npm - @tekyzinc/stt-component - Versions diffs - 0.3.2 → 0.3.3 - Mend

@tekyzinc/stt-component 0.3.2 → 0.3.3

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Files changed (5) hide show

package/dist/index.cjs CHANGED Viewed

@@ -152,6 +152,13 @@ async function resumeCapture(capture) {
 function snapshotAudio(capture) {
   return [...capture.samples];
 }
+function trimAudioBuffer(capture, keepSeconds) {
+  const samplesPerChunk = 4096;
+  const chunksToKeep = Math.ceil(keepSeconds * capture.audioCtx.sampleRate / samplesPerChunk);
+  if (capture.samples.length > chunksToKeep) {
+    capture.samples.splice(0, capture.samples.length - chunksToKeep);
+  }
+}
 async function resampleAudio(samples, nativeSr) {
   const totalLength = samples.reduce((sum, s) => sum + s.length, 0);
   if (totalLength === 0) return new Float32Array(0);
@@ -918,6 +925,7 @@ var STTEngine = class extends TypedEventEmitter {
       const text = await this.workerManager.transcribe(audio);
       if (text.trim() && this.capture && !this._stopping) {
         this.emit("correction", text);
+        trimAudioBuffer(this.capture, 30);
       }
     } catch (err) {
       this.emitError(

package/dist/index.cjs.map CHANGED Viewed

	@@ -1 +1 @@
1	- {"version":3,"sources":["../src/index.ts","../src/types.ts","../src/event-emitter.ts","../src/audio-capture.ts","../src/worker-manager.ts","../src/correction-orchestrator.ts","../src/speech-streaming.ts","../src/stt-engine.ts"],"sourcesContent":["// Public types\r\nexport type {\r\n STTModelSize,\r\n STTBackend,\r\n STTStatus,\r\n STTCorrectionProvider,\r\n STTStreamingProvider,\r\n STTCorrectionConfig,\r\n STTStreamingConfig,\r\n STTChunkingConfig,\r\n STTConfig,\r\n ResolvedSTTConfig,\r\n STTState,\r\n STTError,\r\n STTEvents,\r\n AudioCaptureHandle,\r\n} from './types.js';\r\n\r\n// Public values\r\nexport { DEFAULT_STT_CONFIG, resolveConfig } from './types.js';\r\n\r\n// Event emitter\r\nexport { TypedEventEmitter } from './event-emitter.js';\r\n\r\n// Audio capture\r\nexport {\r\n startCapture,\r\n pauseCapture,\r\n resumeCapture,\r\n snapshotAudio,\r\n resampleAudio,\r\n stopCapture,\r\n} from './audio-capture.js';\r\n\r\n// Worker manager\r\nexport { WorkerManager } from './worker-manager.js';\r\nexport type { WorkerManagerEvents } from './worker-manager.js';\r\n\r\n// Correction orchestrator\r\nexport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\n\r\n// Speech streaming\r\nexport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n// STT Engine (main public API)\r\nexport { STTEngine } from './stt-engine.js';\r\n","/** Supported Whisper model sizes. /\r\nexport type STTModelSize = 'tiny' \| 'base' \| 'small' \| 'medium';\r\n\r\n/* Supported compute backends. /\r\nexport type STTBackend = 'webgpu' \| 'wasm' \| 'auto';\r\n\r\n/* Engine lifecycle states. /\r\nexport type STTStatus = 'idle' \| 'loading' \| 'ready' \| 'recording' \| 'processing';\r\n\r\n/* Supported correction engine providers. /\r\nexport type STTCorrectionProvider = 'whisper';\r\n\r\n/* Supported real-time streaming providers. /\r\nexport type STTStreamingProvider = 'web-speech-api';\r\n\r\n/* Correction engine configuration. /\r\nexport interface STTCorrectionConfig {\r\n /* Enable mid-recording correction. Default: true /\r\n enabled?: boolean;\r\n /* Correction engine provider. Default: 'whisper' /\r\n provider?: STTCorrectionProvider;\r\n /* Silence duration (ms) before triggering correction. Default: 1000 /\r\n pauseThreshold?: number;\r\n /* Maximum interval (ms) between forced corrections. Default: 3000 /\r\n forcedInterval?: number;\r\n}\r\n\r\n/* Real-time streaming preview configuration. /\r\nexport interface STTStreamingConfig {\r\n /* Enable real-time streaming transcript. Default: true /\r\n enabled?: boolean;\r\n /* Streaming provider. Default: 'web-speech-api' /\r\n provider?: STTStreamingProvider;\r\n}\r\n\r\n/* Audio chunking configuration for long-form audio. /\r\nexport interface STTChunkingConfig {\r\n /* Chunk length in seconds for Whisper processing. Default: 30 /\r\n chunkLengthS?: number;\r\n /* Stride length in seconds for overlapping chunks. Default: 5 /\r\n strideLengthS?: number;\r\n}\r\n\r\n/* Full engine configuration. All fields optional — sensible defaults applied. /\r\nexport interface STTConfig {\r\n /* Whisper model size. Default: 'tiny' /\r\n model?: STTModelSize;\r\n /* Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) /\r\n backend?: STTBackend;\r\n /* Transcription language. Default: 'en' /\r\n language?: string;\r\n /* Model quantization dtype. Default: 'q4' /\r\n dtype?: string;\r\n /* Mid-recording correction settings. /\r\n correction?: STTCorrectionConfig;\r\n /* Audio chunking settings for long-form audio. /\r\n chunking?: STTChunkingConfig;\r\n /* Web Speech API streaming preview settings. /\r\n streaming?: STTStreamingConfig;\r\n}\r\n\r\n/* Resolved configuration with all defaults applied. /\r\nexport interface ResolvedSTTConfig {\r\n model: STTModelSize;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n correction: Required<STTCorrectionConfig>;\r\n chunking: Required<STTChunkingConfig>;\r\n streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/* Engine state exposed to consumers via status events. /\r\nexport interface STTState {\r\n status: STTStatus;\r\n isModelLoaded: boolean;\r\n /* Model download progress (0–100). /\r\n loadProgress: number;\r\n /* Active compute backend, or null if not yet determined. /\r\n backend: 'webgpu' \| 'wasm' \| null;\r\n error: string \| null;\r\n}\r\n\r\n/* Structured error emitted via the 'error' event. /\r\nexport interface STTError {\r\n code: string;\r\n message: string;\r\n}\r\n\r\n/* Event map for the typed event emitter. /\r\nexport type STTEvents = {\r\n /* Streaming interim text during recording. /\r\n transcript: (text: string) => void;\r\n /* Whisper-corrected text replacing interim text. /\r\n correction: (text: string) => void;\r\n /* Actionable error (mic denied, model fail, transcription fail). /\r\n error: (error: STTError) => void;\r\n /* Engine state change. /\r\n status: (state: STTState) => void;\r\n /* Diagnostic log for debugging (subscribe to capture all internal events). /\r\n debug: (message: string) => void;\r\n};\r\n\r\n/* Handle returned by audio capture — used internally. /\r\nexport interface AudioCaptureHandle {\r\n audioCtx: AudioContext;\r\n stream: MediaStream;\r\n samples: Float32Array[];\r\n /* Retain reference to prevent GC from stopping audio processing. /\r\n _processor: ScriptProcessorNode;\r\n /* Source node for disconnect/reconnect on pause/resume. /\r\n _source: MediaStreamAudioSourceNode;\r\n /* Gain node (silent) to prevent mic playback. /\r\n _silencer: GainNode;\r\n}\r\n\r\n/* Message sent from main thread to Whisper worker. /\r\nexport interface WorkerMessage {\r\n type: 'load' \| 'transcribe' \| 'cancel';\r\n audio?: Float32Array;\r\n config?: {\r\n model: string;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n chunkLengthS: number;\r\n strideLengthS: number;\r\n };\r\n}\r\n\r\n/* Response sent from Whisper worker to main thread. /\r\nexport interface WorkerResponse {\r\n type: 'progress' \| 'ready' \| 'result' \| 'error';\r\n data?: unknown;\r\n}\r\n\r\n/* Default configuration values. /\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n model: 'tiny',\r\n backend: 'auto',\r\n language: 'en',\r\n dtype: 'q4',\r\n correction: {\r\n enabled: true,\r\n provider: 'whisper',\r\n pauseThreshold: 1_000,\r\n forcedInterval: 3_000,\r\n },\r\n chunking: {\r\n chunkLengthS: 30,\r\n strideLengthS: 5,\r\n },\r\n streaming: {\r\n enabled: true,\r\n provider: 'web-speech-api',\r\n },\r\n};\r\n\r\n/* Merge user config with defaults to produce resolved config. /\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n return {\r\n model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n correction: {\r\n enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n pauseThreshold:\r\n config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n forcedInterval:\r\n config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n },\r\n chunking: {\r\n chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n },\r\n streaming: {\r\n enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n },\r\n };\r\n}\r\n","/\n A generic, typed event emitter.\n \n Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n /\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n /* Subscribe to an event. /\n on<K extends keyof T>(event: K, listener: T[K]): void {\n let set = this.listeners.get(event);\n if (!set) {\n set = new Set();\n this.listeners.set(event, set);\n }\n set.add(listener as T[keyof T]);\n }\n\n /* Unsubscribe a specific listener. No-op if not registered. /\n off<K extends keyof T>(event: K, listener: T[K]): void {\n this.listeners.get(event)?.delete(listener as T[keyof T]);\n }\n\n /* Emit an event, calling all registered listeners in insertion order. /\n emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n const set = this.listeners.get(event);\n if (!set) return;\n for (const listener of set) {\n (listener as (...a: Parameters<T[K]>) => void)(...args);\n }\n }\n\n /* Remove all listeners, optionally for a single event. /\n removeAllListeners(event?: keyof T): void {\n if (event !== undefined) {\n this.listeners.delete(event);\n } else {\n this.listeners.clear();\n }\n }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/\r\n Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n /\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n const stream = await navigator.mediaDevices.getUserMedia({\r\n audio: { channelCount: 1 },\r\n });\r\n const audioCtx = new AudioContext();\r\n\r\n // Chrome may suspend AudioContext — must resume within user gesture\r\n if (audioCtx.state === 'suspended') {\r\n await audioCtx.resume();\r\n }\r\n\r\n const source = audioCtx.createMediaStreamSource(stream);\r\n const samples: Float32Array[] = [];\r\n\r\n const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n };\r\n\r\n // Connect through a silent gain node so mic audio doesn't play back\r\n const silencer = audioCtx.createGain();\r\n silencer.gain.value = 0;\r\n source.connect(processor);\r\n processor.connect(silencer);\r\n silencer.connect(audioCtx.destination);\r\n\r\n return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };\r\n}\r\n\r\n/\r\n Pause capture without releasing mic or AudioContext.\r\n * Disconnects the audio source so no new samples are collected.\r\n * Returns resampled audio from the recording period.\r\n * Call resumeCapture() to start collecting again.\r\n /\r\nexport async function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n capture._source.disconnect();\r\n const currentSamples = [...capture.samples];\r\n capture.samples.length = 0;\r\n return resampleAudio(currentSamples, capture.audioCtx.sampleRate);\r\n}\r\n\r\n/\r\n Resume a paused capture. Reconnects the audio source to the processor.\r\n * AudioContext is resumed if suspended.\r\n /\r\nexport async function resumeCapture(capture: AudioCaptureHandle): Promise<void> {\r\n if (capture.audioCtx.state === 'suspended') {\r\n await capture.audioCtx.resume();\r\n }\r\n capture._source.connect(capture._processor);\r\n}\r\n\r\n/\r\n Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n /\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n return [...capture.samples];\r\n}\r\n\r\n/\r\n Concatenate sample chunks and resample to 16kHz for Whisper.\r\n /\r\nexport async function resampleAudio(\r\n samples: Float32Array[],\r\n nativeSr: number,\r\n): Promise<Float32Array> {\r\n const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n if (totalLength === 0) return new Float32Array(0);\r\n\r\n const fullAudio = new Float32Array(totalLength);\r\n let offset = 0;\r\n for (const s of samples) {\r\n fullAudio.set(s, offset);\r\n offset += s.length;\r\n }\r\n\r\n if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n const duration = fullAudio.length / nativeSr;\r\n const outLength = Math.round(duration TARGET_SAMPLE_RATE);\r\n const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n buffer.getChannelData(0).set(fullAudio);\r\n const src = offline.createBufferSource();\r\n src.buffer = buffer;\r\n src.connect(offline.destination);\r\n src.start(0);\r\n const resampled = await offline.startRendering();\r\n return resampled.getChannelData(0);\r\n}\r\n\r\n/*\r\n Stop capturing and return resampled audio at 16kHz.\r\n /\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n // Disconnect processor to stop capturing\r\n try {\r\n _processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n\r\n // Stop microphone tracks\r\n for (const track of stream.getTracks()) {\r\n track.stop();\r\n }\r\n\r\n const nativeSr = audioCtx.sampleRate;\r\n await audioCtx.close();\r\n\r\n return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/* Events emitted by the WorkerManager. /\nexport type WorkerManagerEvents = {\n progress: (percent: number) => void;\n ready: () => void;\n result: (text: string) => void;\n error: (message: string) => void;\n};\n\n/\n Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n /\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n private worker: Worker \| null = null;\n private transcribeResolve: ((text: string) => void) \| null = null;\n private currentTranscribePromise: Promise<string> \| null = null;\n private modelReadyResolve: (() => void) \| null = null;\n private modelReadyReject: ((err: Error) => void) \| null = null;\n\n /* True while a transcription job is running in the worker. /\n get isTranscribing(): boolean {\n return this.transcribeResolve !== null;\n }\n\n /* Await the current in-flight transcription without starting a new one. /\n awaitCurrentTranscription(): Promise<string> {\n return this.currentTranscribePromise ?? Promise.resolve('');\n }\n\n /* Spawn the Web Worker. Must be called before loadModel/transcribe. /\n spawn(workerUrl?: URL): void {\n if (this.worker) return;\n\n const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n this.worker = new Worker(url, { type: 'module' });\n this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n this.handleMessage(e.data);\n };\n this.worker.onerror = (e: ErrorEvent) => {\n this.emit('error', e.message ?? 'Worker error');\n };\n }\n\n /* Load the Whisper model in the worker. Resolves when ready. /\n async loadModel(config: ResolvedSTTConfig): Promise<void> {\n if (!this.worker) throw new Error('Worker not spawned');\n\n return new Promise<void>((resolve, reject) => {\n this.modelReadyResolve = resolve;\n this.modelReadyReject = reject;\n this.worker!.postMessage({\n type: 'load',\n config: {\n model: config.model,\n backend: config.backend,\n language: config.language,\n dtype: config.dtype,\n chunkLengthS: config.chunking.chunkLengthS,\n strideLengthS: config.chunking.strideLengthS,\n },\n });\n });\n }\n\n /* Send audio to the worker for transcription. Resolves with text. /\n async transcribe(audio: Float32Array): Promise<string> {\n if (!this.worker) throw new Error('Worker not spawned');\n if (audio.length === 0) return '';\n\n this.currentTranscribePromise = new Promise<string>((resolve) => {\n this.transcribeResolve = resolve;\n this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n });\n return this.currentTranscribePromise;\n }\n\n /* Cancel any in-flight transcription. /\n cancel(): void {\n this.worker?.postMessage({ type: 'cancel' });\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n }\n\n /* Terminate the worker and release resources. /\n destroy(): void {\n this.cancel();\n this.worker?.terminate();\n this.worker = null;\n this.removeAllListeners();\n }\n\n private handleMessage(msg: WorkerResponse): void {\n switch (msg.type) {\n case 'progress':\n this.emit('progress', msg.data as number);\n break;\n case 'ready':\n this.emit('ready');\n this.modelReadyResolve?.();\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n break;\n case 'result':\n this.emit('result', msg.data as string);\n this.transcribeResolve?.(msg.data as string);\n this.transcribeResolve = null;\n break;\n case 'error': {\n const errMsg = msg.data as string;\n this.emit('error', errMsg);\n // Reject model load if still pending\n if (this.modelReadyReject) {\n this.modelReadyReject(new Error(errMsg));\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n }\n // Resolve transcribe with empty string on error\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n break;\n }\n }\n }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/\n Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n /\n/* Delay (ms) before the first correction fires after recording starts. /\nconst INITIAL_CORRECTION_DELAY_MS = 1_000;\n\nexport class CorrectionOrchestrator {\n private forcedTimer: ReturnType<typeof setInterval> \| null = null;\n private initialTimer: ReturnType<typeof setTimeout> \| 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.\n * Fires a quick initial correction after 1s for early feedback, then\n * switches to the regular forcedInterval cadence from that point. /\n start(): void {\n if (!this.config.enabled) return;\n\n this.lastCorrectionTime = Date.now();\n // One-shot early correction — gives the user immediate Whisper feedback\n // before the Web Speech API has produced its first interim result (~1-2s).\n this.initialTimer = setTimeout(() => {\n this.initialTimer = null;\n this.correctionFn?.