */\nexport interface ResolvedSTTConfig {\n  model: STTModelSize;\n  backend: STTBackend;\n  language: string;\n  dtype: string;\n  correction: Required<STTCorrectionConfig>;\n  chunking: Required<STTChunkingConfig>;\n  streaming: Required<STTStreamingConfig>;\n}\n\n/** Engine state exposed to consumers via status events. */\nexport interface STTState {\n  status: STTStatus;\n  isModelLoaded: boolean;\n  /** Model download progress (0–100). */\n  loadProgress: number;\n  /** Active compute backend, or null if not yet determined. */\n  backend: 'webgpu' | 'wasm' | null;\n  error: string | null;\n}\n\n/** Structured error emitted via the 'error' event. */\nexport interface STTError {\n  code: string;\n  message: string;\n}\n\n/** Event map for the typed event emitter. */\nexport type STTEvents = {\n  /** Streaming interim text during recording. */\n  transcript: (text: string) => void;\n  /** Whisper-corrected text replacing interim text. */\n  correction: (text: string) => void;\n  /** Actionable error (mic denied, model fail, transcription fail). */\n  error: (error: STTError) => void;\n  /** Engine state change. */\n  status: (state: STTState) => void;\n};\n\n/** Handle returned by audio capture — used internally. */\nexport interface AudioCaptureHandle {\n  audioCtx: AudioContext;\n  stream: MediaStream;\n  samples: Float32Array[];\n  /** Retain reference to prevent GC from stopping audio processing. */\n  _processor: ScriptProcessorNode;\n}\n\n/** Message sent from main thread to Whisper worker. */\nexport interface WorkerMessage {\n  type: 'load' | 'transcribe' | 'cancel';\n  audio?: Float32Array;\n  config?: {\n    model: string;\n    backend: STTBackend;\n    language: string;\n    dtype: string;\n    chunkLengthS: number;\n    strideLengthS: number;\n  };\n}\n\n/** Response sent from Whisper worker to main thread. */\nexport interface WorkerResponse {\n  type: 'progress' | 'ready' | 'result' | 'error';\n  data?: unknown;\n}\n\n/** Default configuration values. */\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\n  model: 'tiny',\n  backend: 'auto',\n  language: 'en',\n  dtype: 'q4',\n  correction: {\n    enabled: true,\n    provider: 'whisper',\n    pauseThreshold: 3_000,\n    forcedInterval: 5_000,\n  },\n  chunking: {\n    chunkLengthS: 30,\n    strideLengthS: 5,\n  },\n  streaming: {\n    enabled: true,\n    provider: 'web-speech-api',\n  },\n};\n\n/** Merge user config with defaults to produce resolved config. */\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\n  return {\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\n    correction: {\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\n      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\n      pauseThreshold:\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\n      forcedInterval:\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\n    },\n    chunking: {\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\n    },\n    streaming: {\n      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\n      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\n    },\n  };\n}\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';\n\nconst TARGET_SAMPLE_RATE = 16_000;\n\n/**\n * Start capturing raw PCM audio from the microphone.\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\n */\nexport async function startCapture(): Promise<AudioCaptureHandle> {\n  const stream = await navigator.mediaDevices.getUserMedia({\n    audio: { channelCount: 1 },\n  });\n  const audioCtx = new AudioContext();\n\n  // Chrome may suspend AudioContext — must resume within user gesture\n  if (audioCtx.state === 'suspended') {\n    await audioCtx.resume();\n  }\n\n  const source = audioCtx.createMediaStreamSource(stream);\n  const samples: Float32Array[] = [];\n\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\n  };\n\n  // Connect through a silent gain node so mic audio doesn't play back\n  const silencer = audioCtx.createGain();\n  silencer.gain.value = 0;\n  source.connect(processor);\n  processor.connect(silencer);\n  silencer.connect(audioCtx.destination);\n\n  return { audioCtx, stream, samples, _processor: processor };\n}\n\n/**\n * Copy current audio buffer without stopping capture.\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\n */\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\n  return [...capture.samples];\n}\n\n/**\n * Concatenate sample chunks and resample to 16kHz for Whisper.\n */\nexport async function resampleAudio(\n  samples: Float32Array[],\n  nativeSr: number,\n): Promise<Float32Array> {\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\n  if (totalLength === 0) return new Float32Array(0);\n\n  const fullAudio = new Float32Array(totalLength);\n  let offset = 0;\n  for (const s of samples) {\n    fullAudio.set(s, offset);\n    offset += s.length;\n  }\n\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\n\n  const duration = fullAudio.length / nativeSr;\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\n  buffer.getChannelData(0).set(fullAudio);\n  const src = offline.createBufferSource();\n  src.buffer = buffer;\n  src.connect(offline.destination);\n  src.start(0);\n  const resampled = await offline.startRendering();\n  return resampled.getChannelData(0);\n}\n\n/**\n * Stop capturing and return resampled audio at 16kHz.\n */\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\n  const { audioCtx, stream, samples, _processor } = capture;\n\n  // Disconnect processor to stop capturing\n  try {\n    _processor.disconnect();\n  } catch {\n    /* already disconnected */\n  }\n\n  // Stop microphone tracks\n  for (const track of stream.getTracks()) {\n    track.stop();\n  }\n\n  const nativeSr = audioCtx.sampleRate;\n  await audioCtx.close();\n\n  return resampleAudio(samples, nativeSr);\n}\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 ──────────────────────────────── */\n\ninterface SpeechRecognitionEvent {\n  results: SpeechRecognitionResultList;\n  resultIndex: number;\n}\n\ninterface SpeechRecognitionErrorEvent {\n  error: string;\n}\n\ninterface SpeechRecognitionInstance {\n  continuous: boolean;\n  interimResults: boolean;\n  lang: string;\n  onresult: ((e: SpeechRecognitionEvent) => void) | null;\n  onerror: ((e: SpeechRecognitionErrorEvent) => void) | null;\n  onend: (() => void) | null;\n  start: () => void;\n  stop: () => void;\n  abort: () => void;\n}\n\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\n\nfunction getSpeechRecognition(): SpeechRecognitionCtor | null {\n  if (typeof globalThis === 'undefined') return null;\n  const w = globalThis as unknown as Record<string, unknown>;\n  return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor | null;\n}\n\n/**\n * Manages Web Speech API for real-time streaming transcript preview.\n * Provides word-by-word interim text while Whisper handles corrections.\n */\nconst NO_RESULT_TIMEOUT_MS = 5_000;\n\nexport class SpeechStreamingManager {\n  private recognition: SpeechRecognitionInstance | null = null;\n  private accumulated = '';\n  private active = false;\n  private receivedResult = false;\n  private noResultTimer: ReturnType<typeof setTimeout> | null = null;\n  private onTranscript: ((text: string) => void) | null = null;\n  private onPause: (() => void) | null = null;\n  private onError: ((message: string) => void) | null = null;\n\n  /** Check if the Web Speech API is available in this environment. */\n  static isSupported(): boolean {\n    return getSpeechRecognition() !== null;\n  }\n\n  /** Set callback for streaming transcript updates (interim + final text). */\n  setOnTranscript(fn: (text: string) => void): void {\n    this.onTranscript = fn;\n  }\n\n  /** Set callback for speech pause detection (Speech API onend). */\n  setOnPause(fn: () => void): void {\n    this.onPause = fn;\n  }\n\n  /** Set callback for errors. */\n  setOnError(fn: (message: string) => void): void {\n    this.onError = fn;\n  }\n\n  /** Start streaming recognition. No-op if Speech API unavailable. */\n  start(language: string): void {\n    const SR = getSpeechRecognition();\n    if (!SR) return;\n\n    this.accumulated = '';\n    this.active = true;\n    this.receivedResult = false;\n\n    const recognition = new SR();\n    recognition.continuous = true;\n    recognition.interimResults = true;\n    recognition.lang = language;\n\n    let lastFinalIndex = -1;\n    let lastFinalText = '';\n\n    // Detect silent failure: if no onresult fires within timeout, emit error\n    this.clearNoResultTimer();\n    this.noResultTimer = setTimeout(() => {\n      if (this.active && !this.receivedResult) {\n        this.onError?.