();\n this.lastCorrectionTime = Date.now();\n this.startForcedTimer();\n }, INITIAL_CORRECTION_DELAY_MS);\n }\n\n /* Stop the orchestrator (clear all timers). /\n stop(): void {\n if (this.initialTimer) {\n clearTimeout(this.initialTimer);\n this.initialTimer = null;\n }\n this.stopForcedTimer();\n }\n\n /* Called when a speech pause is detected. Triggers correction if cooldown elapsed. /\n onPauseDetected(): void {\n if (!this.config.enabled) return;\n\n const now = Date.now();\n if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n this.triggerCorrection();\n }\n\n /* Force a correction now (resets timer). /\n forceCorrection(): void {\n this.triggerCorrection();\n }\n\n private triggerCorrection(): void {\n this.lastCorrectionTime = Date.now();\n this.correctionFn?.();\n // Reset forced timer after any correction\n this.restartForcedTimer();\n }\n\n private startForcedTimer(): void {\n this.stopForcedTimer();\n this.forcedTimer = setInterval(() => {\n this.triggerCorrection();\n }, this.config.forcedInterval);\n }\n\n private stopForcedTimer(): void {\n if (this.forcedTimer) {\n clearInterval(this.forcedTimer);\n this.forcedTimer = null;\n }\n }\n\n private restartForcedTimer(): void {\n if (this.forcedTimer) {\n this.startForcedTimer();\n }\n }\n}\n","/ ─── Web Speech API types ──────────────────────────────── /\r\n\r\ninterface SpeechRecognitionEvent {\r\n results: SpeechRecognitionResultList;\r\n resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n continuous: boolean;\r\n interimResults: boolean;\r\n lang: string;\r\n onaudiostart: (() => void) \| null;\r\n onresult: ((e: SpeechRecognitionEvent) => void) \| null;\r\n onerror: ((e: SpeechRecognitionErrorEvent) => void) \| null;\r\n onend: (() => void) \| null;\r\n start: () => void;\r\n stop: () => void;\r\n abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor \| null {\r\n if (typeof globalThis === 'undefined') return null;\r\n const w = globalThis as unknown as Record<string, unknown>;\r\n return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor \| null;\r\n}\r\n\r\n/ ─── Language mapping ──────────────────────────────────── /\r\n\r\n/* Map Whisper language codes to BCP-47 locale tags for the Speech API. /\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n en: 'en-US',\r\n english: 'en-US',\r\n zh: 'zh-CN',\r\n chinese: 'zh-CN',\r\n de: 'de-DE',\r\n german: 'de-DE',\r\n es: 'es-ES',\r\n spanish: 'es-ES',\r\n ru: 'ru-RU',\r\n russian: 'ru-RU',\r\n ko: 'ko-KR',\r\n korean: 'ko-KR',\r\n fr: 'fr-FR',\r\n french: 'fr-FR',\r\n ja: 'ja-JP',\r\n japanese: 'ja-JP',\r\n pt: 'pt-BR',\r\n portuguese: 'pt-BR',\r\n tr: 'tr-TR',\r\n turkish: 'tr-TR',\r\n pl: 'pl-PL',\r\n polish: 'pl-PL',\r\n nl: 'nl-NL',\r\n dutch: 'nl-NL',\r\n ar: 'ar-SA',\r\n arabic: 'ar-SA',\r\n sv: 'sv-SE',\r\n swedish: 'sv-SE',\r\n it: 'it-IT',\r\n italian: 'it-IT',\r\n id: 'id-ID',\r\n indonesian: 'id-ID',\r\n hi: 'hi-IN',\r\n hindi: 'hi-IN',\r\n fi: 'fi-FI',\r\n finnish: 'fi-FI',\r\n vi: 'vi-VN',\r\n vietnamese: 'vi-VN',\r\n he: 'he-IL',\r\n hebrew: 'he-IL',\r\n uk: 'uk-UA',\r\n ukrainian: 'uk-UA',\r\n el: 'el-GR',\r\n greek: 'el-GR',\r\n ms: 'ms-MY',\r\n malay: 'ms-MY',\r\n cs: 'cs-CZ',\r\n czech: 'cs-CZ',\r\n ro: 'ro-RO',\r\n romanian: 'ro-RO',\r\n da: 'da-DK',\r\n danish: 'da-DK',\r\n hu: 'hu-HU',\r\n hungarian: 'hu-HU',\r\n no: 'nb-NO',\r\n norwegian: 'nb-NO',\r\n th: 'th-TH',\r\n thai: 'th-TH',\r\n};\r\n\r\n/\r\n Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n /\r\nfunction toBCP47(language: string): string {\r\n if (language.includes('-')) return language;\r\n return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/ ─── SpeechStreamingManager ────────────────────────────── /\r\n\r\n/\r\n Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n \r\n Keep-alive design: the recognition session is never stopped between user\r\n * sessions (only muted). This eliminates the 1-2s Google server handshake\r\n * on every click — start() resolves instantly when the session is warm.\r\n /\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n/* How long to keep recognition running silently after stop() before truly releasing it. /\r\nconst IDLE_TIMEOUT_MS = 30_000;\r\n\r\nexport class SpeechStreamingManager {\r\n private recognition: SpeechRecognitionInstance \| null = null;\r\n private accumulated = '';\r\n private active = false; // user is recording — emit results, trigger onPause on end\r\n private keepingWarm = false; // recognition running silently between user sessions\r\n private currentLang = ''; // BCP-47 of running recognition (for fast-path check)\r\n private receivedResult = false;\r\n private lastFinalIndex = -1; // instance fields so warm-restart resets them cleanly\r\n private lastFinalText = '';\r\n private noResultTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private idleTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private onTranscript: ((text: string) => void) \| null = null;\r\n private onPause: (() => void) \| null = null;\r\n private onError: ((message: string) => void) \| null = null;\r\n private onDebug: ((message: string) => void) \| null = null;\r\n\r\n /* Check if the Web Speech API is available in this environment. /\r\n static isSupported(): boolean {\r\n return getSpeechRecognition() !== null;\r\n }\r\n\r\n /* Set callback for streaming transcript updates (interim + final text). /\r\n setOnTranscript(fn: (text: string) => void): void {\r\n this.onTranscript = fn;\r\n }\r\n\r\n /* Set callback for speech pause detection (Speech API onend). /\r\n setOnPause(fn: () => void): void {\r\n this.onPause = fn;\r\n }\r\n\r\n /* Set callback for errors. /\r\n setOnError(fn: (message: string) => void): void {\r\n this.onError = fn;\r\n }\r\n\r\n /* Set callback for diagnostic debug messages. /\r\n setOnDebug(fn: (message: string) => void): void {\r\n this.onDebug = fn;\r\n }\r\n\r\n private log(message: string): void {\r\n this.onDebug?.(message);\r\n console.warn(message);\r\n }\r\n\r\n /\r\n Pre-warm: start recognition in muted mode so it's ready before the user\r\n * clicks. Call after engine.init() completes. Eliminates startup latency on\r\n * first click by keeping the Google Speech session alive.\r\n /\r\n preWarm(language: string): void {\r\n const SR = getSpeechRecognition();\r\n if (!SR) return;\r\n const bcp47 = toBCP47(language);\r\n if (this.recognition && this.currentLang === bcp47) return; // already warm\r\n this.log(`[SSM] preWarm() — lang: \"${language}\" → \"${bcp47}\"`);\r\n this.keepingWarm = true;\r\n this.active = false;\r\n this.clearIdleTimer();\r\n this.spawnRecognition(language);\r\n }\r\n\r\n /\r\n Start streaming recognition. If recognition is already warm (session\r\n * running from preWarm or a previous session within the idle window),\r\n * activates instantly — no Google handshake. Otherwise cold-starts.\r\n /\r\n start(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition();\r\n if (!SR) {\r\n this.log('[SSM] SpeechRecognition not available in this environment');\r\n return Promise.resolve();\r\n }\r\n\r\n const bcp47 = toBCP47(language);\r\n this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n this.accumulated = '';\r\n this.receivedResult = false;\r\n this.lastFinalIndex = -1;\r\n this.lastFinalText = '';\r\n this.clearIdleTimer();\r\n\r\n // ── Fast path: session already running with the right language ──────────\r\n if (this.recognition && this.currentLang === bcp47) {\r\n this.log('[SSM] start() — warm session, activating immediately');\r\n this.keepingWarm = false;\r\n this.active = true;\r\n // Arm the no-result watchdog for silent-failure detection\r\n this.clearNoResultTimer();\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n return Promise.resolve();\r\n }\r\n\r\n // ── Cold start: spin up a fresh recognition session ─────────────────────\r\n this.keepingWarm = false;\r\n this.active = true;\r\n return this.spawnRecognition(language, skipMicWait);\r\n }\r\n\r\n /\r\n Create and start a new SpeechRecognition instance.\r\n * Used by both preWarm() (active=false) and start() cold path (active=true).\r\n /\r\n private spawnRecognition(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition()!;\r\n const bcp47 = toBCP47(language);\r\n\r\n // Tear down any stale session first\r\n if (this.recognition) {\r\n const old = this.recognition;\r\n this.recognition = null;\r\n try { old.stop(); } catch { / already stopped / }\r\n }\r\n\r\n this.currentLang = bcp47;\r\n\r\n const recognition = new SR();\r\n recognition.continuous = true;\r\n recognition.interimResults = true;\r\n recognition.lang = bcp47;\r\n\r\n // Mic-claim promise (used on cold active starts only)\r\n let micReady = false;\r\n const micClaimPromise = new Promise<void>((resolve) => {\r\n recognition.onaudiostart = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n if (!micReady) { micReady = true; resolve(); }\r\n };\r\n setTimeout(() => {\r\n if (!micReady) {\r\n micReady = true;\r\n this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n resolve();\r\n }\r\n }, 300);\r\n });\r\n\r\n // No-result watchdog — only when actively recording\r\n this.clearNoResultTimer();\r\n if (this.active) {\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n }\r\n\r\n recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.receivedResult = true;\r\n this.clearNoResultTimer();\r\n\r\n // Discard results while muted (keeping warm between user sessions)\r\n if (!this.active) return;\r\n\r\n let final_ = '';\r\n let interim = '';\r\n for (let i = e.resultIndex; i < e.results.length; i++) {\r\n const t = e.results[i][0].transcript;\r\n if (e.results[i].isFinal) {\r\n if (i > this.lastFinalIndex) {\r\n final_ += t;\r\n this.lastFinalIndex = i;\r\n }\r\n } else {\r\n interim += t;\r\n }\r\n }\r\n\r\n this.log(\r\n `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n );\r\n\r\n if (final_ && final_.trim() !== this.lastFinalText) {\r\n this.lastFinalText = final_.trim();\r\n this.accumulated = this.accumulated\r\n ? this.accumulated + ' ' + final_.trim()\r\n : final_.trim();\r\n this.onTranscript?.(this.accumulated);\r\n } else if (interim) {\r\n const trimmed = interim.trimStart();\r\n const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n this.onTranscript?.(full);\r\n }\r\n };\r\n\r\n recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.log(`[SSM] onerror — ${e.error}`);\r\n // Suppress errors while muted — they're not relevant to the user\r\n if (this.active) this.onError?.(e.error);\r\n };\r\n\r\n recognition.onend = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log(\r\n `[SSM] onend — active: ${this.active}, keepingWarm: ${this.keepingWarm}, receivedResult: ${this.receivedResult}`,\r\n );\r\n\r\n if (this.active) {\r\n // User is recording — trigger correction and restart for continued streaming\r\n this.onPause?.();\r\n try {\r\n recognition.start();\r\n this.log('[SSM] restarted after pause');\r\n } catch (err) {\r\n this.log(`[SSM] restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n this.onError?.('Speech recognition failed to restart after pause.');\r\n }\r\n } else if (this.keepingWarm) {\r\n // Between user sessions — restart silently to keep the session alive\r\n try {\r\n recognition.start();\r\n this.log('[SSM] warm restart');\r\n } catch (err) {\r\n this.log(`[SSM] warm restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.keepingWarm = false;\r\n this.currentLang = '';\r\n }\r\n } else {\r\n this.recognition = null;\r\n this.currentLang = '';\r\n }\r\n };\r\n\r\n this.recognition = recognition;\r\n try {\r\n recognition.start();\r\n this.log('[SSM] recognition.start() succeeded');\r\n } catch (err) {\r\n this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n const wasActive = this.active;\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n if (wasActive) {\r\n this.onError?.(\r\n `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n );\r\n }\r\n return Promise.resolve();\r\n }\r\n\r\n // preWarm path or skipMicWait — no need to wait for mic claim\r\n if (!this.active \|\| skipMicWait) {\r\n if (skipMicWait) this.log('[SSM] skipMicWait — warm restart, returning immediately');\r\n return Promise.resolve();\r\n }\r\n\r\n return micClaimPromise;\r\n }\r\n\r\n private clearNoResultTimer(): void {\r\n if (this.noResultTimer) {\r\n clearTimeout(this.noResultTimer);\r\n this.noResultTimer = null;\r\n }\r\n }\r\n\r\n private clearIdleTimer(): void {\r\n if (this.idleTimer) {\r\n clearTimeout(this.idleTimer);\r\n this.idleTimer = null;\r\n }\r\n }\r\n\r\n private startIdleTimer(): void {\r\n this.clearIdleTimer();\r\n this.idleTimer = setTimeout(() => {\r\n this.idleTimer = null;\r\n this.keepingWarm = false;\r\n this.log('[SSM] idle timeout — stopping recognition');\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n try { rec.stop(); } catch { / already stopped / }\r\n }\r\n }, IDLE_TIMEOUT_MS);\r\n }\r\n\r\n /* Stop streaming recognition and return accumulated text.\r\n * Keeps the recognition session alive (muted) for instant restart. /\r\n stop(): string {\r\n this.active = false;\r\n this.keepingWarm = true;\r\n this.clearNoResultTimer();\r\n const result = this.accumulated;\r\n this.accumulated = '';\r\n // Keep recognition running silently; release after idle timeout\r\n this.startIdleTimer();\r\n return result;\r\n }\r\n\r\n /* Abort immediately and release all resources. /\r\n destroy(): void {\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n this.clearIdleTimer();\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n rec.abort();\r\n }\r\n this.accumulated = '';\r\n this.onTranscript = null;\r\n this.onPause = null;\r\n this.onError = null;\r\n this.onDebug = null;\r\n }\r\n}\r\n","import type {\r\n STTConfig,\r\n STTState,\r\n STTEvents,\r\n STTStatus,\r\n ResolvedSTTConfig,\r\n AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, pauseCapture, resumeCapture, snapshotAudio, resampleAudio } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/\r\n Main STT engine — the public API for speech-to-text with Whisper correction.\r\n \r\n Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n /\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n private config: ResolvedSTTConfig;\r\n private workerManager: WorkerManager;\r\n private correctionOrchestrator: CorrectionOrchestrator;\r\n private speechStreaming: SpeechStreamingManager;\r\n private capture: AudioCaptureHandle \| null = null;\r\n private state: STTState;\r\n private workerUrl?: URL;\r\n /* Prevents performCorrection from emitting while stop() is consuming the in-flight result. /\r\n private _stopping = false;\r\n\r\n /\r\n Create a new STT engine instance.\r\n * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n /\r\n constructor(config?: STTConfig, workerUrl?: URL) {\r\n super();\r\n this.config = resolveConfig(config);\r\n this.workerManager = new WorkerManager();\r\n this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n this.speechStreaming = new SpeechStreamingManager();\r\n this.workerUrl = workerUrl;\r\n\r\n this.state = {\r\n status: 'idle',\r\n isModelLoaded: false,\r\n loadProgress: 0,\r\n backend: null,\r\n error: null,\r\n };\r\n\r\n this.correctionOrchestrator.setCorrectionFn(() => {\r\n this.performCorrection();\r\n });\r\n\r\n this.setupWorkerListeners();\r\n this.setupStreamingCallbacks();\r\n }\r\n\r\n /* Initialize the engine: spawn worker and load model. /\r\n async init(): Promise<void> {\r\n this.updateStatus('loading');\r\n this.workerManager.spawn(this.workerUrl);\r\n\r\n try {\r\n await this.workerManager.loadModel(this.config);\r\n this.state.isModelLoaded = true;\r\n this.updateStatus('ready');\r\n // Pre-warm recognition so the first click is instant (no Google handshake delay)\r\n if (this.config.streaming.enabled) {\r\n this.speechStreaming.preWarm(this.config.language);\r\n }\r\n } catch (err) {\r\n this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n this.updateStatus('idle');\r\n throw err;\r\n }\r\n }\r\n\r\n /* Start recording audio and enable correction cycles. /\r\n async start(): Promise<void> {\r\n if (this.state.status !== 'ready') {\r\n throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n }\r\n\r\n try {\r\n // Check if mic is already warm from a previous recording session.\r\n // When warm, skip getUserMedia and the 300ms SR mic-claim wait.\r\n const warmCapture =\r\n this.capture &&\r\n this.capture.stream.getTracks().every((t) => t.readyState === 'live');\r\n\r\n this.emitDebug(\r\n `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\", warm: ${!!warmCapture}`,\r\n );\r\n\r\n if (this.config.streaming.enabled) {\r\n // On warm restart, skip the mic-claim wait — no getUserMedia race.\r\n await this.speechStreaming.start(this.config.language, !!warmCapture);\r\n if (!warmCapture) {\r\n this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n }\r\n }\r\n\r\n if (warmCapture) {\r\n await resumeCapture(this.capture!);\r\n this.emitDebug('[STT] warm mic resumed — skipped getUserMedia');\r\n } else {\r\n // First start or stale capture: full mic init with getUserMedia.\r\n this.capture = await startCapture();\r\n }\r\n\r\n this.updateStatus('recording');\r\n this.correctionOrchestrator.start();\r\n } catch (err) {\r\n this.emitError(\r\n 'MIC_DENIED',\r\n err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n );\r\n }\r\n }\r\n\r\n /* Stop recording, run final transcription, return text.\r\n * Mic and AudioContext stay alive for fast restart — call destroy() to fully release. /\r\n async stop(): Promise<string> {\r\n if (!this.capture) return '';\r\n\r\n // Prevent any in-flight performCorrection from emitting — stop() will own the final emit.\r\n this._stopping = true;\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.stop();\r\n\r\n this.updateStatus('processing');\r\n\r\n // If a correction job is already running, reuse it: Whisper can't be interrupted mid-inference\r\n // anyway, so cancelling just wastes the work and causes a second full run.\r\n if (this.workerManager.isTranscribing) {\r\n try {\r\n // Resample audio and await the in-flight result in parallel.\r\n const [audio, inFlightText] = await Promise.all([\r\n pauseCapture(this.capture),\r\n this.workerManager.awaitCurrentTranscription(),\r\n ]);\r\n this._stopping = false;\r\n\r\n const text = inFlightText.trim();\r\n if (text) {\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n }\r\n\r\n // In-flight returned empty (cancelled or error) — fall through to fresh transcription.\r\n if (audio.length > 0) {\r\n const freshText = await this.workerManager.transcribe(audio);\r\n this.emit('correction', freshText);\r\n this.updateStatus('ready');\r\n return freshText;\r\n }\r\n\r\n this.updateStatus('ready');\r\n return '';\r\n } catch (err) {\r\n this._stopping = false;\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n // No job in-flight — cancel is a no-op, run fresh transcription.\r\n this.workerManager.cancel();\r\n this._stopping = false;\r\n\r\n try {\r\n // Soft pause — keeps stream and AudioContext alive for fast restart.\r\n const audio = await pauseCapture(this.capture);\r\n\r\n if (audio.length === 0) {\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n /* Destroy the engine: terminate worker, release all resources. /\r\n destroy(): void {\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.destroy();\r\n\r\n if (this.capture) {\r\n try {\r\n this.capture._processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n for (const track of this.capture.stream.getTracks()) {\r\n track.stop();\r\n }\r\n this.capture.audioCtx.close().catch(() => {});\r\n this.capture = null;\r\n }\r\n\r\n this.workerManager.destroy();\r\n 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1	+ {"version":3,"sources":["../src/index.ts","../src/types.ts","../src/event-emitter.ts","../src/audio-capture.ts","../src/worker-manager.ts","../src/correction-orchestrator.ts","../src/speech-streaming.ts","../src/stt-engine.ts"],"sourcesContent":["// Public types\r\nexport type {\r\n STTModelSize,\r\n STTBackend,\r\n STTStatus,\r\n STTCorrectionProvider,\r\n STTStreamingProvider,\r\n STTCorrectionConfig,\r\n STTStreamingConfig,\r\n STTChunkingConfig,\r\n STTConfig,\r\n ResolvedSTTConfig,\r\n STTState,\r\n STTError,\r\n STTEvents,\r\n AudioCaptureHandle,\r\n} from './types.js';\r\n\r\n// Public values\r\nexport { DEFAULT_STT_CONFIG, resolveConfig } from './types.js';\r\n\r\n// Event emitter\r\nexport { TypedEventEmitter } from './event-emitter.js';\r\n\r\n// Audio capture\r\nexport {\r\n startCapture,\r\n pauseCapture,\r\n resumeCapture,\r\n snapshotAudio,\r\n resampleAudio,\r\n stopCapture,\r\n} from './audio-capture.js';\r\n\r\n// Worker manager\r\nexport { WorkerManager } from './worker-manager.js';\r\nexport type { WorkerManagerEvents } from './worker-manager.js';\r\n\r\n// Correction orchestrator\r\nexport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\n\r\n// Speech streaming\r\nexport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n// STT Engine (main public API)\r\nexport { STTEngine } from './stt-engine.js';\r\n","/** Supported Whisper model sizes. /\r\nexport type STTModelSize = 'tiny' \| 'base' \| 'small' \| 'medium';\r\n\r\n/* Supported compute backends. /\r\nexport type STTBackend = 'webgpu' \| 'wasm' \| 'auto';\r\n\r\n/* Engine lifecycle states. /\r\nexport type STTStatus = 'idle' \| 'loading' \| 'ready' \| 'recording' \| 'processing';\r\n\r\n/* Supported correction engine providers. /\r\nexport type STTCorrectionProvider = 'whisper';\r\n\r\n/* Supported real-time streaming providers. /\r\nexport type STTStreamingProvider = 'web-speech-api';\r\n\r\n/* Correction engine configuration. /\r\nexport interface STTCorrectionConfig {\r\n /* Enable mid-recording correction. Default: true /\r\n enabled?: boolean;\r\n /* Correction engine provider. Default: 'whisper' /\r\n provider?: STTCorrectionProvider;\r\n /* Silence duration (ms) before triggering correction. Default: 1000 /\r\n pauseThreshold?: number;\r\n /* Maximum interval (ms) between forced corrections. Default: 3000 /\r\n forcedInterval?: number;\r\n}\r\n\r\n/* Real-time streaming preview configuration. /\r\nexport interface STTStreamingConfig {\r\n /* Enable real-time streaming transcript. Default: true /\r\n enabled?: boolean;\r\n /* Streaming provider. Default: 'web-speech-api' /\r\n provider?: STTStreamingProvider;\r\n}\r\n\r\n/* Audio chunking configuration for long-form audio. /\r\nexport interface STTChunkingConfig {\r\n /* Chunk length in seconds for Whisper processing. Default: 30 /\r\n chunkLengthS?: number;\r\n /* Stride length in seconds for overlapping chunks. Default: 5 /\r\n strideLengthS?: number;\r\n}\r\n\r\n/* Full engine configuration. All fields optional — sensible defaults applied. /\r\nexport interface STTConfig {\r\n /* Whisper model size. Default: 'tiny' /\r\n model?: STTModelSize;\r\n /* Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) /\r\n backend?: STTBackend;\r\n /* Transcription language. Default: 'en' /\r\n language?: string;\r\n /* Model quantization dtype. Default: 'q4' /\r\n dtype?: string;\r\n /* Mid-recording correction settings. /\r\n correction?: STTCorrectionConfig;\r\n /* Audio chunking settings for long-form audio. /\r\n chunking?: STTChunkingConfig;\r\n /* Web Speech API streaming preview settings. /\r\n streaming?: STTStreamingConfig;\r\n}\r\n\r\n/* Resolved configuration with all defaults applied. /\r\nexport interface ResolvedSTTConfig {\r\n model: STTModelSize;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n correction: Required<STTCorrectionConfig>;\r\n chunking: Required<STTChunkingConfig>;\r\n streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/* Engine state exposed to consumers via status events. /\r\nexport interface STTState {\r\n status: STTStatus;\r\n isModelLoaded: boolean;\r\n /* Model download progress (0–100). /\r\n loadProgress: number;\r\n /* Active compute backend, or null if not yet determined. /\r\n backend: 'webgpu' \| 'wasm' \| null;\r\n error: string \| null;\r\n}\r\n\r\n/* Structured error emitted via the 'error' event. /\r\nexport interface STTError {\r\n code: string;\r\n message: string;\r\n}\r\n\r\n/* Event map for the typed event emitter. /\r\nexport type STTEvents = {\r\n /* Streaming interim text during recording. /\r\n transcript: (text: string) => void;\r\n /* Whisper-corrected text replacing interim text. /\r\n correction: (text: string) => void;\r\n /* Actionable error (mic denied, model fail, transcription fail). /\r\n error: (error: STTError) => void;\r\n /* Engine state change. /\r\n status: (state: STTState) => void;\r\n /* Diagnostic log for debugging (subscribe to capture all internal events). /\r\n debug: (message: string) => void;\r\n};\r\n\r\n/* Handle returned by audio capture — used internally. /\r\nexport interface AudioCaptureHandle {\r\n audioCtx: AudioContext;\r\n stream: MediaStream;\r\n samples: Float32Array[];\r\n /* Retain reference to prevent GC from stopping audio processing. /\r\n _processor: ScriptProcessorNode;\r\n /* Source node for disconnect/reconnect on pause/resume. /\r\n _source: MediaStreamAudioSourceNode;\r\n /* Gain node (silent) to prevent mic playback. /\r\n _silencer: GainNode;\r\n}\r\n\r\n/* Message sent from main thread to Whisper worker. /\r\nexport interface WorkerMessage {\r\n type: 'load' \| 'transcribe' \| 'cancel';\r\n audio?: Float32Array;\r\n config?: {\r\n model: string;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n chunkLengthS: number;\r\n strideLengthS: number;\r\n };\r\n}\r\n\r\n/* Response sent from Whisper worker to main thread. /\r\nexport interface WorkerResponse {\r\n type: 'progress' \| 'ready' \| 'result' \| 'error';\r\n data?: unknown;\r\n}\r\n\r\n/* Default configuration values. /\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n model: 'tiny',\r\n backend: 'auto',\r\n language: 'en',\r\n dtype: 'q4',\r\n correction: {\r\n enabled: true,\r\n provider: 'whisper',\r\n pauseThreshold: 1_000,\r\n forcedInterval: 3_000,\r\n },\r\n chunking: {\r\n chunkLengthS: 30,\r\n strideLengthS: 5,\r\n },\r\n streaming: {\r\n enabled: true,\r\n provider: 'web-speech-api',\r\n },\r\n};\r\n\r\n/* Merge user config with defaults to produce resolved config. /\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n return {\r\n model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n correction: {\r\n enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n pauseThreshold:\r\n config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n forcedInterval:\r\n config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n },\r\n chunking: {\r\n chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n },\r\n streaming: {\r\n enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n },\r\n };\r\n}\r\n","/\n A generic, typed event emitter.\n \n Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n /\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n /* Subscribe to an event. /\n on<K extends keyof T>(event: K, listener: T[K]): void {\n let set = this.listeners.get(event);\n if (!set) {\n set = new Set();\n this.listeners.set(event, set);\n }\n set.add(listener as T[keyof T]);\n }\n\n /* Unsubscribe a specific listener. No-op if not registered. /\n off<K extends keyof T>(event: K, listener: T[K]): void {\n this.listeners.get(event)?.delete(listener as T[keyof T]);\n }\n\n /* Emit an event, calling all registered listeners in insertion order. /\n emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n const set = this.listeners.get(event);\n if (!set) return;\n for (const listener of set) {\n (listener as (...a: Parameters<T[K]>) => void)(...args);\n }\n }\n\n /* Remove all listeners, optionally for a single event. /\n removeAllListeners(event?: keyof T): void {\n if (event !== undefined) {\n this.listeners.delete(event);\n } else {\n this.listeners.clear();\n }\n }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/\r\n Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n /\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n const stream = await navigator.mediaDevices.getUserMedia({\r\n audio: { channelCount: 1 },\r\n });\r\n const audioCtx = new AudioContext();\r\n\r\n // Chrome may suspend AudioContext — must resume within user gesture\r\n if (audioCtx.state === 'suspended') {\r\n await audioCtx.resume();\r\n }\r\n\r\n const source = audioCtx.createMediaStreamSource(stream);\r\n const samples: Float32Array[] = [];\r\n\r\n const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n };\r\n\r\n // Connect through a silent gain node so mic audio doesn't play back\r\n const silencer = audioCtx.createGain();\r\n silencer.gain.value = 0;\r\n source.connect(processor);\r\n processor.connect(silencer);\r\n silencer.connect(audioCtx.destination);\r\n\r\n return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };\r\n}\r\n\r\n/\r\n Pause capture without releasing mic or AudioContext.\r\n * Disconnects the audio source so no new samples are collected.\r\n * Returns resampled audio from the recording period.\r\n * Call resumeCapture() to start collecting again.\r\n /\r\nexport async function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n capture._source.disconnect();\r\n const currentSamples = [...capture.samples];\r\n capture.samples.length = 0;\r\n return resampleAudio(currentSamples, capture.audioCtx.sampleRate);\r\n}\r\n\r\n/\r\n Resume a paused capture. Reconnects the audio source to the processor.\r\n * AudioContext is resumed if suspended.\r\n /\r\nexport async function resumeCapture(capture: AudioCaptureHandle): Promise<void> {\r\n if (capture.audioCtx.state === 'suspended') {\r\n await capture.audioCtx.resume();\r\n }\r\n capture._source.connect(capture._processor);\r\n}\r\n\r\n/\r\n Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n /\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n return [...capture.samples];\r\n}\r\n\r\n/\r\n Trim the audio buffer to keep only the most recent `keepSeconds` of audio.\r\n * Call this after each mid-recording correction to prevent unbounded buffer growth.\r\n * Without trimming, a 15-minute session accumulates ~158 MB and makes each\r\n * correction send 15 minutes of audio to Whisper.\r\n /\r\nexport function trimAudioBuffer(capture: AudioCaptureHandle, keepSeconds: number): void {\r\n const samplesPerChunk = 4096; // matches ScriptProcessorNode buffer size in startCapture()\r\n const chunksToKeep = Math.ceil((keepSeconds capture.audioCtx.sampleRate) / samplesPerChunk);\r\n if (capture.samples.length > chunksToKeep) {\r\n capture.samples.splice(0, capture.samples.length - chunksToKeep);\r\n }\r\n}\r\n\r\n/*\r\n Concatenate sample chunks and resample to 16kHz for Whisper.\r\n /\r\nexport async function resampleAudio(\r\n samples: Float32Array[],\r\n nativeSr: number,\r\n): Promise<Float32Array> {\r\n const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n if (totalLength === 0) return new Float32Array(0);\r\n\r\n const fullAudio = new Float32Array(totalLength);\r\n let offset = 0;\r\n for (const s of samples) {\r\n fullAudio.set(s, offset);\r\n offset += s.length;\r\n }\r\n\r\n if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n const duration = fullAudio.length / nativeSr;\r\n const outLength = Math.round(duration TARGET_SAMPLE_RATE);\r\n const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n buffer.getChannelData(0).set(fullAudio);\r\n const src = offline.createBufferSource();\r\n src.buffer = buffer;\r\n src.connect(offline.destination);\r\n src.start(0);\r\n const resampled = await offline.startRendering();\r\n return resampled.getChannelData(0);\r\n}\r\n\r\n/*\r\n Stop capturing and return resampled audio at 16kHz.\r\n /\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n // Disconnect processor to stop capturing\r\n try {\r\n _processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n\r\n // Stop microphone tracks\r\n for (const track of stream.getTracks()) {\r\n track.stop();\r\n }\r\n\r\n const nativeSr = audioCtx.sampleRate;\r\n await audioCtx.close();\r\n\r\n return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/* Events emitted by the WorkerManager. /\nexport type WorkerManagerEvents = {\n progress: (percent: number) => void;\n ready: () => void;\n result: (text: string) => void;\n error: (message: string) => void;\n};\n\n/\n Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n /\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n private worker: Worker \| null = null;\n private transcribeResolve: ((text: string) => void) \| null = null;\n private currentTranscribePromise: Promise<string> \| null = null;\n private modelReadyResolve: (() => void) \| null = null;\n private modelReadyReject: ((err: Error) => void) \| null = null;\n\n /* True while a transcription job is running in the worker. /\n get isTranscribing(): boolean {\n return this.transcribeResolve !== null;\n }\n\n /* Await the current in-flight transcription without starting a new one. /\n awaitCurrentTranscription(): Promise<string> {\n return this.currentTranscribePromise ?? Promise.resolve('');\n }\n\n /* Spawn the Web Worker. Must be called before loadModel/transcribe. /\n spawn(workerUrl?: URL): void {\n if (this.worker) return;\n\n const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n this.worker = new Worker(url, { type: 'module' });\n this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n this.handleMessage(e.data);\n };\n this.worker.onerror = (e: ErrorEvent) => {\n this.emit('error', e.message ?? 'Worker error');\n };\n }\n\n /* Load the Whisper model in the worker. Resolves when ready. /\n async loadModel(config: ResolvedSTTConfig): Promise<void> {\n if (!this.worker) throw new Error('Worker not spawned');\n\n return new Promise<void>((resolve, reject) => {\n this.modelReadyResolve = resolve;\n this.modelReadyReject = reject;\n this.worker!.postMessage({\n type: 'load',\n config: {\n model: config.model,\n backend: config.backend,\n language: config.language,\n dtype: config.dtype,\n chunkLengthS: config.chunking.chunkLengthS,\n strideLengthS: config.chunking.strideLengthS,\n },\n });\n });\n }\n\n /* Send audio to the worker for transcription. Resolves with text. /\n async transcribe(audio: Float32Array): Promise<string> {\n if (!this.worker) throw new Error('Worker not spawned');\n if (audio.length === 0) return '';\n\n this.currentTranscribePromise = new Promise<string>((resolve) => {\n this.transcribeResolve = resolve;\n this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n });\n return this.currentTranscribePromise;\n }\n\n /* Cancel any in-flight transcription. /\n cancel(): void {\n this.worker?.postMessage({ type: 'cancel' });\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n }\n\n /* Terminate the worker and release resources. /\n destroy(): void {\n this.cancel();\n this.worker?.terminate();\n this.worker = null;\n this.removeAllListeners();\n }\n\n private handleMessage(msg: WorkerResponse): void {\n switch (msg.type) {\n case 'progress':\n this.emit('progress', msg.data as number);\n break;\n case 'ready':\n this.emit('ready');\n this.modelReadyResolve?.();\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n break;\n case 'result':\n this.emit('result', msg.data as string);\n this.transcribeResolve?.(msg.data as string);\n this.transcribeResolve = null;\n break;\n case 'error': {\n const errMsg = msg.data as string;\n this.emit('error', errMsg);\n // Reject model load if still pending\n if (this.modelReadyReject) {\n this.modelReadyReject(new Error(errMsg));\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n }\n // Resolve transcribe with empty string on error\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n break;\n }\n }\n }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/\n Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n /\n/* Delay (ms) before the first correction fires after recording starts. /\nconst INITIAL_CORRECTION_DELAY_MS = 1_000;\n\nexport class CorrectionOrchestrator {\n private forcedTimer: ReturnType<typeof setInterval> \| null = null;\n private initialTimer: ReturnType<typeof setTimeout> \| 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.