(\n          'Speech streaming started but received no results. ' +\n            'Mic may be blocked by another audio capture.',\n        );\n      }\n    }, NO_RESULT_TIMEOUT_MS);\n\n    recognition.onresult = (e: SpeechRecognitionEvent) => {\n      if (this.recognition !== recognition) return;\n      this.receivedResult = true;\n      this.clearNoResultTimer();\n\n      let final_ = '';\n      let interim = '';\n      for (let i = e.resultIndex; i < e.results.length; i++) {\n        const t = e.results[i][0].transcript;\n        if (e.results[i].isFinal) {\n          if (i > lastFinalIndex) {\n            final_ += t;\n            lastFinalIndex = i;\n          }\n        } else {\n          interim += t;\n        }\n      }\n\n      if (final_ && final_.trim() !== lastFinalText) {\n        lastFinalText = final_.trim();\n        this.accumulated = this.accumulated\n          ? this.accumulated + ' ' + final_.trim()\n          : final_.trim();\n        this.onTranscript?.(this.accumulated);\n      } else if (interim) {\n        const trimmed = interim.trimStart();\n        const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\n        this.onTranscript?.(full);\n      }\n    };\n\n    recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\n      if (this.recognition !== recognition) return;\n      this.onError?.(e.error);\n    };\n\n    recognition.onend = () => {\n      if (this.recognition !== recognition) return;\n\n      if (this.active) {\n        // Speech API paused — trigger correction\n        this.onPause?.();\n        // Restart for continued streaming\n        try {\n          recognition.start();\n        } catch {\n          this.recognition = null;\n        }\n      } else {\n        this.recognition = null;\n      }\n    };\n\n    this.recognition = recognition;\n    try {\n      recognition.start();\n    } catch {\n      this.recognition = null;\n      this.active = false;\n    }\n  }\n\n  private clearNoResultTimer(): void {\n    if (this.noResultTimer) {\n      clearTimeout(this.noResultTimer);\n      this.noResultTimer = null;\n    }\n  }\n\n  /** Stop streaming recognition and return accumulated text. */\n  stop(): string {\n    this.active = false;\n    this.clearNoResultTimer();\n    if (this.recognition) {\n      const rec = this.recognition;\n      this.recognition = null;\n      rec.stop();\n    }\n    const result = this.accumulated;\n    this.accumulated = '';\n    return result;\n  }\n\n  /** Abort immediately without returning text. */\n  destroy(): void {\n    this.active = false;\n    this.clearNoResultTimer();\n    if (this.recognition) {\n      const rec = this.recognition;\n      this.recognition = null;\n      rec.abort();\n    }\n    this.accumulated = '';\n    this.onTranscript = null;\n    this.onPause = null;\n    this.onError = null;\n  }\n}\n","import type {\n  STTConfig,\n  STTState,\n  STTEvents,\n  STTStatus,\n  ResolvedSTTConfig,\n  AudioCaptureHandle,\n} from './types.js';\nimport { resolveConfig } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\nimport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\nimport { WorkerManager } from './worker-manager.js';\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\nimport { SpeechStreamingManager } from './speech-streaming.js';\n\n/**\n * Main STT engine — the public API for speech-to-text with Whisper correction.\n *\n * Usage:\n * ```typescript\n * const engine = new STTEngine({ model: 'tiny' });\n * engine.on('transcript', (text) => console.log(text));\n * engine.on('correction', (text) => console.log('corrected:', text));\n * await engine.init();\n * await engine.start();\n * const finalText = await engine.stop();\n * ```\n */\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\n  private config: ResolvedSTTConfig;\n  private workerManager: WorkerManager;\n  private correctionOrchestrator: CorrectionOrchestrator;\n  private speechStreaming: SpeechStreamingManager;\n  private capture: AudioCaptureHandle | null = null;\n  private state: STTState;\n  private workerUrl?: URL;\n\n  /**\n   * Create a new STT engine instance.\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\n   */\n  constructor(config?: STTConfig, workerUrl?: URL) {\n    super();\n    this.config = resolveConfig(config);\n    this.workerManager = new WorkerManager();\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\n    this.speechStreaming = new SpeechStreamingManager();\n    this.workerUrl = workerUrl;\n\n    this.state = {\n      status: 'idle',\n      isModelLoaded: false,\n      loadProgress: 0,\n      backend: null,\n      error: null,\n    };\n\n    this.correctionOrchestrator.setCorrectionFn(() => {\n      this.performCorrection();\n    });\n\n    this.setupWorkerListeners();\n    this.setupStreamingCallbacks();\n  }\n\n  /** Initialize the engine: spawn worker and load model. */\n  async init(): Promise<void> {\n    this.updateStatus('loading');\n    this.workerManager.spawn(this.workerUrl);\n\n    try {\n      await this.workerManager.loadModel(this.config);\n      this.state.isModelLoaded = true;\n      this.updateStatus('ready');\n    } catch (err) {\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\n      this.updateStatus('idle');\n      throw err;\n    }\n  }\n\n  /** Start recording audio and enable correction cycles. */\n  async start(): Promise<void> {\n    if (this.state.status !== 'ready') {\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\n    }\n\n    try {\n      // Start Speech API BEFORE getUserMedia — avoids mic conflict where\n      // the second mic accessor silently fails (no errors, no events).\n      // This matches the working ClaudeWebCLI order.\n      if (this.config.streaming.enabled) {\n        this.speechStreaming.start(this.config.language);\n      }\n      this.capture = await startCapture();\n      this.updateStatus('recording');\n      this.correctionOrchestrator.start();\n    } catch (err) {\n      this.emitError(\n        'MIC_DENIED',\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\n      );\n    }\n  }\n\n  /** Stop recording, run final transcription, return text. */\n  async stop(): Promise<string> {\n    if (!this.capture) return '';\n\n    this.correctionOrchestrator.stop();\n    this.speechStreaming.stop();\n    this.workerManager.cancel();\n\n    this.updateStatus('processing');\n\n    try {\n      const audio = await stopCapture(this.capture);\n      this.capture = null;\n\n      if (audio.length === 0) {\n        this.updateStatus('ready');\n        return '';\n      }\n\n      const text = await this.workerManager.transcribe(audio);\n      this.emit('correction', text);\n      this.updateStatus('ready');\n      return text;\n    } catch (err) {\n      this.emitError(\n        'TRANSCRIPTION_FAILED',\n        err instanceof Error ? err.message : 'Final transcription failed.',\n      );\n      this.updateStatus('ready');\n      return '';\n    }\n  }\n\n  /** Destroy the engine: terminate worker, release all resources. */\n  destroy(): void {\n    this.correctionOrchestrator.stop();\n    this.speechStreaming.destroy();\n\n    if (this.capture) {\n      for (const track of this.capture.stream.getTracks()) {\n        track.stop();\n      }\n      this.capture.audioCtx.close().catch(() => {});\n      this.capture = null;\n    }\n\n    this.workerManager.destroy();\n    this.updateStatus('idle');\n    this.removeAllListeners();\n  }\n\n  /** Get current engine state. */\n  getState(): Readonly<STTState> {\n    return { ...this.state };\n  }\n\n  /** Notify the correction orchestrator of a speech pause. */\n  notifyPause(): void {\n    this.correctionOrchestrator.onPauseDetected();\n  }\n\n  private async performCorrection(): Promise<void> {\n    if (!this.capture || !this.state.isModelLoaded) return;\n\n    this.workerManager.cancel();\n\n    try {\n      const samples = snapshotAudio(this.capture);\n      const nativeSr = this.capture.audioCtx.sampleRate;\n      const audio = await resampleAudio(samples, 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package/dist/index.d.cts