\n * Fires a quick initial correction after 1s for early feedback, then\n * switches to the regular forcedInterval cadence from that point. /\n start(): void {\n if (!this.config.enabled) return;\n\n this.lastCorrectionTime = Date.now();\n // One-shot early correction — gives the user immediate Whisper feedback\n // before the Web Speech API has produced its first interim result (~1-2s).\n this.initialTimer = setTimeout(() => {\n this.initialTimer = null;\n this.correctionFn?.();\n this.lastCorrectionTime = Date.now();\n this.startForcedTimer();\n }, INITIAL_CORRECTION_DELAY_MS);\n }\n\n /* Stop the orchestrator (clear all timers). /\n stop(): void {\n if (this.initialTimer) {\n clearTimeout(this.initialTimer);\n this.initialTimer = null;\n }\n this.stopForcedTimer();\n }\n\n /* Called when a speech pause is detected. Triggers correction if cooldown elapsed. /\n onPauseDetected(): void {\n if (!this.config.enabled) return;\n\n const now = Date.now();\n if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n this.triggerCorrection();\n }\n\n /* Force a correction now (resets timer). /\n forceCorrection(): void {\n this.triggerCorrection();\n }\n\n private triggerCorrection(): void {\n this.lastCorrectionTime = Date.now();\n this.correctionFn?.();\n // Reset forced timer after any correction\n this.restartForcedTimer();\n }\n\n private startForcedTimer(): void {\n this.stopForcedTimer();\n this.forcedTimer = setInterval(() => {\n this.triggerCorrection();\n }, this.config.forcedInterval);\n }\n\n private stopForcedTimer(): void {\n if (this.forcedTimer) {\n clearInterval(this.forcedTimer);\n this.forcedTimer = null;\n }\n }\n\n private restartForcedTimer(): void {\n if (this.forcedTimer) {\n this.startForcedTimer();\n }\n }\n}\n","/ ─── Web Speech API types ──────────────────────────────── /\r\n\r\ninterface SpeechRecognitionEvent {\r\n results: SpeechRecognitionResultList;\r\n resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n continuous: boolean;\r\n interimResults: boolean;\r\n lang: string;\r\n onaudiostart: (() => void) \| null;\r\n onresult: ((e: SpeechRecognitionEvent) => void) \| null;\r\n onerror: ((e: SpeechRecognitionErrorEvent) => void) \| null;\r\n onend: (() => void) \| null;\r\n start: () => void;\r\n stop: () => void;\r\n abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor \| null {\r\n if (typeof globalThis === 'undefined') return null;\r\n const w = globalThis as unknown as Record<string, unknown>;\r\n return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor \| null;\r\n}\r\n\r\n/ ─── Language mapping ──────────────────────────────────── /\r\n\r\n/* Map Whisper language codes to BCP-47 locale tags for the Speech API. /\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n en: 'en-US',\r\n english: 'en-US',\r\n zh: 'zh-CN',\r\n chinese: 'zh-CN',\r\n de: 'de-DE',\r\n german: 'de-DE',\r\n es: 'es-ES',\r\n spanish: 'es-ES',\r\n ru: 'ru-RU',\r\n russian: 'ru-RU',\r\n ko: 'ko-KR',\r\n korean: 'ko-KR',\r\n fr: 'fr-FR',\r\n french: 'fr-FR',\r\n ja: 'ja-JP',\r\n japanese: 'ja-JP',\r\n pt: 'pt-BR',\r\n portuguese: 'pt-BR',\r\n tr: 'tr-TR',\r\n turkish: 'tr-TR',\r\n pl: 'pl-PL',\r\n polish: 'pl-PL',\r\n nl: 'nl-NL',\r\n dutch: 'nl-NL',\r\n ar: 'ar-SA',\r\n arabic: 'ar-SA',\r\n sv: 'sv-SE',\r\n swedish: 'sv-SE',\r\n it: 'it-IT',\r\n italian: 'it-IT',\r\n id: 'id-ID',\r\n indonesian: 'id-ID',\r\n hi: 'hi-IN',\r\n hindi: 'hi-IN',\r\n fi: 'fi-FI',\r\n finnish: 'fi-FI',\r\n vi: 'vi-VN',\r\n vietnamese: 'vi-VN',\r\n he: 'he-IL',\r\n hebrew: 'he-IL',\r\n uk: 'uk-UA',\r\n ukrainian: 'uk-UA',\r\n el: 'el-GR',\r\n greek: 'el-GR',\r\n ms: 'ms-MY',\r\n malay: 'ms-MY',\r\n cs: 'cs-CZ',\r\n czech: 'cs-CZ',\r\n ro: 'ro-RO',\r\n romanian: 'ro-RO',\r\n da: 'da-DK',\r\n danish: 'da-DK',\r\n hu: 'hu-HU',\r\n hungarian: 'hu-HU',\r\n no: 'nb-NO',\r\n norwegian: 'nb-NO',\r\n th: 'th-TH',\r\n thai: 'th-TH',\r\n};\r\n\r\n/\r\n Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n /\r\nfunction toBCP47(language: string): string {\r\n if (language.includes('-')) return language;\r\n return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/ ─── SpeechStreamingManager ────────────────────────────── /\r\n\r\n/\r\n Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n \r\n Keep-alive design: the recognition session is never stopped between user\r\n * sessions (only muted). This eliminates the 1-2s Google server handshake\r\n * on every click — start() resolves instantly when the session is warm.\r\n /\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n/* How long to keep recognition running silently after stop() before truly releasing it. /\r\nconst IDLE_TIMEOUT_MS = 30_000;\r\n\r\nexport class SpeechStreamingManager {\r\n private recognition: SpeechRecognitionInstance \| null = null;\r\n private accumulated = '';\r\n private active = false; // user is recording — emit results, trigger onPause on end\r\n private keepingWarm = false; // recognition running silently between user sessions\r\n private currentLang = ''; // BCP-47 of running recognition (for fast-path check)\r\n private receivedResult = false;\r\n private lastFinalIndex = -1; // instance fields so warm-restart resets them cleanly\r\n private lastFinalText = '';\r\n private noResultTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private idleTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private onTranscript: ((text: string) => void) \| null = null;\r\n private onPause: (() => void) \| null = null;\r\n private onError: ((message: string) => void) \| null = null;\r\n private onDebug: ((message: string) => void) \| null = null;\r\n\r\n /* Check if the Web Speech API is available in this environment. /\r\n static isSupported(): boolean {\r\n return getSpeechRecognition() !== null;\r\n }\r\n\r\n /* Set callback for streaming transcript updates (interim + final text). /\r\n setOnTranscript(fn: (text: string) => void): void {\r\n this.onTranscript = fn;\r\n }\r\n\r\n /* Set callback for speech pause detection (Speech API onend). /\r\n setOnPause(fn: () => void): void {\r\n this.onPause = fn;\r\n }\r\n\r\n /* Set callback for errors. /\r\n setOnError(fn: (message: string) => void): void {\r\n this.onError = fn;\r\n }\r\n\r\n /* Set callback for diagnostic debug messages. /\r\n setOnDebug(fn: (message: string) => void): void {\r\n this.onDebug = fn;\r\n }\r\n\r\n private log(message: string): void {\r\n this.onDebug?.(message);\r\n console.warn(message);\r\n }\r\n\r\n /\r\n Pre-warm: start recognition in muted mode so it's ready before the user\r\n * clicks. Call after engine.init() completes. Eliminates startup latency on\r\n * first click by keeping the Google Speech session alive.\r\n /\r\n preWarm(language: string): void {\r\n const SR = getSpeechRecognition();\r\n if (!SR) return;\r\n const bcp47 = toBCP47(language);\r\n if (this.recognition && this.currentLang === bcp47) return; // already warm\r\n this.log(`[SSM] preWarm() — lang: \"${language}\" → \"${bcp47}\"`);\r\n this.keepingWarm = true;\r\n this.active = false;\r\n this.clearIdleTimer();\r\n this.spawnRecognition(language);\r\n }\r\n\r\n /\r\n Start streaming recognition. If recognition is already warm (session\r\n * running from preWarm or a previous session within the idle window),\r\n * activates instantly — no Google handshake. Otherwise cold-starts.\r\n /\r\n start(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition();\r\n if (!SR) {\r\n this.log('[SSM] SpeechRecognition not available in this environment');\r\n return Promise.resolve();\r\n }\r\n\r\n const bcp47 = toBCP47(language);\r\n this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n this.accumulated = '';\r\n this.receivedResult = false;\r\n this.lastFinalIndex = -1;\r\n this.lastFinalText = '';\r\n this.clearIdleTimer();\r\n\r\n // ── Fast path: session already running with the right language ──────────\r\n if (this.recognition && this.currentLang === bcp47) {\r\n this.log('[SSM] start() — warm session, activating immediately');\r\n this.keepingWarm = false;\r\n this.active = true;\r\n // Arm the no-result watchdog for silent-failure detection\r\n this.clearNoResultTimer();\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n return Promise.resolve();\r\n }\r\n\r\n // ── Cold start: spin up a fresh recognition session ─────────────────────\r\n this.keepingWarm = false;\r\n this.active = true;\r\n return this.spawnRecognition(language, skipMicWait);\r\n }\r\n\r\n /\r\n Create and start a new SpeechRecognition instance.\r\n * Used by both preWarm() (active=false) and start() cold path (active=true).\r\n /\r\n private spawnRecognition(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition()!;\r\n const bcp47 = toBCP47(language);\r\n\r\n // Tear down any stale session first\r\n if (this.recognition) {\r\n const old = this.recognition;\r\n this.recognition = null;\r\n try { old.stop(); } catch { / already stopped / }\r\n }\r\n\r\n this.currentLang = bcp47;\r\n\r\n const recognition = new SR();\r\n recognition.continuous = true;\r\n recognition.interimResults = true;\r\n recognition.lang = bcp47;\r\n\r\n // Mic-claim promise (used on cold active starts only)\r\n let micReady = false;\r\n const micClaimPromise = new Promise<void>((resolve) => {\r\n recognition.onaudiostart = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n if (!micReady) { micReady = true; resolve(); }\r\n };\r\n setTimeout(() => {\r\n if (!micReady) {\r\n micReady = true;\r\n this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n resolve();\r\n }\r\n }, 300);\r\n });\r\n\r\n // No-result watchdog — only when actively recording\r\n this.clearNoResultTimer();\r\n if (this.active) {\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n }\r\n\r\n recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.receivedResult = true;\r\n this.clearNoResultTimer();\r\n\r\n // Discard results while muted (keeping warm between user sessions)\r\n if (!this.active) return;\r\n\r\n let final_ = '';\r\n let interim = '';\r\n for (let i = e.resultIndex; i < e.results.length; i++) {\r\n const t = e.results[i][0].transcript;\r\n if (e.results[i].isFinal) {\r\n if (i > this.lastFinalIndex) {\r\n final_ += t;\r\n this.lastFinalIndex = i;\r\n }\r\n } else {\r\n interim += t;\r\n }\r\n }\r\n\r\n this.log(\r\n `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n );\r\n\r\n if (final_ && final_.trim() !== this.lastFinalText) {\r\n this.lastFinalText = final_.trim();\r\n this.accumulated = this.accumulated\r\n ? this.accumulated + ' ' + final_.trim()\r\n : final_.trim();\r\n this.onTranscript?.(this.accumulated);\r\n } else if (interim) {\r\n const trimmed = interim.trimStart();\r\n const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n this.onTranscript?.(full);\r\n }\r\n };\r\n\r\n recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.log(`[SSM] onerror — ${e.error}`);\r\n // Suppress errors while muted — they're not relevant to the user\r\n if (this.active) this.onError?.(e.error);\r\n };\r\n\r\n recognition.onend = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log(\r\n `[SSM] onend — active: ${this.active}, keepingWarm: ${this.keepingWarm}, receivedResult: ${this.receivedResult}`,\r\n );\r\n\r\n if (this.active) {\r\n // User is recording — trigger correction and restart for continued streaming\r\n this.onPause?.();\r\n try {\r\n recognition.start();\r\n this.log('[SSM] restarted after pause');\r\n } catch (err) {\r\n this.log(`[SSM] restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n this.onError?.('Speech recognition failed to restart after pause.');\r\n }\r\n } else if (this.keepingWarm) {\r\n // Between user sessions — restart silently to keep the session alive\r\n try {\r\n recognition.start();\r\n this.log('[SSM] warm restart');\r\n } catch (err) {\r\n this.log(`[SSM] warm restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.keepingWarm = false;\r\n this.currentLang = '';\r\n }\r\n } else {\r\n this.recognition = null;\r\n this.currentLang = '';\r\n }\r\n };\r\n\r\n this.recognition = recognition;\r\n try {\r\n recognition.start();\r\n this.log('[SSM] recognition.start() succeeded');\r\n } catch (err) {\r\n this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n const wasActive = this.active;\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n if (wasActive) {\r\n this.onError?.(\r\n `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n );\r\n }\r\n return Promise.resolve();\r\n }\r\n\r\n // preWarm path or skipMicWait — no need to wait for mic claim\r\n if (!this.active \|\| skipMicWait) {\r\n if (skipMicWait) this.log('[SSM] skipMicWait — warm restart, returning immediately');\r\n return Promise.resolve();\r\n }\r\n\r\n return micClaimPromise;\r\n }\r\n\r\n private clearNoResultTimer(): void {\r\n if (this.noResultTimer) {\r\n clearTimeout(this.noResultTimer);\r\n this.noResultTimer = null;\r\n }\r\n }\r\n\r\n private clearIdleTimer(): void {\r\n if (this.idleTimer) {\r\n clearTimeout(this.idleTimer);\r\n this.idleTimer = null;\r\n }\r\n }\r\n\r\n private startIdleTimer(): void {\r\n this.clearIdleTimer();\r\n this.idleTimer = setTimeout(() => {\r\n this.idleTimer = null;\r\n this.keepingWarm = false;\r\n this.log('[SSM] idle timeout — stopping recognition');\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n try { rec.stop(); } catch { / already stopped / }\r\n }\r\n }, IDLE_TIMEOUT_MS);\r\n }\r\n\r\n /* Stop streaming recognition and return accumulated text.\r\n * Keeps the recognition session alive (muted) for instant restart. /\r\n stop(): string {\r\n this.active = false;\r\n this.keepingWarm = true;\r\n this.clearNoResultTimer();\r\n const result = this.accumulated;\r\n this.accumulated = '';\r\n // Keep recognition running silently; release after idle timeout\r\n this.startIdleTimer();\r\n return result;\r\n }\r\n\r\n /* Abort immediately and release all resources. /\r\n destroy(): void {\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n this.clearIdleTimer();\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n rec.abort();\r\n }\r\n this.accumulated = '';\r\n this.onTranscript = null;\r\n this.onPause = null;\r\n this.onError = null;\r\n this.onDebug = null;\r\n }\r\n}\r\n","import type {\r\n STTConfig,\r\n STTState,\r\n STTEvents,\r\n STTStatus,\r\n ResolvedSTTConfig,\r\n AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, pauseCapture, resumeCapture, snapshotAudio, resampleAudio, trimAudioBuffer } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/\r\n Main STT engine — the public API for speech-to-text with Whisper correction.\r\n \r\n Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n /\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n private config: ResolvedSTTConfig;\r\n private workerManager: WorkerManager;\r\n private correctionOrchestrator: CorrectionOrchestrator;\r\n private speechStreaming: SpeechStreamingManager;\r\n private capture: AudioCaptureHandle \| null = null;\r\n private state: STTState;\r\n private workerUrl?: URL;\r\n /* Prevents performCorrection from emitting while stop() is consuming the in-flight result. /\r\n private _stopping = false;\r\n\r\n /\r\n Create a new STT engine instance.\r\n * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n /\r\n constructor(config?: STTConfig, workerUrl?: URL) {\r\n super();\r\n this.config = resolveConfig(config);\r\n this.workerManager = new WorkerManager();\r\n this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n this.speechStreaming = new SpeechStreamingManager();\r\n this.workerUrl = workerUrl;\r\n\r\n this.state = {\r\n status: 'idle',\r\n isModelLoaded: false,\r\n loadProgress: 0,\r\n backend: null,\r\n error: null,\r\n };\r\n\r\n this.correctionOrchestrator.setCorrectionFn(() => {\r\n this.performCorrection();\r\n });\r\n\r\n this.setupWorkerListeners();\r\n this.setupStreamingCallbacks();\r\n }\r\n\r\n /* Initialize the engine: spawn worker and load model. /\r\n async init(): Promise<void> {\r\n this.updateStatus('loading');\r\n this.workerManager.spawn(this.workerUrl);\r\n\r\n try {\r\n await this.workerManager.loadModel(this.config);\r\n this.state.isModelLoaded = true;\r\n this.updateStatus('ready');\r\n // Pre-warm recognition so the first click is instant (no Google handshake delay)\r\n if (this.config.streaming.enabled) {\r\n this.speechStreaming.preWarm(this.config.language);\r\n }\r\n } catch (err) {\r\n this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n this.updateStatus('idle');\r\n throw err;\r\n }\r\n }\r\n\r\n /* Start recording audio and enable correction cycles. /\r\n async start(): Promise<void> {\r\n if (this.state.status !== 'ready') {\r\n throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n }\r\n\r\n try {\r\n // Check if mic is already warm from a previous recording session.\r\n // When warm, skip getUserMedia and the 300ms SR mic-claim wait.\r\n const warmCapture =\r\n this.capture &&\r\n this.capture.stream.getTracks().every((t) => t.readyState === 'live');\r\n\r\n this.emitDebug(\r\n `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\", warm: ${!!warmCapture}`,\r\n );\r\n\r\n if (this.config.streaming.enabled) {\r\n // On warm restart, skip the mic-claim wait — no getUserMedia race.\r\n await this.speechStreaming.start(this.config.language, !!warmCapture);\r\n if (!warmCapture) {\r\n this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n }\r\n }\r\n\r\n if (warmCapture) {\r\n await resumeCapture(this.capture!);\r\n this.emitDebug('[STT] warm mic resumed — skipped getUserMedia');\r\n } else {\r\n // First start or stale capture: full mic init with getUserMedia.