@@ -4,15 +4,28 @@ type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';
 type STTBackend = 'webgpu' | 'wasm' | 'auto';
 /** Engine lifecycle states. */
 type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';
-/** Correction timing configuration. */
+/** Supported correction engine providers. */
+type STTCorrectionProvider = 'whisper';
+/** Supported real-time streaming providers. */
+type STTStreamingProvider = 'web-speech-api';
+/** Correction engine configuration. */
 interface STTCorrectionConfig {
-    /** Enable mid-recording Whisper correction. Default: true */
+    /** Enable mid-recording correction. Default: true */
     enabled?: boolean;
+    /** Correction engine provider. Default: 'whisper' */
+    provider?: STTCorrectionProvider;
     /** Silence duration (ms) before triggering correction. Default: 3000 */
     pauseThreshold?: number;
     /** Maximum interval (ms) between forced corrections. Default: 5000 */
     forcedInterval?: number;
 }
+/** Real-time streaming preview configuration. */
+interface STTStreamingConfig {
+    /** Enable real-time streaming transcript. Default: true */
+    enabled?: boolean;
+    /** Streaming provider. Default: 'web-speech-api' */
+    provider?: STTStreamingProvider;
+}
 /** Audio chunking configuration for long-form audio. */
 interface STTChunkingConfig {
     /** Chunk length in seconds for Whisper processing. Default: 30 */
@@ -34,6 +47,8 @@ interface STTConfig {
     correction?: STTCorrectionConfig;
     /** Audio chunking settings for long-form audio. */
     chunking?: STTChunkingConfig;
+    /** Web Speech API streaming preview settings. */
+    streaming?: STTStreamingConfig;
 }
 /** Resolved configuration with all defaults applied. */
 interface ResolvedSTTConfig {
@@ -43,6 +58,7 @@ interface ResolvedSTTConfig {
     dtype: string;
     correction: Required<STTCorrectionConfig>;
     chunking: Required<STTChunkingConfig>;
+    streaming: Required<STTStreamingConfig>;
 }
 /** Engine state exposed to consumers via status events. */
 interface STTState {
@@ -176,6 +192,32 @@ declare class CorrectionOrchestrator {
     private restartForcedTimer;
 }
 
+declare class SpeechStreamingManager {
+    private recognition;
+    private accumulated;
+    private active;
+    private receivedResult;
+    private noResultTimer;
+    private onTranscript;
+    private onPause;
+    private onError;
+    /** Check if the Web Speech API is available in this environment. */
+    static isSupported(): boolean;
+    /** Set callback for streaming transcript updates (interim + final text). */
+    setOnTranscript(fn: (text: string) => void): void;
+    /** Set callback for speech pause detection (Speech API onend). */
+    setOnPause(fn: () => void): void;
+    /** Set callback for errors. */
+    setOnError(fn: (message: string) => void): void;
+    /** Start streaming recognition. No-op if Speech API unavailable. */
+    start(language: string): void;
+    private clearNoResultTimer;
+    /** Stop streaming recognition and return accumulated text. */
+    stop(): string;
+    /** Abort immediately without returning text. */
+    destroy(): void;
+}
+
 /**
  * Main STT engine — the public API for speech-to-text with Whisper correction.
  *
@@ -193,6 +235,7 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     private config;
     private workerManager;
     private correctionOrchestrator;
+    private speechStreaming;
     private capture;
     private state;
     private workerUrl?;
@@ -215,9 +258,10 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     /** Notify the correction orchestrator of a speech pause. */
     notifyPause(): void;
     private performCorrection;
+    private setupStreamingCallbacks;
     private setupWorkerListeners;
     private updateStatus;
     private emitError;
 }
 
-export { type AudioCaptureHandle, CorrectionOrchestrator, DEFAULT_STT_CONFIG, type ResolvedSTTConfig, type STTBackend, type STTChunkingConfig, type STTConfig, type STTCorrectionConfig, STTEngine, type STTError, type STTEvents, type STTModelSize, type STTState, type STTStatus, 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, resampleAudio, resolveConfig, snapshotAudio, startCapture, stopCapture };

package/dist/index.d.ts

@@ -4,15 +4,28 @@ type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';
 type STTBackend = 'webgpu' | 'wasm' | 'auto';
 /** Engine lifecycle states. */
 type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';
-/** Correction timing configuration. */
+/** Supported correction engine providers. */
+type STTCorrectionProvider = 'whisper';
+/** Supported real-time streaming providers. */
+type STTStreamingProvider = 'web-speech-api';
+/** Correction engine configuration. */
 interface STTCorrectionConfig {
-    /** Enable mid-recording Whisper correction. Default: true */
+    /** Enable mid-recording correction. Default: true */
     enabled?: boolean;
+    /** Correction engine provider. Default: 'whisper' */
+    provider?: STTCorrectionProvider;
     /** Silence duration (ms) before triggering correction. Default: 3000 */
     pauseThreshold?: number;
     /** Maximum interval (ms) between forced corrections. Default: 5000 */
     forcedInterval?: number;
 }
+/** Real-time streaming preview configuration. */
+interface STTStreamingConfig {
+    /** Enable real-time streaming transcript. Default: true */
+    enabled?: boolean;
+    /** Streaming provider. Default: 'web-speech-api' */
+    provider?: STTStreamingProvider;
+}
 /** Audio chunking configuration for long-form audio. */
 interface STTChunkingConfig {
     /** Chunk length in seconds for Whisper processing. Default: 30 */
@@ -34,6 +47,8 @@ interface STTConfig {
     correction?: STTCorrectionConfig;
     /** Audio chunking settings for long-form audio. */
     chunking?: STTChunkingConfig;
+    /** Web Speech API streaming preview settings. */
+    streaming?: STTStreamingConfig;
 }
 /** Resolved configuration with all defaults applied. */
 interface ResolvedSTTConfig {
@@ -43,6 +58,7 @@ interface ResolvedSTTConfig {
     dtype: string;
     correction: Required<STTCorrectionConfig>;
     chunking: Required<STTChunkingConfig>;
+    streaming: Required<STTStreamingConfig>;
 }
 /** Engine state exposed to consumers via status events. */
 interface STTState {
@@ -176,6 +192,32 @@ declare class CorrectionOrchestrator {
     private restartForcedTimer;
 }
 
+declare class SpeechStreamingManager {
+    private recognition;
+    private accumulated;
+    private active;
+    private receivedResult;
+    private noResultTimer;
+    private onTranscript;
+    private onPause;
+    private onError;
+    /** Check if the Web Speech API is available in this environment. */
+    static isSupported(): boolean;
+    /** Set callback for streaming transcript updates (interim + final text). */
+    setOnTranscript(fn: (text: string) => void): void;
+    /** Set callback for speech pause detection (Speech API onend). */
+    setOnPause(fn: () => void): void;
+    /** Set callback for errors. */
+    setOnError(fn: (message: string) => void): void;
+    /** Start streaming recognition. No-op if Speech API unavailable. */
+    start(language: string): void;
+    private clearNoResultTimer;
+    /** Stop streaming recognition and return accumulated text. */
+    stop(): string;
+    /** Abort immediately without returning text. */
+    destroy(): void;
+}
+
 /**
  * Main STT engine — the public API for speech-to-text with Whisper correction.