\r\n this.capture = await startCapture();\r\n }\r\n\r\n this.updateStatus('recording');\r\n this.correctionOrchestrator.start();\r\n } catch (err) {\r\n this.emitError(\r\n 'MIC_DENIED',\r\n err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n );\r\n }\r\n }\r\n\r\n /* Stop recording, run final transcription, return text.\r\n * Mic and AudioContext stay alive for fast restart — call destroy() to fully release. /\r\n async stop(): Promise<string> {\r\n if (!this.capture) return '';\r\n\r\n // Prevent any in-flight performCorrection from emitting — stop() will own the final emit.\r\n this._stopping = true;\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.stop();\r\n\r\n this.updateStatus('processing');\r\n\r\n // If a correction job is already running, reuse it: Whisper can't be interrupted mid-inference\r\n // anyway, so cancelling just wastes the work and causes a second full run.\r\n if (this.workerManager.isTranscribing) {\r\n try {\r\n // Resample audio and await the in-flight result in parallel.\r\n const [audio, inFlightText] = await Promise.all([\r\n pauseCapture(this.capture),\r\n this.workerManager.awaitCurrentTranscription(),\r\n ]);\r\n this._stopping = false;\r\n\r\n const text = inFlightText.trim();\r\n if (text) {\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n }\r\n\r\n // In-flight returned empty (cancelled or error) — fall through to fresh transcription.\r\n if (audio.length > 0) {\r\n const freshText = await this.workerManager.transcribe(audio);\r\n this.emit('correction', freshText);\r\n this.updateStatus('ready');\r\n return freshText;\r\n }\r\n\r\n this.updateStatus('ready');\r\n return '';\r\n } catch (err) {\r\n this._stopping = false;\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n // No job in-flight — cancel is a no-op, run fresh transcription.\r\n this.workerManager.cancel();\r\n this._stopping = false;\r\n\r\n try {\r\n // Soft pause — keeps stream and AudioContext alive for fast restart.\r\n const audio = await pauseCapture(this.capture);\r\n\r\n if (audio.length === 0) {\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n /* Destroy the engine: terminate worker, release all resources. /\r\n destroy(): void {\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.destroy();\r\n\r\n if (this.capture) {\r\n try {\r\n this.capture._processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n for (const track of this.capture.stream.getTracks()) {\r\n track.stop();\r\n }\r\n this.capture.audioCtx.close().catch(() => {});\r\n this.capture = null;\r\n }\r\n\r\n this.workerManager.destroy();\r\n this.updateStatus('idle');\r\n this.removeAllListeners();\r\n }\r\n\r\n /* Get current engine state. /\r\n getState(): Readonly<STTState> {\r\n return { ...this.state };\r\n }\r\n\r\n /* Notify the correction orchestrator of a speech pause. */\r\n notifyPause(): void {\r\n this.correctionOrchestrator.onPauseDetected();\r\n }\r\n\r\n private async performCorrection(): Promise<void> {\r\n if (!this.capture \|\| !this.state.isModelLoaded) return;\r\n\r\n this.workerManager.cancel();\r\n\r\n try {\r\n const samples = snapshotAudio(this.capture);\r\n const nativeSr = this.capture.audioCtx.sampleRate;\r\n const audio = await resampleAudio(samples, nativeSr);\r\n\r\n if (audio.length === 0) return;\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n if (text.trim() && this.capture && !this._stopping) {\r\n this.emit('correction', text);\r\n // Trim old audio — keep last 30s so the buffer doesn't grow unboundedly\r\n // during long sessions. Without this, a 15-min session accumulates ~158 MB\r\n // and makes each correction (and the final stop) process 15 min of audio.\r\n trimAudioBuffer(this.capture, 30);\r\n }\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Correction transcription failed.',\r\n );\r\n // Recording continues — error is non-fatal\r\n }\r\n }\r\n\r\n private setupStreamingCallbacks(): void {\r\n this.speechStreaming.setOnDebug((message) => {\r\n this.emit('debug', message);\r\n });\r\n\r\n this.speechStreaming.setOnTranscript((text) => {\r\n this.emitDebug(`[STT] transcript callback — \"${text}\"`);\r\n this.emit('transcript', text);\r\n });\r\n\r\n this.speechStreaming.setOnPause(() => {\r\n this.emitDebug('[STT] pause callback — triggering correction');\r\n this.correctionOrchestrator.onPauseDetected();\r\n });\r\n\r\n this.speechStreaming.setOnError((message) => {\r\n this.emitDebug(`[STT] streaming error — \"${message}\"`);\r\n this.emitError('STREAMING_ERROR', message);\r\n });\r\n }\r\n\r\n private setupWorkerListeners(): void {\r\n this.workerManager.on('progress', (percent) => {\r\n this.state.loadProgress = percent;\r\n this.emit('status', { ...this.state });\r\n });\r\n\r\n this.workerManager.on('error', (message) => {\r\n this.emitError('WORKER_ERROR', message);\r\n });\r\n }\r\n\r\n private updateStatus(status: STTStatus): void {\r\n this.state.status = status;\r\n this.state.error = null;\r\n this.emit('status', { ...this.state });\r\n }\r\n\r\n private emitError(code: string, message: string): void {\r\n this.state.error = message;\r\n this.emit('error', { code, message });\r\n }\r\n\r\n private emitDebug(message: string): void {\r\n console.warn(message);\r\n this.emit('debug', message);\r\n 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package/dist/index.js CHANGED Viewed

@@ -114,6 +114,13 @@ async function resumeCapture(capture) {
 function snapshotAudio(capture) {
   return [...capture.samples];
 }
+function trimAudioBuffer(capture, keepSeconds) {
+  const samplesPerChunk = 4096;
+  const chunksToKeep = Math.ceil(keepSeconds * capture.audioCtx.sampleRate / samplesPerChunk);
+  if (capture.samples.length > chunksToKeep) {
+    capture.samples.splice(0, capture.samples.length - chunksToKeep);
+  }
+}
 async function resampleAudio(samples, nativeSr) {
   const totalLength = samples.reduce((sum, s) => sum + s.length, 0);
   if (totalLength === 0) return new Float32Array(0);
@@ -879,6 +886,7 @@ var STTEngine = class extends TypedEventEmitter {
       const text = await this.workerManager.transcribe(audio);
       if (text.trim() && this.capture && !this._stopping) {
         this.emit("correction", text);
+        trimAudioBuffer(this.capture, 30);
       }
     } catch (err) {
       this.emitError(

package/dist/index.js.map CHANGED Viewed

	@@ -1 +1 @@
1	- {"version":3,"sources":["../src/types.ts","../src/event-emitter.ts","../src/audio-capture.ts","../src/worker-manager.ts","../src/correction-orchestrator.ts","../src/speech-streaming.ts","../src/stt-engine.ts"],"sourcesContent":["/** Supported Whisper model sizes. /\r\nexport type STTModelSize = 'tiny' \| 'base' \| 'small' \| 'medium';\r\n\r\n/* Supported compute backends. /\r\nexport type STTBackend = 'webgpu' \| 'wasm' \| 'auto';\r\n\r\n/* Engine lifecycle states. /\r\nexport type STTStatus = 'idle' \| 'loading' \| 'ready' \| 'recording' \| 'processing';\r\n\r\n/* Supported correction engine providers. /\r\nexport type STTCorrectionProvider = 'whisper';\r\n\r\n/* Supported real-time streaming providers. /\r\nexport type STTStreamingProvider = 'web-speech-api';\r\n\r\n/* Correction engine configuration. /\r\nexport interface STTCorrectionConfig {\r\n /* Enable mid-recording correction. Default: true /\r\n enabled?: boolean;\r\n /* Correction engine provider. Default: 'whisper' /\r\n provider?: STTCorrectionProvider;\r\n /* Silence duration (ms) before triggering correction. Default: 1000 /\r\n pauseThreshold?: number;\r\n /* Maximum interval (ms) between forced corrections. Default: 3000 /\r\n forcedInterval?: number;\r\n}\r\n\r\n/* Real-time streaming preview configuration. /\r\nexport interface STTStreamingConfig {\r\n /* Enable real-time streaming transcript. Default: true /\r\n enabled?: boolean;\r\n /* Streaming provider. Default: 'web-speech-api' /\r\n provider?: STTStreamingProvider;\r\n}\r\n\r\n/* Audio chunking configuration for long-form audio. /\r\nexport interface STTChunkingConfig {\r\n /* Chunk length in seconds for Whisper processing. Default: 30 /\r\n chunkLengthS?: number;\r\n /* Stride length in seconds for overlapping chunks. Default: 5 /\r\n strideLengthS?: number;\r\n}\r\n\r\n/* Full engine configuration. All fields optional — sensible defaults applied. /\r\nexport interface STTConfig {\r\n /* Whisper model size. Default: 'tiny' /\r\n model?: STTModelSize;\r\n /* Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) /\r\n backend?: STTBackend;\r\n /* Transcription language. Default: 'en' /\r\n language?: string;\r\n /* Model quantization dtype. Default: 'q4' /\r\n dtype?: string;\r\n /* Mid-recording correction settings. /\r\n correction?: STTCorrectionConfig;\r\n /* Audio chunking settings for long-form audio. /\r\n chunking?: STTChunkingConfig;\r\n /* Web Speech API streaming preview settings. /\r\n streaming?: STTStreamingConfig;\r\n}\r\n\r\n/* Resolved configuration with all defaults applied. /\r\nexport interface ResolvedSTTConfig {\r\n model: STTModelSize;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n correction: Required<STTCorrectionConfig>;\r\n chunking: Required<STTChunkingConfig>;\r\n streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/* Engine state exposed to consumers via status events. /\r\nexport interface STTState {\r\n status: STTStatus;\r\n isModelLoaded: boolean;\r\n /* Model download progress (0–100). /\r\n loadProgress: number;\r\n /* Active compute backend, or null if not yet determined. /\r\n backend: 'webgpu' \| 'wasm' \| null;\r\n error: string \| null;\r\n}\r\n\r\n/* Structured error emitted via the 'error' event. /\r\nexport interface STTError {\r\n code: string;\r\n message: string;\r\n}\r\n\r\n/* Event map for the typed event emitter. /\r\nexport type STTEvents = {\r\n /* Streaming interim text during recording. /\r\n transcript: (text: string) => void;\r\n /* Whisper-corrected text replacing interim text. /\r\n correction: (text: string) => void;\r\n /* Actionable error (mic denied, model fail, transcription fail). /\r\n error: (error: STTError) => void;\r\n /* Engine state change. /\r\n status: (state: STTState) => void;\r\n /* Diagnostic log for debugging (subscribe to capture all internal events). /\r\n debug: (message: string) => void;\r\n};\r\n\r\n/* Handle returned by audio capture — used internally. /\r\nexport interface AudioCaptureHandle {\r\n audioCtx: AudioContext;\r\n stream: MediaStream;\r\n samples: Float32Array[];\r\n /* Retain reference to prevent GC from stopping audio processing. /\r\n _processor: ScriptProcessorNode;\r\n /* Source node for disconnect/reconnect on pause/resume. /\r\n _source: MediaStreamAudioSourceNode;\r\n /* Gain node (silent) to prevent mic playback. /\r\n _silencer: GainNode;\r\n}\r\n\r\n/* Message sent from main thread to Whisper worker. /\r\nexport interface WorkerMessage {\r\n type: 'load' \| 'transcribe' \| 'cancel';\r\n audio?: Float32Array;\r\n config?: {\r\n model: string;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n chunkLengthS: number;\r\n strideLengthS: number;\r\n };\r\n}\r\n\r\n/* Response sent from Whisper worker to main thread. /\r\nexport interface WorkerResponse {\r\n type: 'progress' \| 'ready' \| 'result' \| 'error';\r\n data?: unknown;\r\n}\r\n\r\n/* Default configuration values. /\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n model: 'tiny',\r\n backend: 'auto',\r\n language: 'en',\r\n dtype: 'q4',\r\n correction: {\r\n enabled: true,\r\n provider: 'whisper',\r\n pauseThreshold: 1_000,\r\n forcedInterval: 3_000,\r\n },\r\n chunking: {\r\n chunkLengthS: 30,\r\n strideLengthS: 5,\r\n },\r\n streaming: {\r\n enabled: true,\r\n provider: 'web-speech-api',\r\n },\r\n};\r\n\r\n/* Merge user config with defaults to produce resolved config. /\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n return {\r\n model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n correction: {\r\n enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n pauseThreshold:\r\n config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n forcedInterval:\r\n config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n },\r\n chunking: {\r\n chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n },\r\n streaming: {\r\n enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n },\r\n };\r\n}\r\n","/\n A generic, typed event emitter.\n \n Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n /\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n /* Subscribe to an event. /\n on<K extends keyof T>(event: K, listener: T[K]): void {\n let set = this.listeners.get(event);\n if (!set) {\n set = new Set();\n this.listeners.set(event, set);\n }\n set.add(listener as T[keyof T]);\n }\n\n /* Unsubscribe a specific listener. No-op if not registered. /\n off<K extends keyof T>(event: K, listener: T[K]): void {\n this.listeners.get(event)?.delete(listener as T[keyof T]);\n }\n\n /* Emit an event, calling all registered listeners in insertion order. /\n emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n const set = this.listeners.get(event);\n if (!set) return;\n for (const listener of set) {\n (listener as (...a: Parameters<T[K]>) => void)(...args);\n }\n }\n\n /* Remove all listeners, optionally for a single event. /\n removeAllListeners(event?: keyof T): void {\n if (event !== undefined) {\n this.listeners.delete(event);\n } else {\n this.listeners.clear();\n }\n }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/\r\n Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n /\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n const stream = await navigator.mediaDevices.getUserMedia({\r\n audio: { channelCount: 1 },\r\n });\r\n const audioCtx = new AudioContext();\r\n\r\n // Chrome may suspend AudioContext — must resume within user gesture\r\n if (audioCtx.state === 'suspended') {\r\n await audioCtx.resume();\r\n }\r\n\r\n const source = audioCtx.createMediaStreamSource(stream);\r\n const samples: Float32Array[] = [];\r\n\r\n const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n };\r\n\r\n // Connect through a silent gain node so mic audio doesn't play back\r\n const silencer = audioCtx.createGain();\r\n silencer.gain.value = 0;\r\n source.connect(processor);\r\n processor.connect(silencer);\r\n silencer.connect(audioCtx.destination);\r\n\r\n return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };\r\n}\r\n\r\n/\r\n Pause capture without releasing mic or AudioContext.\r\n * Disconnects the audio source so no new samples are collected.\r\n * Returns resampled audio from the recording period.\r\n * Call resumeCapture() to start collecting again.\r\n /\r\nexport async function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n capture._source.disconnect();\r\n const currentSamples = [...capture.samples];\r\n capture.samples.length = 0;\r\n return resampleAudio(currentSamples, capture.audioCtx.sampleRate);\r\n}\r\n\r\n/\r\n Resume a paused capture. Reconnects the audio source to the processor.\r\n * AudioContext is resumed if suspended.\r\n /\r\nexport async function resumeCapture(capture: AudioCaptureHandle): Promise<void> {\r\n if (capture.audioCtx.state === 'suspended') {\r\n await capture.audioCtx.resume();\r\n }\r\n capture._source.connect(capture._processor);\r\n}\r\n\r\n/\r\n Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n /\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n return [...capture.samples];\r\n}\r\n\r\n/\r\n Concatenate sample chunks and resample to 16kHz for Whisper.\r\n /\r\nexport async function resampleAudio(\r\n samples: Float32Array[],\r\n nativeSr: number,\r\n): Promise<Float32Array> {\r\n const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n if (totalLength === 0) return new Float32Array(0);\r\n\r\n const fullAudio = new Float32Array(totalLength);\r\n let offset = 0;\r\n for (const s of samples) {\r\n fullAudio.set(s, offset);\r\n offset += s.length;\r\n }\r\n\r\n if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n const duration = fullAudio.length / nativeSr;\r\n const outLength = Math.round(duration TARGET_SAMPLE_RATE);\r\n const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n buffer.getChannelData(0).set(fullAudio);\r\n const src = offline.createBufferSource();\r\n src.buffer = buffer;\r\n src.connect(offline.destination);\r\n src.start(0);\r\n const resampled = await offline.startRendering();\r\n return resampled.getChannelData(0);\r\n}\r\n\r\n/*\r\n Stop capturing and return resampled audio at 16kHz.\r\n /\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n // Disconnect processor to stop capturing\r\n try {\r\n _processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n\r\n // Stop microphone tracks\r\n for (const track of stream.getTracks()) {\r\n track.stop();\r\n }\r\n\r\n const nativeSr = audioCtx.sampleRate;\r\n await audioCtx.close();\r\n\r\n return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/* Events emitted by the WorkerManager. /\nexport type WorkerManagerEvents = {\n progress: (percent: number) => void;\n ready: () => void;\n result: (text: string) => void;\n error: (message: string) => void;\n};\n\n/\n Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n /\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n private worker: Worker \| null = null;\n private transcribeResolve: ((text: string) => void) \| null = null;\n private currentTranscribePromise: Promise<string> \| null = null;\n private modelReadyResolve: (() => void) \| null = null;\n private modelReadyReject: ((err: Error) => void) \| null = null;\n\n /* True while a transcription job is running in the worker. /\n get isTranscribing(): boolean {\n return this.transcribeResolve !