  *
@@ -193,6 +235,7 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     private config;
     private workerManager;
     private correctionOrchestrator;
+    private speechStreaming;
     private capture;
     private state;
     private workerUrl?;
@@ -215,9 +258,10 @@ declare class STTEngine extends TypedEventEmitter<STTEvents> {
     /** Notify the correction orchestrator of a speech pause. */
     notifyPause(): void;
     private performCorrection;
+    private setupStreamingCallbacks;
     private setupWorkerListeners;
     private updateStatus;
     private emitError;
 }
 
-export { type AudioCaptureHandle, CorrectionOrchestrator, DEFAULT_STT_CONFIG, type ResolvedSTTConfig, type STTBackend, type STTChunkingConfig, type STTConfig, type STTCorrectionConfig, STTEngine, type STTError, type STTEvents, type STTModelSize, type STTState, type STTStatus, 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, resampleAudio, resolveConfig, snapshotAudio, startCapture, stopCapture };

package/dist/index.js

@@ -6,12 +6,17 @@ var DEFAULT_STT_CONFIG = {
   dtype: "q4",
   correction: {
     enabled: true,
+    provider: "whisper",
     pauseThreshold: 3e3,
     forcedInterval: 5e3
   },
   chunking: {
     chunkLengthS: 30,
     strideLengthS: 5
+  },
+  streaming: {
+    enabled: true,
+    provider: "web-speech-api"
   }
 };
 function resolveConfig(config) {
@@ -22,12 +27,17 @@ function resolveConfig(config) {
     dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,
     correction: {
       enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,
+      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,
       pauseThreshold: config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,
       forcedInterval: config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval
     },
     chunking: {
       chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,
       strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS
+    },
+    streaming: {
+      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,
+      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider
     }
   };
 }
@@ -282,11 +292,152 @@ var CorrectionOrchestrator = class {
   }
 };
 
+// src/speech-streaming.ts
+function getSpeechRecognition() {
+  if (typeof globalThis === "undefined") return null;
+  const w = globalThis;
+  return w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null;
+}
+var NO_RESULT_TIMEOUT_MS = 5e3;
+var SpeechStreamingManager = class {
+  recognition = null;
+  accumulated = "";
+  active = false;
+  receivedResult = false;
+  noResultTimer = null;
+  onTranscript = null;
+  onPause = null;
+  onError = null;
+  /** Check if the Web Speech API is available in this environment. */
+  static isSupported() {
+    return getSpeechRecognition() !== null;
+  }
+  /** Set callback for streaming transcript updates (interim + final text). */
+  setOnTranscript(fn) {
+    this.onTranscript = fn;
+  }
+  /** Set callback for speech pause detection (Speech API onend). */
+  setOnPause(fn) {
+    this.onPause = fn;
+  }
+  /** Set callback for errors. */
+  setOnError(fn) {
+    this.onError = fn;
+  }
+  /** Start streaming recognition. No-op if Speech API unavailable. */
+  start(language) {
+    const SR = getSpeechRecognition();
+    if (!SR) return;
+    this.accumulated = "";
+    this.active = true;
+    this.receivedResult = false;
+    const recognition = new SR();
+    recognition.continuous = true;
+    recognition.interimResults = true;
+    recognition.lang = language;
+    let lastFinalIndex = -1;
+    let lastFinalText = "";
+    this.clearNoResultTimer();
+    this.noResultTimer = setTimeout(() => {
+      if (this.active && !this.receivedResult) {
+        this.onError?.(
+          "Speech streaming started but received no results. Mic may be blocked by another audio capture."
+        );
+      }
+    }, NO_RESULT_TIMEOUT_MS);
+    recognition.onresult = (e) => {
+      if (this.recognition !== recognition) return;
+      this.receivedResult = true;
+      this.clearNoResultTimer();
+      let final_ = "";
+      let interim = "";
+      for (let i = e.resultIndex; i < e.results.length; i++) {
+        const t = e.results[i][0].transcript;
+        if (e.results[i].isFinal) {
+          if (i > lastFinalIndex) {
+            final_ += t;
+            lastFinalIndex = i;
+          }
+        } else {
+          interim += t;
+        }
+      }
+      if (final_ && final_.trim() !== lastFinalText) {
+        lastFinalText = final_.trim();
+        this.accumulated = this.accumulated ? this.accumulated + " " + final_.trim() : final_.trim();
+        this.onTranscript?.(this.accumulated);
+      } else if (interim) {
+        const trimmed = interim.trimStart();
+        const full = this.accumulated ? this.accumulated + " " + trimmed : trimmed;
+        this.onTranscript?.(full);
+      }
+    };
+    recognition.onerror = (e) => {
+      if (this.recognition !== recognition) return;
+      this.onError?.(e.error);
+    };
+    recognition.onend = () => {
+      if (this.recognition !== recognition) return;
+      if (this.active) {
+        this.onPause?.();
+        try {
+          recognition.start();
+        } catch {
+          this.recognition = null;
+        }
+      } else {
+        this.recognition = null;
+      }
+    };
+    this.recognition = recognition;
+    try {
+      recognition.start();
+    } catch {
+      this.recognition = null;
+      this.active = false;
+    }
+  }
+  clearNoResultTimer() {
+    if (this.noResultTimer) {
+      clearTimeout(this.noResultTimer);
+      this.noResultTimer = null;
+    }
+  }
+  /** Stop streaming recognition and return accumulated text. */
+  stop() {
+    this.active = false;
+    this.clearNoResultTimer();
+    if (this.recognition) {
+      const rec = this.recognition;
+      this.recognition = null;
+      rec.stop();
+    }
+    const result = this.accumulated;
+    this.accumulated = "";
+    return result;
+  }
+  /** Abort immediately without returning text. */
+  destroy() {
+    this.active = false;
+    this.clearNoResultTimer();
+    if (this.recognition) {
+      const rec = this.recognition;
+      this.recognition = null;
+      rec.abort();
+    }
+    this.accumulated = "";
+    this.onTranscript = null;
+    this.onPause = null;
+    this.onError = null;
+  }
+};
+
 // src/stt-engine.ts
 var STTEngine = class extends TypedEventEmitter {
   config;
   workerManager;
   correctionOrchestrator;
+  speechStreaming;
   capture = null;
   state;
   workerUrl;
@@ -300,6 +451,7 @@ var STTEngine = class extends TypedEventEmitter {
     this.config = resolveConfig(config);
     this.workerManager = new WorkerManager();
     this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);
+    this.speechStreaming = new SpeechStreamingManager();
     this.workerUrl = workerUrl;
     this.state = {
       status: "idle",
@@ -312,6 +464,7 @@ var STTEngine = class extends TypedEventEmitter {
       this.performCorrection();
     });
     this.setupWorkerListeners();
+    this.setupStreamingCallbacks();
   }
   /** Initialize the engine: spawn worker and load model. */
   async init() {
@@ -333,6 +486,9 @@ var STTEngine = class extends TypedEventEmitter {
       throw new Error(`Cannot start: engine is "${this.state.status}", expected "ready"`);
     }
     try {
+      if (this.config.streaming.enabled) {
+        this.speechStreaming.start(this.config.language);
+      }
       this.capture = await startCapture();
       this.updateStatus("recording");
       this.correctionOrchestrator.start();
@@ -347,6 +503,7 @@ var STTEngine = class extends TypedEventEmitter {
   async stop() {
     if (!this.capture) return "";
     this.correctionOrchestrator.stop();
+    this.speechStreaming.stop();
     this.workerManager.cancel();
     this.updateStatus("processing");
     try {
@@ -372,6 +529,7 @@ var STTEngine = class extends TypedEventEmitter {
   /** Destroy the engine: terminate worker, release all resources. */
   destroy() {
     this.correctionOrchestrator.stop();
+    this.speechStreaming.destroy();
     if (this.capture) {
       for (const track of this.capture.stream.getTracks()) {
         track.stop();
@@ -411,6 +569,17 @@ var STTEngine = class extends TypedEventEmitter {
       );
     }
   }
+  setupStreamingCallbacks() {
+    this.speechStreaming.setOnTranscript((text) => {
+      this.emit("transcript", text);
+    });
+    this.speechStreaming.setOnPause(() => {
+      this.correctionOrchestrator.onPauseDetected();
+    });
+    this.speechStreaming.setOnError((message) => {
+      this.emitError("STREAMING_ERROR", message);
+    });
+  }
   setupWorkerListeners() {
     this.workerManager.on("progress", (percent) => {
       this.state.loadProgress = percent;
@@ -434,6 +603,7 @@ export {
   CorrectionOrchestrator,
   DEFAULT_STT_CONFIG,
   STTEngine,
+  SpeechStreamingManager,
   TypedEventEmitter,
   WorkerManager,
   resampleAudio,

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/stt-engine.ts"],"sourcesContent":["/** Supported Whisper model sizes. */\nexport type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';\n\n/** Supported compute backends. */\nexport type STTBackend = 'webgpu' | 'wasm' | 'auto';\n\n/** Engine lifecycle states. */\nexport type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';\n\n/** Correction timing configuration. */\nexport interface STTCorrectionConfig {\n  /** Enable mid-recording Whisper correction. Default: true */\n  enabled?: boolean;\n  /** Silence duration (ms) before triggering correction. Default: 3000 */\n  pauseThreshold?: number;\n  /** Maximum interval (ms) between forced corrections. Default: 5000 */\n  forcedInterval?: number;\n}\n\n/** Audio chunking configuration for long-form audio. */\nexport interface STTChunkingConfig {\n  /** Chunk length in seconds for Whisper processing. Default: 30 */\n  chunkLengthS?: number;\n  /** Stride length in seconds for overlapping chunks. Default: 5 */\n  strideLengthS?: number;\n}\n\n/** Full engine configuration. All fields optional — sensible defaults applied. */\nexport interface STTConfig {\n  /** Whisper model size. Default: 'tiny' */\n  model?: STTModelSize;\n  /** Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) */\n  backend?: STTBackend;\n  /** Transcription language. Default: 'en' */\n  language?: string;\n  /** Model quantization dtype. Default: 'q4' */\n  dtype?: string;\n  /** Mid-recording correction settings. */\n  correction?: STTCorrectionConfig;\n  /** Audio chunking settings for long-form audio. */\n  chunking?: STTChunkingConfig;\n}\n\n/** Resolved configuration with all defaults applied. */\nexport interface ResolvedSTTConfig {\n  model: STTModelSize;\n  backend: STTBackend;\n  language: string;\n  dtype: string;\n  correction: Required<STTCorrectionConfig>;\n  chunking: Required<STTChunkingConfig>;\n}\n\n/** Engine state exposed to consumers via status events. */\nexport interface STTState {\n  status: STTStatus;\n  isModelLoaded: boolean;\n  /** Model download progress (0–100). */\n  loadProgress: number;\n  /** Active compute backend, or null if not yet determined. */\n  backend: 'webgpu' | 'wasm' | null;\n  error: string | null;\n}\n\n/** Structured error emitted via the 'error' event. */\nexport interface STTError {\n  code: string;\n  message: string;\n}\n\n/** Event map for the typed event emitter. */\nexport type STTEvents = {\n  /** Streaming interim text during recording. */\n  transcript: (text: string) => void;\n  /** Whisper-corrected text replacing interim text. */\n  correction: (text: string) => void;\n  /** Actionable error (mic denied, model fail, transcription fail). */\n  error: (error: STTError) => void;\n  /** Engine state change. */\n  status: (state: STTState) => void;\n};\n\n/** Handle returned by audio capture — used internally. */\nexport interface AudioCaptureHandle {\n  audioCtx: AudioContext;\n  stream: MediaStream;\n  samples: Float32Array[];\n  /** Retain reference to prevent GC from stopping audio processing. */\n  _processor: ScriptProcessorNode;\n}\n\n/** Message sent from main thread to Whisper worker. */\nexport interface WorkerMessage {\n  type: 'load' | 'transcribe' | 'cancel';\n  audio?: Float32Array;\n  config?: {\n    model: string;\n    backend: STTBackend;\n    language: string;\n    dtype: string;\n    chunkLengthS: number;\n    strideLengthS: number;\n  };\n}\n\n/** Response sent from Whisper worker to main thread. */\nexport interface WorkerResponse {\n  type: 'progress' | 'ready' | 'result' | 'error';\n  data?: unknown;\n}\n\n/** Default configuration values. */\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\n  model: 'tiny',\n  backend: 'auto',\n  language: 'en',\n  dtype: 'q4',\n  correction: {\n    enabled: true,\n    pauseThreshold: 3_000,\n    forcedInterval: 5_000,\n  },\n  chunking: {\n    chunkLengthS: 30,\n    strideLengthS: 5,\n  },\n};\n\n/** Merge user config with defaults to produce resolved config. */\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\n  return {\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\n    correction: {\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\n      pauseThreshold:\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\n      forcedInterval:\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\n    },\n    chunking: {\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\n    },\n  };\n}\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';\n\nconst TARGET_SAMPLE_RATE = 16_000;\n\n/**\n * Start capturing raw PCM audio from the microphone.\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\n */\nexport async function startCapture(): Promise<AudioCaptureHandle> {\n  const stream = await navigator.mediaDevices.getUserMedia({\n    audio: { channelCount: 1 },\n  });\n  const audioCtx = new AudioContext();\n\n  // Chrome may suspend AudioContext — must resume within user gesture\n  if (audioCtx.state === 'suspended') {\n    await audioCtx.resume();\n  }\n\n  const source = audioCtx.createMediaStreamSource(stream);\n  const samples: Float32Array[] = [];\n\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\n  };\n\n  // Connect through a silent gain node so mic audio doesn't play back\n  const silencer = audioCtx.createGain();\n  silencer.gain.value = 0;\n  source.connect(processor);\n  processor.connect(silencer);\n  silencer.connect(audioCtx.destination);\n\n  return { audioCtx, stream, samples, _processor: processor };\n}\n\n/**\n * Copy current audio buffer without stopping capture.\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\n */\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\n  return [...capture.samples];\n}\n\n/**\n * Concatenate sample chunks and resample to 16kHz for Whisper.\n */\nexport async function resampleAudio(\n  samples: Float32Array[],\n  nativeSr: number,\n): Promise<Float32Array> {\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\n  if (totalLength === 0) return new Float32Array(0);\n\n  const fullAudio = new Float32Array(totalLength);\n  let offset = 0;\n  for (const s of samples) {\n    fullAudio.set(s, offset);\n    offset += s.length;\n  }\n\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\n\n  const duration = fullAudio.length / nativeSr;\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\n  buffer.getChannelData(0).set(fullAudio);\n  const src = offline.createBufferSource();\n  src.buffer = buffer;\n  src.connect(offline.destination);\n  src.start(0);\n  const resampled = await offline.startRendering();\n  return resampled.getChannelData(0);\n}\n\n/**\n * Stop capturing and return resampled audio at 16kHz.\n */\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\n  const { audioCtx, stream, samples, _processor } = capture;\n\n  // Disconnect processor to stop capturing\n  try {\n    _processor.disconnect();\n  } catch {\n    /* already disconnected */\n  }\n\n  // Stop microphone tracks\n  for (const track of stream.getTracks()) {\n    track.stop();\n  }\n\n  const nativeSr = audioCtx.sampleRate;\n  await audioCtx.close();\n\n  return resampleAudio(samples, nativeSr);\n}\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","import type {\n  STTConfig,\n  STTState,\n  STTEvents,\n  STTStatus,\n  ResolvedSTTConfig,\n  AudioCaptureHandle,\n} from './types.js';\nimport { resolveConfig } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\nimport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\nimport { WorkerManager } from './worker-manager.js';\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\n\n/**\n * Main STT engine — the public API for speech-to-text with Whisper correction.