== null;\n }\n\n /* Await the current in-flight transcription without starting a new one. /\n awaitCurrentTranscription(): Promise<string> {\n return this.currentTranscribePromise ?? Promise.resolve('');\n }\n\n /* Spawn the Web Worker. Must be called before loadModel/transcribe. /\n spawn(workerUrl?: URL): void {\n if (this.worker) return;\n\n const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n this.worker = new Worker(url, { type: 'module' });\n this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n this.handleMessage(e.data);\n };\n this.worker.onerror = (e: ErrorEvent) => {\n this.emit('error', e.message ?? 'Worker error');\n };\n }\n\n /* Load the Whisper model in the worker. Resolves when ready. /\n async loadModel(config: ResolvedSTTConfig): Promise<void> {\n if (!this.worker) throw new Error('Worker not spawned');\n\n return new Promise<void>((resolve, reject) => {\n this.modelReadyResolve = resolve;\n this.modelReadyReject = reject;\n this.worker!.postMessage({\n type: 'load',\n config: {\n model: config.model,\n backend: config.backend,\n language: config.language,\n dtype: config.dtype,\n chunkLengthS: config.chunking.chunkLengthS,\n strideLengthS: config.chunking.strideLengthS,\n },\n });\n });\n }\n\n /* Send audio to the worker for transcription. Resolves with text. /\n async transcribe(audio: Float32Array): Promise<string> {\n if (!this.worker) throw new Error('Worker not spawned');\n if (audio.length === 0) return '';\n\n this.currentTranscribePromise = new Promise<string>((resolve) => {\n this.transcribeResolve = resolve;\n this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n });\n return this.currentTranscribePromise;\n }\n\n /* Cancel any in-flight transcription. /\n cancel(): void {\n this.worker?.postMessage({ type: 'cancel' });\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n }\n\n /* Terminate the worker and release resources. /\n destroy(): void {\n this.cancel();\n this.worker?.terminate();\n this.worker = null;\n this.removeAllListeners();\n }\n\n private handleMessage(msg: WorkerResponse): void {\n switch (msg.type) {\n case 'progress':\n this.emit('progress', msg.data as number);\n break;\n case 'ready':\n this.emit('ready');\n this.modelReadyResolve?.();\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n break;\n case 'result':\n this.emit('result', msg.data as string);\n this.transcribeResolve?.(msg.data as string);\n this.transcribeResolve = null;\n break;\n case 'error': {\n const errMsg = msg.data as string;\n this.emit('error', errMsg);\n // Reject model load if still pending\n if (this.modelReadyReject) {\n this.modelReadyReject(new Error(errMsg));\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n }\n // Resolve transcribe with empty string on error\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n break;\n }\n }\n }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/\n Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n /\n/* Delay (ms) before the first correction fires after recording starts. /\nconst INITIAL_CORRECTION_DELAY_MS = 1_000;\n\nexport class CorrectionOrchestrator {\n private forcedTimer: ReturnType<typeof setInterval> \| null = null;\n private initialTimer: ReturnType<typeof setTimeout> \| 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.\n * Fires a quick initial correction after 1s for early feedback, then\n * switches to the regular forcedInterval cadence from that point. /\n start(): void {\n if (!this.config.enabled) return;\n\n this.lastCorrectionTime = Date.now();\n // One-shot early correction — gives the user immediate Whisper feedback\n // before the Web Speech API has produced its first interim result (~1-2s).\n this.initialTimer = setTimeout(() => {\n this.initialTimer = null;\n this.correctionFn?.();\n this.lastCorrectionTime = Date.now();\n this.startForcedTimer();\n }, INITIAL_CORRECTION_DELAY_MS);\n }\n\n /* Stop the orchestrator (clear all timers). /\n stop(): void {\n if (this.initialTimer) {\n clearTimeout(this.initialTimer);\n this.initialTimer = null;\n }\n this.stopForcedTimer();\n }\n\n /* Called when a speech pause is detected. Triggers correction if cooldown elapsed. /\n onPauseDetected(): void {\n if (!this.config.enabled) return;\n\n const now = Date.now();\n if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n this.triggerCorrection();\n }\n\n /* Force a correction now (resets timer). /\n forceCorrection(): void {\n this.triggerCorrection();\n }\n\n private triggerCorrection(): void {\n this.lastCorrectionTime = Date.now();\n this.correctionFn?.();\n // Reset forced timer after any correction\n this.restartForcedTimer();\n }\n\n private startForcedTimer(): void {\n this.stopForcedTimer();\n this.forcedTimer = setInterval(() => {\n this.triggerCorrection();\n }, this.config.forcedInterval);\n }\n\n private stopForcedTimer(): void {\n if (this.forcedTimer) {\n clearInterval(this.forcedTimer);\n this.forcedTimer = null;\n }\n }\n\n private restartForcedTimer(): void {\n if (this.forcedTimer) {\n this.startForcedTimer();\n }\n }\n}\n","/ ─── Web Speech API types ──────────────────────────────── /\r\n\r\ninterface SpeechRecognitionEvent {\r\n results: SpeechRecognitionResultList;\r\n resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n continuous: boolean;\r\n interimResults: boolean;\r\n lang: string;\r\n onaudiostart: (() => void) \| null;\r\n onresult: ((e: SpeechRecognitionEvent) => void) \| null;\r\n onerror: ((e: SpeechRecognitionErrorEvent) => void) \| null;\r\n onend: (() => void) \| null;\r\n start: () => void;\r\n stop: () => void;\r\n abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor \| null {\r\n if (typeof globalThis === 'undefined') return null;\r\n const w = globalThis as unknown as Record<string, unknown>;\r\n return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor \| null;\r\n}\r\n\r\n/ ─── Language mapping ──────────────────────────────────── /\r\n\r\n/* Map Whisper language codes to BCP-47 locale tags for the Speech API. /\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n en: 'en-US',\r\n english: 'en-US',\r\n zh: 'zh-CN',\r\n chinese: 'zh-CN',\r\n de: 'de-DE',\r\n german: 'de-DE',\r\n es: 'es-ES',\r\n spanish: 'es-ES',\r\n ru: 'ru-RU',\r\n russian: 'ru-RU',\r\n ko: 'ko-KR',\r\n korean: 'ko-KR',\r\n fr: 'fr-FR',\r\n french: 'fr-FR',\r\n ja: 'ja-JP',\r\n japanese: 'ja-JP',\r\n pt: 'pt-BR',\r\n portuguese: 'pt-BR',\r\n tr: 'tr-TR',\r\n turkish: 'tr-TR',\r\n pl: 'pl-PL',\r\n polish: 'pl-PL',\r\n nl: 'nl-NL',\r\n dutch: 'nl-NL',\r\n ar: 'ar-SA',\r\n arabic: 'ar-SA',\r\n sv: 'sv-SE',\r\n swedish: 'sv-SE',\r\n it: 'it-IT',\r\n italian: 'it-IT',\r\n id: 'id-ID',\r\n indonesian: 'id-ID',\r\n hi: 'hi-IN',\r\n hindi: 'hi-IN',\r\n fi: 'fi-FI',\r\n finnish: 'fi-FI',\r\n vi: 'vi-VN',\r\n vietnamese: 'vi-VN',\r\n he: 'he-IL',\r\n hebrew: 'he-IL',\r\n uk: 'uk-UA',\r\n ukrainian: 'uk-UA',\r\n el: 'el-GR',\r\n greek: 'el-GR',\r\n ms: 'ms-MY',\r\n malay: 'ms-MY',\r\n cs: 'cs-CZ',\r\n czech: 'cs-CZ',\r\n ro: 'ro-RO',\r\n romanian: 'ro-RO',\r\n da: 'da-DK',\r\n danish: 'da-DK',\r\n hu: 'hu-HU',\r\n hungarian: 'hu-HU',\r\n no: 'nb-NO',\r\n norwegian: 'nb-NO',\r\n th: 'th-TH',\r\n thai: 'th-TH',\r\n};\r\n\r\n/\r\n Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n /\r\nfunction toBCP47(language: string): string {\r\n if (language.includes('-')) return language;\r\n return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/ ─── SpeechStreamingManager ────────────────────────────── /\r\n\r\n/\r\n Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n \r\n Keep-alive design: the recognition session is never stopped between user\r\n * sessions (only muted). This eliminates the 1-2s Google server handshake\r\n * on every click — start() resolves instantly when the session is warm.\r\n /\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n/* How long to keep recognition running silently after stop() before truly releasing it. /\r\nconst IDLE_TIMEOUT_MS = 30_000;\r\n\r\nexport class SpeechStreamingManager {\r\n private recognition: SpeechRecognitionInstance \| null = null;\r\n private accumulated = '';\r\n private active = false; // user is recording — emit results, trigger onPause on end\r\n private keepingWarm = false; // recognition running silently between user sessions\r\n private currentLang = ''; // BCP-47 of running recognition (for fast-path check)\r\n private receivedResult = false;\r\n private lastFinalIndex = -1; // instance fields so warm-restart resets them cleanly\r\n private lastFinalText = '';\r\n private noResultTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private idleTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private onTranscript: ((text: string) => void) \| null = null;\r\n private onPause: (() => void) \| null = null;\r\n private onError: ((message: string) => void) \| null = null;\r\n private onDebug: ((message: string) => void) \| null = null;\r\n\r\n /* Check if the Web Speech API is available in this environment. /\r\n static isSupported(): boolean {\r\n return getSpeechRecognition() !== null;\r\n }\r\n\r\n /* Set callback for streaming transcript updates (interim + final text). /\r\n setOnTranscript(fn: (text: string) => void): void {\r\n this.onTranscript = fn;\r\n }\r\n\r\n /* Set callback for speech pause detection (Speech API onend). /\r\n setOnPause(fn: () => void): void {\r\n this.onPause = fn;\r\n }\r\n\r\n /* Set callback for errors. /\r\n setOnError(fn: (message: string) => void): void {\r\n this.onError = fn;\r\n }\r\n\r\n /* Set callback for diagnostic debug messages. /\r\n setOnDebug(fn: (message: string) => void): void {\r\n this.onDebug = fn;\r\n }\r\n\r\n private log(message: string): void {\r\n this.onDebug?.(message);\r\n console.warn(message);\r\n }\r\n\r\n /\r\n Pre-warm: start recognition in muted mode so it's ready before the user\r\n * clicks. Call after engine.init() completes. Eliminates startup latency on\r\n * first click by keeping the Google Speech session alive.\r\n /\r\n preWarm(language: string): void {\r\n const SR = getSpeechRecognition();\r\n if (!SR) return;\r\n const bcp47 = toBCP47(language);\r\n if (this.recognition && this.currentLang === bcp47) return; // already warm\r\n this.log(`[SSM] preWarm() — lang: \"${language}\" → \"${bcp47}\"`);\r\n this.keepingWarm = true;\r\n this.active = false;\r\n this.clearIdleTimer();\r\n this.spawnRecognition(language);\r\n }\r\n\r\n /\r\n Start streaming recognition. If recognition is already warm (session\r\n * running from preWarm or a previous session within the idle window),\r\n * activates instantly — no Google handshake. Otherwise cold-starts.\r\n /\r\n start(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition();\r\n if (!SR) {\r\n this.log('[SSM] SpeechRecognition not available in this environment');\r\n return Promise.resolve();\r\n }\r\n\r\n const bcp47 = toBCP47(language);\r\n this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n this.accumulated = '';\r\n this.receivedResult = false;\r\n this.lastFinalIndex = -1;\r\n this.lastFinalText = '';\r\n this.clearIdleTimer();\r\n\r\n // ── Fast path: session already running with the right language ──────────\r\n if (this.recognition && this.currentLang === bcp47) {\r\n this.log('[SSM] start() — warm session, activating immediately');\r\n this.keepingWarm = false;\r\n this.active = true;\r\n // Arm the no-result watchdog for silent-failure detection\r\n this.clearNoResultTimer();\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n return Promise.resolve();\r\n }\r\n\r\n // ── Cold start: spin up a fresh recognition session ─────────────────────\r\n this.keepingWarm = false;\r\n this.active = true;\r\n return this.spawnRecognition(language, skipMicWait);\r\n }\r\n\r\n /\r\n Create and start a new SpeechRecognition instance.\r\n * Used by both preWarm() (active=false) and start() cold path (active=true).\r\n /\r\n private spawnRecognition(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition()!;\r\n const bcp47 = toBCP47(language);\r\n\r\n // Tear down any stale session first\r\n if (this.recognition) {\r\n const old = this.recognition;\r\n this.recognition = null;\r\n try { old.stop(); } catch { / already stopped / }\r\n }\r\n\r\n this.currentLang = bcp47;\r\n\r\n const recognition = new SR();\r\n recognition.continuous = true;\r\n recognition.interimResults = true;\r\n recognition.lang = bcp47;\r\n\r\n // Mic-claim promise (used on cold active starts only)\r\n let micReady = false;\r\n const micClaimPromise = new Promise<void>((resolve) => {\r\n recognition.onaudiostart = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n if (!micReady) { micReady = true; resolve(); }\r\n };\r\n setTimeout(() => {\r\n if (!micReady) {\r\n micReady = true;\r\n this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n resolve();\r\n }\r\n }, 300);\r\n });\r\n\r\n // No-result watchdog — only when actively recording\r\n this.clearNoResultTimer();\r\n if (this.active) {\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n }\r\n\r\n recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.receivedResult = true;\r\n this.clearNoResultTimer();\r\n\r\n // Discard results while muted (keeping warm between user sessions)\r\n if (!this.active) return;\r\n\r\n let final_ = '';\r\n let interim = '';\r\n for (let i = e.resultIndex; i < e.results.length; i++) {\r\n const t = e.results[i][0].transcript;\r\n if (e.results[i].isFinal) {\r\n if (i > this.lastFinalIndex) {\r\n final_ += t;\r\n this.lastFinalIndex = i;\r\n }\r\n } else {\r\n interim += t;\r\n }\r\n }\r\n\r\n this.log(\r\n `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n );\r\n\r\n if (final_ && final_.trim() !== this.lastFinalText) {\r\n this.lastFinalText = final_.trim();\r\n this.accumulated = this.accumulated\r\n ? this.accumulated + ' ' + final_.trim()\r\n : final_.trim();\r\n this.onTranscript?.(this.accumulated);\r\n } else if (interim) {\r\n const trimmed = interim.trimStart();\r\n const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n this.onTranscript?.(full);\r\n }\r\n };\r\n\r\n recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.log(`[SSM] onerror — ${e.error}`);\r\n // Suppress errors while muted — they're not relevant to the user\r\n if (this.active) this.onError?.(e.error);\r\n };\r\n\r\n recognition.onend = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log(\r\n `[SSM] onend — active: ${this.active}, keepingWarm: ${this.keepingWarm}, receivedResult: ${this.receivedResult}`,\r\n );\r\n\r\n if (this.active) {\r\n // User is recording — trigger correction and restart for continued streaming\r\n this.onPause?.();\r\n try {\r\n recognition.start();\r\n this.log('[SSM] restarted after pause');\r\n } catch (err) {\r\n this.log(`[SSM] restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n this.onError?.('Speech recognition failed to restart after pause.');\r\n }\r\n } else if (this.keepingWarm) {\r\n // Between user sessions — restart silently to keep the session alive\r\n try {\r\n recognition.start();\r\n this.log('[SSM] warm restart');\r\n } catch (err) {\r\n this.log(`[SSM] warm restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.keepingWarm = false;\r\n this.currentLang = '';\r\n }\r\n } else {\r\n this.recognition = null;\r\n this.currentLang = '';\r\n }\r\n };\r\n\r\n this.recognition = recognition;\r\n try {\r\n recognition.start();\r\n this.log('[SSM] recognition.start() succeeded');\r\n } catch (err) {\r\n this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n const wasActive = this.active;\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n if (wasActive) {\r\n this.onError?.(\r\n `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n );\r\n }\r\n return Promise.resolve();\r\n }\r\n\r\n // preWarm path or skipMicWait — no need to wait for mic claim\r\n if (!this.active \|\| skipMicWait) {\r\n if (skipMicWait) this.log('[SSM] skipMicWait — warm restart, returning immediately');\r\n return Promise.resolve();\r\n }\r\n\r\n return micClaimPromise;\r\n }\r\n\r\n private clearNoResultTimer(): void {\r\n if (this.noResultTimer) {\r\n clearTimeout(this.noResultTimer);\r\n this.noResultTimer = null;\r\n }\r\n }\r\n\r\n private clearIdleTimer(): void {\r\n if (this.idleTimer) {\r\n clearTimeout(this.idleTimer);\r\n this.idleTimer = null;\r\n }\r\n }\r\n\r\n private startIdleTimer(): void {\r\n this.clearIdleTimer();\r\n this.idleTimer = setTimeout(() => {\r\n this.idleTimer = null;\r\n this.keepingWarm = false;\r\n this.log('[SSM] idle timeout — stopping recognition');\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n try { rec.stop(); } catch { / already stopped / }\r\n }\r\n }, IDLE_TIMEOUT_MS);\r\n }\r\n\r\n /* Stop streaming recognition and return accumulated text.\r\n * Keeps the recognition session alive (muted) for instant restart. /\r\n stop(): string {\r\n this.active = false;\r\n this.keepingWarm = true;\r\n this.clearNoResultTimer();\r\n const result = this.accumulated;\r\n this.accumulated = '';\r\n // Keep recognition running silently; release after idle timeout\r\n this.startIdleTimer();\r\n return result;\r\n }\r\n\r\n /* Abort immediately and release all resources. /\r\n destroy(): void {\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n this.clearIdleTimer();\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n rec.abort();\r\n }\r\n this.accumulated = '';\r\n this.onTranscript = null;\r\n this.onPause = null;\r\n this.onError = null;\r\n this.onDebug = null;\r\n }\r\n}\r\n","import type {\r\n STTConfig,\r\n STTState,\r\n STTEvents,\r\n STTStatus,\r\n ResolvedSTTConfig,\r\n AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, pauseCapture, resumeCapture, snapshotAudio, resampleAudio } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/\r\n Main STT engine — the public API for speech-to-text with Whisper correction.