\n *\n * Usage:\n * ```typescript\n * const engine = new STTEngine({ model: 'tiny' });\n * engine.on('transcript', (text) => console.log(text));\n * engine.on('correction', (text) => console.log('corrected:', text));\n * await engine.init();\n * await engine.start();\n * const finalText = await engine.stop();\n * ```\n */\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\n  private config: ResolvedSTTConfig;\n  private workerManager: WorkerManager;\n  private correctionOrchestrator: CorrectionOrchestrator;\n  private capture: AudioCaptureHandle | null = null;\n  private state: STTState;\n  private workerUrl?: URL;\n\n  /**\n   * Create a new STT engine instance.\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\n   */\n  constructor(config?: STTConfig, workerUrl?: URL) {\n    super();\n    this.config = resolveConfig(config);\n    this.workerManager = new WorkerManager();\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\n    this.workerUrl = workerUrl;\n\n    this.state = {\n      status: 'idle',\n      isModelLoaded: false,\n      loadProgress: 0,\n      backend: null,\n      error: null,\n    };\n\n    this.correctionOrchestrator.setCorrectionFn(() => {\n      this.performCorrection();\n    });\n\n    this.setupWorkerListeners();\n  }\n\n  /** Initialize the engine: spawn worker and load model. */\n  async init(): Promise<void> {\n    this.updateStatus('loading');\n    this.workerManager.spawn(this.workerUrl);\n\n    try {\n      await this.workerManager.loadModel(this.config);\n      this.state.isModelLoaded = true;\n      this.updateStatus('ready');\n    } catch (err) {\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\n      this.updateStatus('idle');\n      throw err;\n    }\n  }\n\n  /** Start recording audio and enable correction cycles. */\n  async start(): Promise<void> {\n    if (this.state.status !== 'ready') {\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\n    }\n\n    try {\n      this.capture = await startCapture();\n      this.updateStatus('recording');\n      this.correctionOrchestrator.start();\n    } catch (err) {\n      this.emitError(\n        'MIC_DENIED',\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\n      );\n    }\n  }\n\n  /** Stop recording, run final transcription, return text. */\n  async stop(): Promise<string> {\n    if (!this.capture) return '';\n\n    this.correctionOrchestrator.stop();\n    this.workerManager.cancel();\n\n    this.updateStatus('processing');\n\n    try {\n      const audio = await stopCapture(this.capture);\n      this.capture = null;\n\n      if (audio.length === 0) {\n        this.updateStatus('ready');\n        return '';\n      }\n\n      const text = await this.workerManager.transcribe(audio);\n      this.emit('correction', text);\n      this.updateStatus('ready');\n      return text;\n    } catch (err) {\n      this.emitError(\n        'TRANSCRIPTION_FAILED',\n        err instanceof Error ? err.message : 'Final transcription failed.',\n      );\n      this.updateStatus('ready');\n      return '';\n    }\n  }\n\n  /** Destroy the engine: terminate worker, release all resources. */\n  destroy(): void {\n    this.correctionOrchestrator.stop();\n\n    if (this.capture) {\n      for (const track of this.capture.stream.getTracks()) {\n        track.stop();\n      }\n      this.capture.audioCtx.close().catch(() => {});\n      this.capture = null;\n    }\n\n    this.workerManager.destroy();\n    this.updateStatus('idle');\n    this.removeAllListeners();\n  }\n\n  /** Get current engine state. */\n  getState(): Readonly<STTState> {\n    return { ...this.state };\n  }\n\n  /** Notify the correction orchestrator of a speech pause. */\n  notifyPause(): void {\n    this.correctionOrchestrator.onPauseDetected();\n  }\n\n  private async performCorrection(): Promise<void> {\n    if (!this.capture || !this.state.isModelLoaded) return;\n\n    this.workerManager.cancel();\n\n    try {\n      const samples = snapshotAudio(this.capture);\n      const nativeSr = this.capture.audioCtx.sampleRate;\n      const audio = await resampleAudio(samples, nativeSr);\n\n      if (audio.length === 0) return;\n\n      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+{"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. */\nexport type STTModelSize = 'tiny' | 'base' | 'small' | 'medium';\n\n/** Supported compute backends. */\nexport type STTBackend = 'webgpu' | 'wasm' | 'auto';\n\n/** Engine lifecycle states. */\nexport type STTStatus = 'idle' | 'loading' | 'ready' | 'recording' | 'processing';\n\n/** Supported correction engine providers. */\nexport type STTCorrectionProvider = 'whisper';\n\n/** Supported real-time streaming providers. */\nexport type STTStreamingProvider = 'web-speech-api';\n\n/** Correction engine configuration. */\nexport interface STTCorrectionConfig {\n  /** Enable mid-recording correction. Default: true */\n  enabled?: boolean;\n  /** Correction engine provider. 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Default: 'q4' */\n  dtype?: string;\n  /** Mid-recording correction settings. */\n  correction?: STTCorrectionConfig;\n  /** Audio chunking settings for long-form audio. */\n  chunking?: STTChunkingConfig;\n  /** Web Speech API streaming preview settings. */\n  streaming?: STTStreamingConfig;\n}\n\n/** Resolved configuration with all defaults applied. */\nexport interface ResolvedSTTConfig {\n  model: STTModelSize;\n  backend: STTBackend;\n  language: string;\n  dtype: string;\n  correction: Required<STTCorrectionConfig>;\n  chunking: Required<STTChunkingConfig>;\n  streaming: Required<STTStreamingConfig>;\n}\n\n/** Engine state exposed to consumers via status events. */\nexport interface STTState {\n  status: STTStatus;\n  isModelLoaded: boolean;\n  /** Model download progress (0–100). */\n  loadProgress: number;\n  /** Active compute backend, or null if not yet determined. */\n  backend: 'webgpu' | 'wasm' | null;\n  error: string | null;\n}\n\n/** Structured error emitted via the 'error' event. */\nexport interface STTError {\n  code: string;\n  message: string;\n}\n\n/** Event map for the typed event emitter. */\nexport type STTEvents = {\n  /** Streaming interim text during recording. */\n  transcript: (text: string) => void;\n  /** Whisper-corrected text replacing interim text. */\n  correction: (text: string) => void;\n  /** Actionable error (mic denied, model fail, transcription fail). */\n  error: (error: STTError) => void;\n  /** Engine state change. */\n  status: (state: STTState) => void;\n};\n\n/** Handle returned by audio capture — used internally. */\nexport interface AudioCaptureHandle {\n  audioCtx: AudioContext;\n  stream: MediaStream;\n  samples: Float32Array[];\n  /** Retain reference to prevent GC from stopping audio processing. */\n  _processor: ScriptProcessorNode;\n}\n\n/** Message sent from main thread to Whisper worker. */\nexport interface WorkerMessage {\n  type: 'load' | 'transcribe' | 'cancel';\n  audio?: Float32Array;\n  config?: {\n    model: string;\n    backend: STTBackend;\n    language: string;\n    dtype: string;\n    chunkLengthS: number;\n    strideLengthS: number;\n  };\n}\n\n/** Response sent from Whisper worker to main thread. */\nexport interface WorkerResponse {\n  type: 'progress' | 'ready' | 'result' | 'error';\n  data?: unknown;\n}\n\n/** Default configuration values. */\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\n  model: 'tiny',\n  backend: 'auto',\n  language: 'en',\n  dtype: 'q4',\n  correction: {\n    enabled: true,\n    provider: 'whisper',\n    pauseThreshold: 3_000,\n    forcedInterval: 5_000,\n  },\n  chunking: {\n    chunkLengthS: 30,\n    strideLengthS: 5,\n  },\n  streaming: {\n    enabled: true,\n    provider: 'web-speech-api',\n  },\n};\n\n/** Merge user config with defaults to produce resolved config. */\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\n  return {\n    model: config?.model ?? DEFAULT_STT_CONFIG.model,\n    backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\n    language: config?.language ?? DEFAULT_STT_CONFIG.language,\n    dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\n    correction: {\n      enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\n      provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\n      pauseThreshold:\n        config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\n      forcedInterval:\n        config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\n    },\n    chunking: {\n      chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\n      strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\n    },\n    streaming: {\n      enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\n      provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\n    },\n  };\n}\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';\n\nconst TARGET_SAMPLE_RATE = 16_000;\n\n/**\n * Start capturing raw PCM audio from the microphone.