\r\n \r\n Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n /\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n private config: ResolvedSTTConfig;\r\n private workerManager: WorkerManager;\r\n private correctionOrchestrator: CorrectionOrchestrator;\r\n private speechStreaming: SpeechStreamingManager;\r\n private capture: AudioCaptureHandle \| null = null;\r\n private state: STTState;\r\n private workerUrl?: URL;\r\n /* Prevents performCorrection from emitting while stop() is consuming the in-flight result. /\r\n private _stopping = false;\r\n\r\n /\r\n Create a new STT engine instance.\r\n * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n /\r\n constructor(config?: STTConfig, workerUrl?: URL) {\r\n super();\r\n this.config = resolveConfig(config);\r\n this.workerManager = new WorkerManager();\r\n this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n this.speechStreaming = new SpeechStreamingManager();\r\n this.workerUrl = workerUrl;\r\n\r\n this.state = {\r\n status: 'idle',\r\n isModelLoaded: false,\r\n loadProgress: 0,\r\n backend: null,\r\n error: null,\r\n };\r\n\r\n this.correctionOrchestrator.setCorrectionFn(() => {\r\n this.performCorrection();\r\n });\r\n\r\n this.setupWorkerListeners();\r\n this.setupStreamingCallbacks();\r\n }\r\n\r\n /* Initialize the engine: spawn worker and load model. /\r\n async init(): Promise<void> {\r\n this.updateStatus('loading');\r\n this.workerManager.spawn(this.workerUrl);\r\n\r\n try {\r\n await this.workerManager.loadModel(this.config);\r\n this.state.isModelLoaded = true;\r\n this.updateStatus('ready');\r\n // Pre-warm recognition so the first click is instant (no Google handshake delay)\r\n if (this.config.streaming.enabled) {\r\n this.speechStreaming.preWarm(this.config.language);\r\n }\r\n } catch (err) {\r\n this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n this.updateStatus('idle');\r\n throw err;\r\n }\r\n }\r\n\r\n /* Start recording audio and enable correction cycles. /\r\n async start(): Promise<void> {\r\n if (this.state.status !== 'ready') {\r\n throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n }\r\n\r\n try {\r\n // Check if mic is already warm from a previous recording session.\r\n // When warm, skip getUserMedia and the 300ms SR mic-claim wait.\r\n const warmCapture =\r\n this.capture &&\r\n this.capture.stream.getTracks().every((t) => t.readyState === 'live');\r\n\r\n this.emitDebug(\r\n `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\", warm: ${!!warmCapture}`,\r\n );\r\n\r\n if (this.config.streaming.enabled) {\r\n // On warm restart, skip the mic-claim wait — no getUserMedia race.\r\n await this.speechStreaming.start(this.config.language, !!warmCapture);\r\n if (!warmCapture) {\r\n this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n }\r\n }\r\n\r\n if (warmCapture) {\r\n await resumeCapture(this.capture!);\r\n this.emitDebug('[STT] warm mic resumed — skipped getUserMedia');\r\n } else {\r\n // First start or stale capture: full mic init with getUserMedia.\r\n this.capture = await startCapture();\r\n }\r\n\r\n this.updateStatus('recording');\r\n this.correctionOrchestrator.start();\r\n } catch (err) {\r\n this.emitError(\r\n 'MIC_DENIED',\r\n err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n );\r\n }\r\n }\r\n\r\n /* Stop recording, run final transcription, return text.\r\n * Mic and AudioContext stay alive for fast restart — call destroy() to fully release. /\r\n async stop(): Promise<string> {\r\n if (!this.capture) return '';\r\n\r\n // Prevent any in-flight performCorrection from emitting — stop() will own the final emit.\r\n this._stopping = true;\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.stop();\r\n\r\n this.updateStatus('processing');\r\n\r\n // If a correction job is already running, reuse it: Whisper can't be interrupted mid-inference\r\n // anyway, so cancelling just wastes the work and causes a second full run.\r\n if (this.workerManager.isTranscribing) {\r\n try {\r\n // Resample audio and await the in-flight result in parallel.\r\n const [audio, inFlightText] = await Promise.all([\r\n pauseCapture(this.capture),\r\n this.workerManager.awaitCurrentTranscription(),\r\n ]);\r\n this._stopping = false;\r\n\r\n const text = inFlightText.trim();\r\n if (text) {\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n }\r\n\r\n // In-flight returned empty (cancelled or error) — fall through to fresh transcription.\r\n if (audio.length > 0) {\r\n const freshText = await this.workerManager.transcribe(audio);\r\n this.emit('correction', freshText);\r\n this.updateStatus('ready');\r\n return freshText;\r\n }\r\n\r\n this.updateStatus('ready');\r\n return '';\r\n } catch (err) {\r\n this._stopping = false;\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n // No job in-flight — cancel is a no-op, run fresh transcription.\r\n this.workerManager.cancel();\r\n this._stopping = false;\r\n\r\n try {\r\n // Soft pause — keeps stream and AudioContext alive for fast restart.\r\n const audio = await pauseCapture(this.capture);\r\n\r\n if (audio.length === 0) {\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n /* Destroy the engine: terminate worker, release all resources. /\r\n destroy(): void {\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.destroy();\r\n\r\n if (this.capture) {\r\n try {\r\n this.capture._processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n for (const track of this.capture.stream.getTracks()) {\r\n track.stop();\r\n }\r\n this.capture.audioCtx.close().catch(() => {});\r\n this.capture = null;\r\n }\r\n\r\n this.workerManager.destroy();\r\n this.updateStatus('idle');\r\n this.removeAllListeners();\r\n }\r\n\r\n /* Get current engine state. /\r\n getState(): Readonly<STTState> {\r\n return { ...this.state };\r\n }\r\n\r\n /* Notify the correction orchestrator of a speech pause. */\r\n notifyPause(): void {\r\n this.correctionOrchestrator.onPauseDetected();\r\n }\r\n\r\n private async performCorrection(): Promise<void> {\r\n if (!this.capture \|\| !this.state.isModelLoaded) return;\r\n\r\n this.workerManager.cancel();\r\n\r\n try {\r\n const samples = snapshotAudio(this.capture);\r\n const nativeSr = this.capture.audioCtx.sampleRate;\r\n const audio = await resampleAudio(samples, nativeSr);\r\n\r\n if (audio.length === 0) return;\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n if (text.trim() && this.capture && !this._stopping) {\r\n this.emit('correction', text);\r\n }\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Correction transcription failed.',\r\n );\r\n // Recording continues — error is non-fatal\r\n }\r\n }\r\n\r\n private setupStreamingCallbacks(): void {\r\n this.speechStreaming.setOnDebug((message) => {\r\n this.emit('debug', message);\r\n });\r\n\r\n this.speechStreaming.setOnTranscript((text) => {\r\n this.emitDebug(`[STT] transcript callback — \"${text}\"`);\r\n this.emit('transcript', text);\r\n });\r\n\r\n this.speechStreaming.setOnPause(() => {\r\n this.emitDebug('[STT] pause callback — triggering correction');\r\n this.correctionOrchestrator.onPauseDetected();\r\n });\r\n\r\n this.speechStreaming.setOnError((message) => {\r\n this.emitDebug(`[STT] streaming error — \"${message}\"`);\r\n this.emitError('STREAMING_ERROR', message);\r\n });\r\n }\r\n\r\n private setupWorkerListeners(): void {\r\n this.workerManager.on('progress', (percent) => {\r\n this.state.loadProgress = percent;\r\n this.emit('status', { ...this.state });\r\n });\r\n\r\n this.workerManager.on('error', (message) => {\r\n this.emitError('WORKER_ERROR', message);\r\n });\r\n }\r\n\r\n private updateStatus(status: STTStatus): void {\r\n this.state.status = status;\r\n this.state.error = null;\r\n this.emit('status', { ...this.state });\r\n }\r\n\r\n private emitError(code: string, message: string): void {\r\n this.state.error = message;\r\n this.emit('error', { code, message });\r\n }\r\n\r\n private emitDebug(message: string): void {\r\n console.warn(message);\r\n this.emit('debug', message);\r\n 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1	+ {"version":3,"sources":["../src/types.ts","../src/event-emitter.ts","../src/audio-capture.ts","../src/worker-manager.ts","../src/correction-orchestrator.ts","../src/speech-streaming.ts","../src/stt-engine.ts"],"sourcesContent":["/** Supported Whisper model sizes. /\r\nexport type STTModelSize = 'tiny' \| 'base' \| 'small' \| 'medium';\r\n\r\n/* Supported compute backends. /\r\nexport type STTBackend = 'webgpu' \| 'wasm' \| 'auto';\r\n\r\n/* Engine lifecycle states. /\r\nexport type STTStatus = 'idle' \| 'loading' \| 'ready' \| 'recording' \| 'processing';\r\n\r\n/* Supported correction engine providers. /\r\nexport type STTCorrectionProvider = 'whisper';\r\n\r\n/* Supported real-time streaming providers. /\r\nexport type STTStreamingProvider = 'web-speech-api';\r\n\r\n/* Correction engine configuration. /\r\nexport interface STTCorrectionConfig {\r\n /* Enable mid-recording correction. Default: true /\r\n enabled?: boolean;\r\n /* Correction engine provider. Default: 'whisper' /\r\n provider?: STTCorrectionProvider;\r\n /* Silence duration (ms) before triggering correction. Default: 1000 /\r\n pauseThreshold?: number;\r\n /* Maximum interval (ms) between forced corrections. Default: 3000 /\r\n forcedInterval?: number;\r\n}\r\n\r\n/* Real-time streaming preview configuration. /\r\nexport interface STTStreamingConfig {\r\n /* Enable real-time streaming transcript. Default: true /\r\n enabled?: boolean;\r\n /* Streaming provider. Default: 'web-speech-api' /\r\n provider?: STTStreamingProvider;\r\n}\r\n\r\n/* Audio chunking configuration for long-form audio. /\r\nexport interface STTChunkingConfig {\r\n /* Chunk length in seconds for Whisper processing. Default: 30 /\r\n chunkLengthS?: number;\r\n /* Stride length in seconds for overlapping chunks. Default: 5 /\r\n strideLengthS?: number;\r\n}\r\n\r\n/* Full engine configuration. All fields optional — sensible defaults applied. /\r\nexport interface STTConfig {\r\n /* Whisper model size. Default: 'tiny' /\r\n model?: STTModelSize;\r\n /* Compute backend preference. Default: 'auto' (WebGPU with WASM fallback) /\r\n backend?: STTBackend;\r\n /* Transcription language. Default: 'en' /\r\n language?: string;\r\n /* Model quantization dtype. Default: 'q4' /\r\n dtype?: string;\r\n /* Mid-recording correction settings. /\r\n correction?: STTCorrectionConfig;\r\n /* Audio chunking settings for long-form audio. /\r\n chunking?: STTChunkingConfig;\r\n /* Web Speech API streaming preview settings. /\r\n streaming?: STTStreamingConfig;\r\n}\r\n\r\n/* Resolved configuration with all defaults applied. /\r\nexport interface ResolvedSTTConfig {\r\n model: STTModelSize;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n correction: Required<STTCorrectionConfig>;\r\n chunking: Required<STTChunkingConfig>;\r\n streaming: Required<STTStreamingConfig>;\r\n}\r\n\r\n/* Engine state exposed to consumers via status events. /\r\nexport interface STTState {\r\n status: STTStatus;\r\n isModelLoaded: boolean;\r\n /* Model download progress (0–100). /\r\n loadProgress: number;\r\n /* Active compute backend, or null if not yet determined. /\r\n backend: 'webgpu' \| 'wasm' \| null;\r\n error: string \| null;\r\n}\r\n\r\n/* Structured error emitted via the 'error' event. /\r\nexport interface STTError {\r\n code: string;\r\n message: string;\r\n}\r\n\r\n/* Event map for the typed event emitter. /\r\nexport type STTEvents = {\r\n /* Streaming interim text during recording. /\r\n transcript: (text: string) => void;\r\n /* Whisper-corrected text replacing interim text. /\r\n correction: (text: string) => void;\r\n /* Actionable error (mic denied, model fail, transcription fail). /\r\n error: (error: STTError) => void;\r\n /* Engine state change. /\r\n status: (state: STTState) => void;\r\n /* Diagnostic log for debugging (subscribe to capture all internal events). /\r\n debug: (message: string) => void;\r\n};\r\n\r\n/* Handle returned by audio capture — used internally. /\r\nexport interface AudioCaptureHandle {\r\n audioCtx: AudioContext;\r\n stream: MediaStream;\r\n samples: Float32Array[];\r\n /* Retain reference to prevent GC from stopping audio processing. /\r\n _processor: ScriptProcessorNode;\r\n /* Source node for disconnect/reconnect on pause/resume. /\r\n _source: MediaStreamAudioSourceNode;\r\n /* Gain node (silent) to prevent mic playback. /\r\n _silencer: GainNode;\r\n}\r\n\r\n/* Message sent from main thread to Whisper worker. /\r\nexport interface WorkerMessage {\r\n type: 'load' \| 'transcribe' \| 'cancel';\r\n audio?: Float32Array;\r\n config?: {\r\n model: string;\r\n backend: STTBackend;\r\n language: string;\r\n dtype: string;\r\n chunkLengthS: number;\r\n strideLengthS: number;\r\n };\r\n}\r\n\r\n/* Response sent from Whisper worker to main thread. /\r\nexport interface WorkerResponse {\r\n type: 'progress' \| 'ready' \| 'result' \| 'error';\r\n data?: unknown;\r\n}\r\n\r\n/* Default configuration values. /\r\nexport const DEFAULT_STT_CONFIG: ResolvedSTTConfig = {\r\n model: 'tiny',\r\n backend: 'auto',\r\n language: 'en',\r\n dtype: 'q4',\r\n correction: {\r\n enabled: true,\r\n provider: 'whisper',\r\n pauseThreshold: 1_000,\r\n forcedInterval: 3_000,\r\n },\r\n chunking: {\r\n chunkLengthS: 30,\r\n strideLengthS: 5,\r\n },\r\n streaming: {\r\n enabled: true,\r\n provider: 'web-speech-api',\r\n },\r\n};\r\n\r\n/* Merge user config with defaults to produce resolved config. /\r\nexport function resolveConfig(config?: STTConfig): ResolvedSTTConfig {\r\n return {\r\n model: config?.model ?? DEFAULT_STT_CONFIG.model,\r\n backend: config?.backend ?? DEFAULT_STT_CONFIG.backend,\r\n language: config?.language ?? DEFAULT_STT_CONFIG.language,\r\n dtype: config?.dtype ?? DEFAULT_STT_CONFIG.dtype,\r\n correction: {\r\n enabled: config?.correction?.enabled ?? DEFAULT_STT_CONFIG.correction.enabled,\r\n provider: config?.correction?.provider ?? DEFAULT_STT_CONFIG.correction.provider,\r\n pauseThreshold:\r\n config?.correction?.pauseThreshold ?? DEFAULT_STT_CONFIG.correction.pauseThreshold,\r\n forcedInterval:\r\n config?.correction?.forcedInterval ?? DEFAULT_STT_CONFIG.correction.forcedInterval,\r\n },\r\n chunking: {\r\n chunkLengthS: config?.chunking?.chunkLengthS ?? DEFAULT_STT_CONFIG.chunking.chunkLengthS,\r\n strideLengthS: config?.chunking?.strideLengthS ?? DEFAULT_STT_CONFIG.chunking.strideLengthS,\r\n },\r\n streaming: {\r\n enabled: config?.streaming?.enabled ?? DEFAULT_STT_CONFIG.streaming.enabled,\r\n provider: config?.streaming?.provider ?? DEFAULT_STT_CONFIG.streaming.provider,\r\n },\r\n };\r\n}\r\n","/\n A generic, typed event emitter.\n \n Type parameter `T` is a map of event names to listener signatures,\n * giving consumers compile-time safety on event names and callback args.\n /\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nexport class TypedEventEmitter<T extends Record<string, (...args: any[]) => void>> {\n private listeners = new Map<keyof T, Set<T[keyof T]>>();\n\n /* Subscribe to an event. /\n on<K extends keyof T>(event: K, listener: T[K]): void {\n let set = this.listeners.get(event);\n if (!set) {\n set = new Set();\n this.listeners.set(event, set);\n }\n set.add(listener as T[keyof T]);\n }\n\n /* Unsubscribe a specific listener. No-op if not registered. /\n off<K extends keyof T>(event: K, listener: T[K]): void {\n this.listeners.get(event)?.delete(listener as T[keyof T]);\n }\n\n /* Emit an event, calling all registered listeners in insertion order. /\n emit<K extends keyof T>(event: K, ...args: Parameters<T[K]>): void {\n const set = this.listeners.get(event);\n if (!set) return;\n for (const listener of set) {\n (listener as (...a: Parameters<T[K]>) => void)(...args);\n }\n }\n\n /* Remove all listeners, optionally for a single event. /\n removeAllListeners(event?: keyof T): void {\n if (event !== undefined) {\n this.listeners.delete(event);\n } else {\n this.listeners.clear();\n }\n }\n}\n","import type { AudioCaptureHandle } from './types.js';\r\n\r\nconst TARGET_SAMPLE_RATE = 16_000;\r\n\r\n/\r\n Start capturing raw PCM audio from the microphone.\r\n * Uses ScriptProcessorNode to collect Float32Array samples directly.\r\n /\r\nexport async function startCapture(): Promise<AudioCaptureHandle> {\r\n const stream = await navigator.mediaDevices.getUserMedia({\r\n audio: { channelCount: 1 },\r\n });\r\n const audioCtx = new AudioContext();\r\n\r\n // Chrome may suspend AudioContext — must resume within user gesture\r\n if (audioCtx.state === 'suspended') {\r\n await audioCtx.resume();\r\n }\r\n\r\n const source = audioCtx.createMediaStreamSource(stream);\r\n const samples: Float32Array[] = [];\r\n\r\n const processor = audioCtx.createScriptProcessor(4096, 1, 1);\r\n processor.onaudioprocess = (e: AudioProcessingEvent) => {\r\n samples.push(new Float32Array(e.inputBuffer.getChannelData(0)));\r\n };\r\n\r\n // Connect through a silent gain node so mic audio doesn't play back\r\n const silencer = audioCtx.createGain();\r\n silencer.gain.value = 0;\r\n source.connect(processor);\r\n processor.connect(silencer);\r\n silencer.connect(audioCtx.destination);\r\n\r\n return { audioCtx, stream, samples, _processor: processor, _source: source, _silencer: silencer };\r\n}\r\n\r\n/\r\n Pause capture without releasing mic or AudioContext.\r\n * Disconnects the audio source so no new samples are collected.\r\n * Returns resampled audio from the recording period.\r\n * Call resumeCapture() to start collecting again.\r\n /\r\nexport async function pauseCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n capture._source.disconnect();\r\n const currentSamples = [...capture.samples];\r\n capture.samples.length = 0;\r\n return resampleAudio(currentSamples, capture.audioCtx.sampleRate);\r\n}\r\n\r\n/\r\n Resume a paused capture. Reconnects the audio source to the processor.\r\n * AudioContext is resumed if suspended.