\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\n */\nexport async function startCapture(): Promise<AudioCaptureHandle> {\n  const stream = await navigator.mediaDevices.getUserMedia({\n    audio: { channelCount: 1 },\n  });\n  const audioCtx = new AudioContext();\n\n  // Chrome may suspend AudioContext — must resume within user gesture\n  if (audioCtx.state === 'suspended') {\n    await audioCtx.resume();\n  }\n\n  const source = audioCtx.createMediaStreamSource(stream);\n  const samples: Float32Array[] = [];\n\n  const processor = audioCtx.createScriptProcessor(4096, 1, 1);\n  processor.onaudioprocess = (e: AudioProcessingEvent) => {\n    samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\n  };\n\n  // Connect through a silent gain node so mic audio doesn't play back\n  const silencer = audioCtx.createGain();\n  silencer.gain.value = 0;\n  source.connect(processor);\n  processor.connect(silencer);\n  silencer.connect(audioCtx.destination);\n\n  return { audioCtx, stream, samples, _processor: processor };\n}\n\n/**\n * Copy current audio buffer without stopping capture.\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\n */\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\n  return [...capture.samples];\n}\n\n/**\n * Concatenate sample chunks and resample to 16kHz for Whisper.\n */\nexport async function resampleAudio(\n  samples: Float32Array[],\n  nativeSr: number,\n): Promise<Float32Array> {\n  const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\n  if (totalLength === 0) return new Float32Array(0);\n\n  const fullAudio = new Float32Array(totalLength);\n  let offset = 0;\n  for (const s of samples) {\n    fullAudio.set(s, offset);\n    offset += s.length;\n  }\n\n  if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\n\n  const duration = fullAudio.length / nativeSr;\n  const outLength = Math.round(duration * TARGET_SAMPLE_RATE);\n  const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\n  const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\n  buffer.getChannelData(0).set(fullAudio);\n  const src = offline.createBufferSource();\n  src.buffer = buffer;\n  src.connect(offline.destination);\n  src.start(0);\n  const resampled = await offline.startRendering();\n  return resampled.getChannelData(0);\n}\n\n/**\n * Stop capturing and return resampled audio at 16kHz.\n */\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\n  const { audioCtx, stream, samples, _processor } = capture;\n\n  // Disconnect processor to stop capturing\n  try {\n    _processor.disconnect();\n  } catch {\n    /* already disconnected */\n  }\n\n  // Stop microphone tracks\n  for (const track of stream.getTracks()) {\n    track.stop();\n  }\n\n  const nativeSr = audioCtx.sampleRate;\n  await audioCtx.close();\n\n  return resampleAudio(samples, nativeSr);\n}\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 ──────────────────────────────── */\n\ninterface SpeechRecognitionEvent {\n  results: SpeechRecognitionResultList;\n  resultIndex: number;\n}\n\ninterface SpeechRecognitionErrorEvent {\n  error: string;\n}\n\ninterface SpeechRecognitionInstance {\n  continuous: boolean;\n  interimResults: boolean;\n  lang: string;\n  onresult: ((e: SpeechRecognitionEvent) => void) | null;\n  onerror: ((e: SpeechRecognitionErrorEvent) => void) | null;\n  onend: (() => void) | null;\n  start: () => void;\n  stop: () => void;\n  abort: () => void;\n}\n\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\n\nfunction getSpeechRecognition(): SpeechRecognitionCtor | null {\n  if (typeof globalThis === 'undefined') return null;\n  const w = globalThis as unknown as Record<string, unknown>;\n  return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor | null;\n}\n\n/**\n * Manages Web Speech API for real-time streaming transcript preview.\n * Provides word-by-word interim text while Whisper handles corrections.\n */\nconst NO_RESULT_TIMEOUT_MS = 5_000;\n\nexport class SpeechStreamingManager {\n  private recognition: SpeechRecognitionInstance | null = null;\n  private accumulated = '';\n  private active = false;\n  private receivedResult = false;\n  private noResultTimer: ReturnType<typeof setTimeout> | null = null;\n  private onTranscript: ((text: string) => void) | null = null;\n  private onPause: (() => void) | null = null;\n  private onError: ((message: string) => void) | null = null;\n\n  /** Check if the Web Speech API is available in this environment. */\n  static isSupported(): boolean {\n    return getSpeechRecognition() !== null;\n  }\n\n  /** Set callback for streaming transcript updates (interim + final text). */\n  setOnTranscript(fn: (text: string) => void): void {\n    this.onTranscript = fn;\n  }\n\n  /** Set callback for speech pause detection (Speech API onend). */\n  setOnPause(fn: () => void): void {\n    this.onPause = fn;\n  }\n\n  /** Set callback for errors. */\n  setOnError(fn: (message: string) => void): void {\n    this.onError = fn;\n  }\n\n  /** Start streaming recognition. No-op if Speech API unavailable. */\n  start(language: string): void {\n    const SR = getSpeechRecognition();\n    if (!SR) return;\n\n    this.accumulated = '';\n    this.active = true;\n    this.receivedResult = false;\n\n    const recognition = new SR();\n    recognition.continuous = true;\n    recognition.interimResults = true;\n    recognition.lang = language;\n\n    let lastFinalIndex = -1;\n    let lastFinalText = '';\n\n    // Detect silent failure: if no onresult fires within timeout, emit error\n    this.clearNoResultTimer();\n    this.noResultTimer = setTimeout(() => {\n      if (this.active && !this.receivedResult) {\n        this.onError?.(\n          'Speech streaming started but received no results. ' +\n            'Mic may be blocked by another audio capture.',\n        );\n      }\n    }, NO_RESULT_TIMEOUT_MS);\n\n    recognition.onresult = (e: SpeechRecognitionEvent) => {\n      if (this.recognition !== recognition) return;\n      this.receivedResult = true;\n      this.clearNoResultTimer();\n\n      let final_ = '';\n      let interim = '';\n      for (let i = e.resultIndex; i < e.results.length; i++) {\n        const t = e.results[i][0].transcript;\n        if (e.results[i].isFinal) {\n          if (i > lastFinalIndex) {\n            final_ += t;\n            lastFinalIndex = i;\n          }\n        } else {\n          interim += t;\n        }\n      }\n\n      if (final_ && final_.trim() !== lastFinalText) {\n        lastFinalText = final_.trim();\n        this.accumulated = this.accumulated\n          ? this.accumulated + ' ' + final_.trim()\n          : final_.trim();\n        this.onTranscript?.(this.accumulated);\n      } else if (interim) {\n        const trimmed = interim.trimStart();\n        const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\n        this.onTranscript?.(full);\n      }\n    };\n\n    recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\n      if (this.recognition !== recognition) return;\n      this.onError?.(e.error);\n    };\n\n    recognition.onend = () => {\n      if (this.recognition !== recognition) return;\n\n      if (this.active) {\n        // Speech API paused — trigger correction\n        this.onPause?.();\n        // Restart for continued streaming\n        try {\n          recognition.start();\n        } catch {\n          this.recognition = null;\n        }\n      } else {\n        this.recognition = null;\n      }\n    };\n\n    this.recognition = recognition;\n    try {\n      recognition.start();\n    } catch {\n      this.recognition = null;\n      this.active = false;\n    }\n  }\n\n  private clearNoResultTimer(): void {\n    if (this.noResultTimer) {\n      clearTimeout(this.noResultTimer);\n      this.noResultTimer = null;\n    }\n  }\n\n  /** Stop streaming recognition and return accumulated text. */\n  stop(): string {\n    this.active = false;\n    this.clearNoResultTimer();\n    if (this.recognition) {\n      const rec = this.recognition;\n      this.recognition = null;\n      rec.stop();\n    }\n    const result = this.accumulated;\n    this.accumulated = '';\n    return result;\n  }\n\n  /** Abort immediately without returning text. */\n  destroy(): void {\n    this.active = false;\n    this.clearNoResultTimer();\n    if (this.recognition) {\n      const rec = this.recognition;\n      this.recognition = null;\n      rec.abort();\n    }\n    this.accumulated = '';\n    this.onTranscript = null;\n    this.onPause = null;\n    this.onError = null;\n  }\n}\n","import type {\n  STTConfig,\n  STTState,\n  STTEvents,\n  STTStatus,\n  ResolvedSTTConfig,\n  AudioCaptureHandle,\n} from './types.js';\nimport { resolveConfig } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\nimport { startCapture, snapshotAudio, resampleAudio, stopCapture } from './audio-capture.js';\nimport { WorkerManager } from './worker-manager.js';\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\nimport { SpeechStreamingManager } from './speech-streaming.js';\n\n/**\n * Main STT engine — the public API for speech-to-text with Whisper correction.\n *\n * Usage:\n * ```typescript\n * const engine = new STTEngine({ model: 'tiny' });\n * engine.on('transcript', (text) => console.log(text));\n * engine.on('correction', (text) => console.log('corrected:', text));\n * await engine.init();\n * await engine.start();\n * const finalText = await engine.stop();\n * ```\n */\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\n  private config: ResolvedSTTConfig;\n  private workerManager: WorkerManager;\n  private correctionOrchestrator: CorrectionOrchestrator;\n  private speechStreaming: SpeechStreamingManager;\n  private capture: AudioCaptureHandle | null = null;\n  private state: STTState;\n  private workerUrl?: URL;\n\n  /**\n   * Create a new STT engine instance.\n   * @param config - Optional configuration overrides (model, backend, language, etc.).\n   * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\n   */\n  constructor(config?: STTConfig, workerUrl?: URL) {\n    super();\n    this.config = resolveConfig(config);\n    this.workerManager = new WorkerManager();\n    this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\n    this.speechStreaming = new SpeechStreamingManager();\n    this.workerUrl = workerUrl;\n\n    this.state = {\n      status: 'idle',\n      isModelLoaded: false,\n      loadProgress: 0,\n      backend: null,\n      error: null,\n    };\n\n    this.correctionOrchestrator.setCorrectionFn(() => {\n      this.performCorrection();\n    });\n\n    this.setupWorkerListeners();\n    this.setupStreamingCallbacks();\n  }\n\n  /** Initialize the engine: spawn worker and load model. */\n  async init(): Promise<void> {\n    this.updateStatus('loading');\n    this.workerManager.spawn(this.workerUrl);\n\n    try {\n      await this.workerManager.loadModel(this.config);\n      this.state.isModelLoaded = true;\n      this.updateStatus('ready');\n    } catch (err) {\n      this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\n      this.updateStatus('idle');\n      throw err;\n    }\n  }\n\n  /** Start recording audio and enable correction cycles. */\n  async start(): Promise<void> {\n    if (this.state.status !== 'ready') {\n      throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\n    }\n\n    try {\n      // Start Speech API BEFORE getUserMedia — avoids mic conflict where\n      // the second mic accessor silently fails (no errors, no events).\n      // This matches the working ClaudeWebCLI order.\n      if (this.config.streaming.enabled) {\n        this.speechStreaming.start(this.config.language);\n      }\n      this.capture = await startCapture();\n      this.updateStatus('recording');\n      this.correctionOrchestrator.start();\n    } catch (err) {\n      this.emitError(\n        'MIC_DENIED',\n        err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\n      );\n    }\n  }\n\n  /** Stop recording, run final transcription, return text. */\n  async stop(): Promise<string> {\n    if (!this.capture) return '';\n\n    this.correctionOrchestrator.stop();\n    this.speechStreaming.stop();\n    this.workerManager.cancel();\n\n    this.updateStatus('processing');\n\n    try {\n      const audio = await stopCapture(this.capture);\n      this.capture = null;\n\n      if (audio.length === 0) {\n        this.updateStatus('ready');\n        return '';\n      }\n\n      const text = await this.workerManager.transcribe(audio);\n      this.emit('correction', text);\n      this.updateStatus('ready');\n      return text;\n    } catch (err) {\n      this.emitError(\n        'TRANSCRIPTION_FAILED',\n        err instanceof Error ? err.message : 'Final transcription failed.',\n      );\n      this.updateStatus('ready');\n      return '';\n    }\n  }\n\n  /** Destroy the engine: terminate worker, release all resources. */\n  destroy(): void {\n    this.correctionOrchestrator.stop();\n    this.speechStreaming.destroy();\n\n    if (this.capture) {\n      for (const track of this.capture.stream.getTracks()) {\n        track.stop();\n      }\n      this.capture.audioCtx.close().catch(() => {});\n      this.capture = null;\n    }\n\n    this.workerManager.destroy();\n    this.updateStatus('idle');\n    this.removeAllListeners();\n  }\n\n  /** Get current engine state. */\n  getState(): Readonly<STTState> {\n    return { ...this.state };\n  }\n\n  /** Notify the correction orchestrator of a speech pause. */\n  notifyPause(): void {\n    this.correctionOrchestrator.onPauseDetected();\n  }\n\n  private async performCorrection(): Promise<void> {\n    if (!this.capture || !this.state.isModelLoaded) return;\n\n    this.workerManager.cancel();\n\n    try {\n      const samples = snapshotAudio(this.capture);\n      const nativeSr = this.capture.audioCtx.sampleRate;\n      const audio = await resampleAudio(samples, nativeSr);\n\n      if (audio.length === 0) return;\n\n      const text = await this.workerManager.transcribe(audio);\n      if (text.trim() && this.capture) {\n        this.emit('correction', text);\n      }\n    } catch (err) {\n      this.emitError(\n        'TRANSCRIPTION_FAILED',\n        err instanceof Error ? err.message : 'Correction transcription failed.',\n      );\n      // Recording continues — error is non-fatal\n    }\n  }\n\n  private setupStreamingCallbacks(): void {\n    this.speechStreaming.setOnTranscript((text) => {\n      this.emit('transcript', text);\n    });\n\n    this.speechStreaming.setOnPause(() => {\n      this.correctionOrchestrator.onPauseDetected();\n    });\n\n    this.speechStreaming.setOnError((message) => {\n      this.emitError('STREAMING_ERROR', message);\n    });\n  }\n\n  private setupWorkerListeners(): void {\n    this.workerManager.on('progress', (percent) => {\n      this.state.loadProgress = percent;\n      this.emit('status', { ...this.state 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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@tekyzinc/stt-component",
-  "version": "0.1.0",
+  "version": "0.2.1",
   "description": "Framework-agnostic speech-to-text with real-time streaming transcription and mid-recording Whisper correction",
   "type": "module",
   "main": "dist/index.cjs",