\r\n /\r\nexport async function resumeCapture(capture: AudioCaptureHandle): Promise<void> {\r\n if (capture.audioCtx.state === 'suspended') {\r\n await capture.audioCtx.resume();\r\n }\r\n capture._source.connect(capture._processor);\r\n}\r\n\r\n/\r\n Copy current audio buffer without stopping capture.\r\n * Returns a shallow copy of the samples array (each chunk is shared, not cloned).\r\n /\r\nexport function snapshotAudio(capture: AudioCaptureHandle): Float32Array[] {\r\n return [...capture.samples];\r\n}\r\n\r\n/\r\n Trim the audio buffer to keep only the most recent `keepSeconds` of audio.\r\n * Call this after each mid-recording correction to prevent unbounded buffer growth.\r\n * Without trimming, a 15-minute session accumulates ~158 MB and makes each\r\n * correction send 15 minutes of audio to Whisper.\r\n /\r\nexport function trimAudioBuffer(capture: AudioCaptureHandle, keepSeconds: number): void {\r\n const samplesPerChunk = 4096; // matches ScriptProcessorNode buffer size in startCapture()\r\n const chunksToKeep = Math.ceil((keepSeconds capture.audioCtx.sampleRate) / samplesPerChunk);\r\n if (capture.samples.length > chunksToKeep) {\r\n capture.samples.splice(0, capture.samples.length - chunksToKeep);\r\n }\r\n}\r\n\r\n/*\r\n Concatenate sample chunks and resample to 16kHz for Whisper.\r\n /\r\nexport async function resampleAudio(\r\n samples: Float32Array[],\r\n nativeSr: number,\r\n): Promise<Float32Array> {\r\n const totalLength = samples.reduce((sum, s) => sum + s.length, 0);\r\n if (totalLength === 0) return new Float32Array(0);\r\n\r\n const fullAudio = new Float32Array(totalLength);\r\n let offset = 0;\r\n for (const s of samples) {\r\n fullAudio.set(s, offset);\r\n offset += s.length;\r\n }\r\n\r\n if (nativeSr === TARGET_SAMPLE_RATE) return fullAudio;\r\n\r\n const duration = fullAudio.length / nativeSr;\r\n const outLength = Math.round(duration TARGET_SAMPLE_RATE);\r\n const offline = new OfflineAudioContext(1, outLength, TARGET_SAMPLE_RATE);\r\n const buffer = offline.createBuffer(1, fullAudio.length, nativeSr);\r\n buffer.getChannelData(0).set(fullAudio);\r\n const src = offline.createBufferSource();\r\n src.buffer = buffer;\r\n src.connect(offline.destination);\r\n src.start(0);\r\n const resampled = await offline.startRendering();\r\n return resampled.getChannelData(0);\r\n}\r\n\r\n/*\r\n Stop capturing and return resampled audio at 16kHz.\r\n /\r\nexport async function stopCapture(capture: AudioCaptureHandle): Promise<Float32Array> {\r\n const { audioCtx, stream, samples, _processor } = capture;\r\n\r\n // Disconnect processor to stop capturing\r\n try {\r\n _processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n\r\n // Stop microphone tracks\r\n for (const track of stream.getTracks()) {\r\n track.stop();\r\n }\r\n\r\n const nativeSr = audioCtx.sampleRate;\r\n await audioCtx.close();\r\n\r\n return resampleAudio(samples, nativeSr);\r\n}\r\n","import type { ResolvedSTTConfig, WorkerResponse } from './types.js';\nimport { TypedEventEmitter } from './event-emitter.js';\n\n/* Events emitted by the WorkerManager. /\nexport type WorkerManagerEvents = {\n progress: (percent: number) => void;\n ready: () => void;\n result: (text: string) => void;\n error: (message: string) => void;\n};\n\n/\n Manages the Whisper Web Worker lifecycle.\n * Provides typed message passing and a promise-based transcription API.\n /\nexport class WorkerManager extends TypedEventEmitter<WorkerManagerEvents> {\n private worker: Worker \| null = null;\n private transcribeResolve: ((text: string) => void) \| null = null;\n private currentTranscribePromise: Promise<string> \| null = null;\n private modelReadyResolve: (() => void) \| null = null;\n private modelReadyReject: ((err: Error) => void) \| null = null;\n\n /* True while a transcription job is running in the worker. /\n get isTranscribing(): boolean {\n return this.transcribeResolve !== null;\n }\n\n /* Await the current in-flight transcription without starting a new one. /\n awaitCurrentTranscription(): Promise<string> {\n return this.currentTranscribePromise ?? Promise.resolve('');\n }\n\n /* Spawn the Web Worker. Must be called before loadModel/transcribe. /\n spawn(workerUrl?: URL): void {\n if (this.worker) return;\n\n const url = workerUrl ?? new URL('./whisper-worker.js', import.meta.url);\n\n this.worker = new Worker(url, { type: 'module' });\n this.worker.onmessage = (e: MessageEvent<WorkerResponse>) => {\n this.handleMessage(e.data);\n };\n this.worker.onerror = (e: ErrorEvent) => {\n this.emit('error', e.message ?? 'Worker error');\n };\n }\n\n /* Load the Whisper model in the worker. Resolves when ready. /\n async loadModel(config: ResolvedSTTConfig): Promise<void> {\n if (!this.worker) throw new Error('Worker not spawned');\n\n return new Promise<void>((resolve, reject) => {\n this.modelReadyResolve = resolve;\n this.modelReadyReject = reject;\n this.worker!.postMessage({\n type: 'load',\n config: {\n model: config.model,\n backend: config.backend,\n language: config.language,\n dtype: config.dtype,\n chunkLengthS: config.chunking.chunkLengthS,\n strideLengthS: config.chunking.strideLengthS,\n },\n });\n });\n }\n\n /* Send audio to the worker for transcription. Resolves with text. /\n async transcribe(audio: Float32Array): Promise<string> {\n if (!this.worker) throw new Error('Worker not spawned');\n if (audio.length === 0) return '';\n\n this.currentTranscribePromise = new Promise<string>((resolve) => {\n this.transcribeResolve = resolve;\n this.worker!.postMessage({ type: 'transcribe', audio }, [audio.buffer]);\n });\n return this.currentTranscribePromise;\n }\n\n /* Cancel any in-flight transcription. /\n cancel(): void {\n this.worker?.postMessage({ type: 'cancel' });\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n }\n\n /* Terminate the worker and release resources. /\n destroy(): void {\n this.cancel();\n this.worker?.terminate();\n this.worker = null;\n this.removeAllListeners();\n }\n\n private handleMessage(msg: WorkerResponse): void {\n switch (msg.type) {\n case 'progress':\n this.emit('progress', msg.data as number);\n break;\n case 'ready':\n this.emit('ready');\n this.modelReadyResolve?.();\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n break;\n case 'result':\n this.emit('result', msg.data as string);\n this.transcribeResolve?.(msg.data as string);\n this.transcribeResolve = null;\n break;\n case 'error': {\n const errMsg = msg.data as string;\n this.emit('error', errMsg);\n // Reject model load if still pending\n if (this.modelReadyReject) {\n this.modelReadyReject(new Error(errMsg));\n this.modelReadyResolve = null;\n this.modelReadyReject = null;\n }\n // Resolve transcribe with empty string on error\n if (this.transcribeResolve) {\n this.transcribeResolve('');\n this.transcribeResolve = null;\n }\n break;\n }\n }\n }\n}\n","import type { ResolvedSTTConfig } from './types.js';\n\n/\n Manages mid-recording correction timing.\n * Two triggers: pause detection and forced interval.\n /\n/* Delay (ms) before the first correction fires after recording starts. /\nconst INITIAL_CORRECTION_DELAY_MS = 1_000;\n\nexport class CorrectionOrchestrator {\n private forcedTimer: ReturnType<typeof setInterval> \| null = null;\n private initialTimer: ReturnType<typeof setTimeout> \| 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.\n * Fires a quick initial correction after 1s for early feedback, then\n * switches to the regular forcedInterval cadence from that point. /\n start(): void {\n if (!this.config.enabled) return;\n\n this.lastCorrectionTime = Date.now();\n // One-shot early correction — gives the user immediate Whisper feedback\n // before the Web Speech API has produced its first interim result (~1-2s).\n this.initialTimer = setTimeout(() => {\n this.initialTimer = null;\n this.correctionFn?.();\n this.lastCorrectionTime = Date.now();\n this.startForcedTimer();\n }, INITIAL_CORRECTION_DELAY_MS);\n }\n\n /* Stop the orchestrator (clear all timers). /\n stop(): void {\n if (this.initialTimer) {\n clearTimeout(this.initialTimer);\n this.initialTimer = null;\n }\n this.stopForcedTimer();\n }\n\n /* Called when a speech pause is detected. Triggers correction if cooldown elapsed. /\n onPauseDetected(): void {\n if (!this.config.enabled) return;\n\n const now = Date.now();\n if (now - this.lastCorrectionTime < this.config.pauseThreshold) return;\n\n this.triggerCorrection();\n }\n\n /* Force a correction now (resets timer). /\n forceCorrection(): void {\n this.triggerCorrection();\n }\n\n private triggerCorrection(): void {\n this.lastCorrectionTime = Date.now();\n this.correctionFn?.();\n // Reset forced timer after any correction\n this.restartForcedTimer();\n }\n\n private startForcedTimer(): void {\n this.stopForcedTimer();\n this.forcedTimer = setInterval(() => {\n this.triggerCorrection();\n }, this.config.forcedInterval);\n }\n\n private stopForcedTimer(): void {\n if (this.forcedTimer) {\n clearInterval(this.forcedTimer);\n this.forcedTimer = null;\n }\n }\n\n private restartForcedTimer(): void {\n if (this.forcedTimer) {\n this.startForcedTimer();\n }\n }\n}\n","/ ─── Web Speech API types ──────────────────────────────── /\r\n\r\ninterface SpeechRecognitionEvent {\r\n results: SpeechRecognitionResultList;\r\n resultIndex: number;\r\n}\r\n\r\ninterface SpeechRecognitionErrorEvent {\r\n error: string;\r\n}\r\n\r\ninterface SpeechRecognitionInstance {\r\n continuous: boolean;\r\n interimResults: boolean;\r\n lang: string;\r\n onaudiostart: (() => void) \| null;\r\n onresult: ((e: SpeechRecognitionEvent) => void) \| null;\r\n onerror: ((e: SpeechRecognitionErrorEvent) => void) \| null;\r\n onend: (() => void) \| null;\r\n start: () => void;\r\n stop: () => void;\r\n abort: () => void;\r\n}\r\n\r\ntype SpeechRecognitionCtor = new () => SpeechRecognitionInstance;\r\n\r\nfunction getSpeechRecognition(): SpeechRecognitionCtor \| null {\r\n if (typeof globalThis === 'undefined') return null;\r\n const w = globalThis as unknown as Record<string, unknown>;\r\n return (w.SpeechRecognition ?? w.webkitSpeechRecognition ?? null) as SpeechRecognitionCtor \| null;\r\n}\r\n\r\n/ ─── Language mapping ──────────────────────────────────── /\r\n\r\n/* Map Whisper language codes to BCP-47 locale tags for the Speech API. /\r\nconst WHISPER_TO_BCP47: Record<string, string> = {\r\n en: 'en-US',\r\n english: 'en-US',\r\n zh: 'zh-CN',\r\n chinese: 'zh-CN',\r\n de: 'de-DE',\r\n german: 'de-DE',\r\n es: 'es-ES',\r\n spanish: 'es-ES',\r\n ru: 'ru-RU',\r\n russian: 'ru-RU',\r\n ko: 'ko-KR',\r\n korean: 'ko-KR',\r\n fr: 'fr-FR',\r\n french: 'fr-FR',\r\n ja: 'ja-JP',\r\n japanese: 'ja-JP',\r\n pt: 'pt-BR',\r\n portuguese: 'pt-BR',\r\n tr: 'tr-TR',\r\n turkish: 'tr-TR',\r\n pl: 'pl-PL',\r\n polish: 'pl-PL',\r\n nl: 'nl-NL',\r\n dutch: 'nl-NL',\r\n ar: 'ar-SA',\r\n arabic: 'ar-SA',\r\n sv: 'sv-SE',\r\n swedish: 'sv-SE',\r\n it: 'it-IT',\r\n italian: 'it-IT',\r\n id: 'id-ID',\r\n indonesian: 'id-ID',\r\n hi: 'hi-IN',\r\n hindi: 'hi-IN',\r\n fi: 'fi-FI',\r\n finnish: 'fi-FI',\r\n vi: 'vi-VN',\r\n vietnamese: 'vi-VN',\r\n he: 'he-IL',\r\n hebrew: 'he-IL',\r\n uk: 'uk-UA',\r\n ukrainian: 'uk-UA',\r\n el: 'el-GR',\r\n greek: 'el-GR',\r\n ms: 'ms-MY',\r\n malay: 'ms-MY',\r\n cs: 'cs-CZ',\r\n czech: 'cs-CZ',\r\n ro: 'ro-RO',\r\n romanian: 'ro-RO',\r\n da: 'da-DK',\r\n danish: 'da-DK',\r\n hu: 'hu-HU',\r\n hungarian: 'hu-HU',\r\n no: 'nb-NO',\r\n norwegian: 'nb-NO',\r\n th: 'th-TH',\r\n thai: 'th-TH',\r\n};\r\n\r\n/\r\n Convert a Whisper language code to a BCP-47 locale tag for the Speech API.\r\n * Already-BCP-47 codes (containing '-') pass through unchanged.\r\n /\r\nfunction toBCP47(language: string): string {\r\n if (language.includes('-')) return language;\r\n return WHISPER_TO_BCP47[language.toLowerCase()] ?? language;\r\n}\r\n\r\n/ ─── SpeechStreamingManager ────────────────────────────── /\r\n\r\n/\r\n Manages Web Speech API for real-time streaming transcript preview.\r\n * Provides word-by-word interim text while Whisper handles corrections.\r\n \r\n Keep-alive design: the recognition session is never stopped between user\r\n * sessions (only muted). This eliminates the 1-2s Google server handshake\r\n * on every click — start() resolves instantly when the session is warm.\r\n /\r\nconst NO_RESULT_TIMEOUT_MS = 5_000;\r\n/* How long to keep recognition running silently after stop() before truly releasing it. /\r\nconst IDLE_TIMEOUT_MS = 30_000;\r\n\r\nexport class SpeechStreamingManager {\r\n private recognition: SpeechRecognitionInstance \| null = null;\r\n private accumulated = '';\r\n private active = false; // user is recording — emit results, trigger onPause on end\r\n private keepingWarm = false; // recognition running silently between user sessions\r\n private currentLang = ''; // BCP-47 of running recognition (for fast-path check)\r\n private receivedResult = false;\r\n private lastFinalIndex = -1; // instance fields so warm-restart resets them cleanly\r\n private lastFinalText = '';\r\n private noResultTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private idleTimer: ReturnType<typeof setTimeout> \| null = null;\r\n private onTranscript: ((text: string) => void) \| null = null;\r\n private onPause: (() => void) \| null = null;\r\n private onError: ((message: string) => void) \| null = null;\r\n private onDebug: ((message: string) => void) \| null = null;\r\n\r\n /* Check if the Web Speech API is available in this environment. /\r\n static isSupported(): boolean {\r\n return getSpeechRecognition() !== null;\r\n }\r\n\r\n /* Set callback for streaming transcript updates (interim + final text). /\r\n setOnTranscript(fn: (text: string) => void): void {\r\n this.onTranscript = fn;\r\n }\r\n\r\n /* Set callback for speech pause detection (Speech API onend). /\r\n setOnPause(fn: () => void): void {\r\n this.onPause = fn;\r\n }\r\n\r\n /* Set callback for errors. /\r\n setOnError(fn: (message: string) => void): void {\r\n this.onError = fn;\r\n }\r\n\r\n /* Set callback for diagnostic debug messages. /\r\n setOnDebug(fn: (message: string) => void): void {\r\n this.onDebug = fn;\r\n }\r\n\r\n private log(message: string): void {\r\n this.onDebug?.(message);\r\n console.warn(message);\r\n }\r\n\r\n /\r\n Pre-warm: start recognition in muted mode so it's ready before the user\r\n * clicks. Call after engine.init() completes. Eliminates startup latency on\r\n * first click by keeping the Google Speech session alive.\r\n /\r\n preWarm(language: string): void {\r\n const SR = getSpeechRecognition();\r\n if (!SR) return;\r\n const bcp47 = toBCP47(language);\r\n if (this.recognition && this.currentLang === bcp47) return; // already warm\r\n this.log(`[SSM] preWarm() — lang: \"${language}\" → \"${bcp47}\"`);\r\n this.keepingWarm = true;\r\n this.active = false;\r\n this.clearIdleTimer();\r\n this.spawnRecognition(language);\r\n }\r\n\r\n /\r\n Start streaming recognition. If recognition is already warm (session\r\n * running from preWarm or a previous session within the idle window),\r\n * activates instantly — no Google handshake. Otherwise cold-starts.\r\n /\r\n start(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition();\r\n if (!SR) {\r\n this.log('[SSM] SpeechRecognition not available in this environment');\r\n return Promise.resolve();\r\n }\r\n\r\n const bcp47 = toBCP47(language);\r\n this.log(`[SSM] start() — lang: \"${language}\" → \"${bcp47}\"`);\r\n\r\n this.accumulated = '';\r\n this.receivedResult = false;\r\n this.lastFinalIndex = -1;\r\n this.lastFinalText = '';\r\n this.clearIdleTimer();\r\n\r\n // ── Fast path: session already running with the right language ──────────\r\n if (this.recognition && this.currentLang === bcp47) {\r\n this.log('[SSM] start() — warm session, activating immediately');\r\n this.keepingWarm = false;\r\n this.active = true;\r\n // Arm the no-result watchdog for silent-failure detection\r\n this.clearNoResultTimer();\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n return Promise.resolve();\r\n }\r\n\r\n // ── Cold start: spin up a fresh recognition session ─────────────────────\r\n this.keepingWarm = false;\r\n this.active = true;\r\n return this.spawnRecognition(language, skipMicWait);\r\n }\r\n\r\n /\r\n Create and start a new SpeechRecognition instance.\r\n * Used by both preWarm() (active=false) and start() cold path (active=true).\r\n /\r\n private spawnRecognition(language: string, skipMicWait = false): Promise<void> {\r\n const SR = getSpeechRecognition()!;\r\n const bcp47 = toBCP47(language);\r\n\r\n // Tear down any stale session first\r\n if (this.recognition) {\r\n const old = this.recognition;\r\n this.recognition = null;\r\n try { old.stop(); } catch { / already stopped / }\r\n }\r\n\r\n this.currentLang = bcp47;\r\n\r\n const recognition = new SR();\r\n recognition.continuous = true;\r\n recognition.interimResults = true;\r\n recognition.lang = bcp47;\r\n\r\n // Mic-claim promise (used on cold active starts only)\r\n let micReady = false;\r\n const micClaimPromise = new Promise<void>((resolve) => {\r\n recognition.onaudiostart = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log('[SSM] onaudiostart — mic acquired by Speech API');\r\n if (!micReady) { micReady = true; resolve(); }\r\n };\r\n setTimeout(() => {\r\n if (!micReady) {\r\n micReady = true;\r\n this.log('[SSM] mic-claim fallback — proceeding after 300ms');\r\n resolve();\r\n }\r\n }, 300);\r\n });\r\n\r\n // No-result watchdog — only when actively recording\r\n this.clearNoResultTimer();\r\n if (this.active) {\r\n this.noResultTimer = setTimeout(() => {\r\n if (this.active && !this.receivedResult) {\r\n this.log('[SSM] no-result timeout fired — no onresult in 5s');\r\n this.onError?.(\r\n 'Speech streaming started but received no results. ' +\r\n 'Mic may be blocked by another audio capture.',\r\n );\r\n }\r\n }, NO_RESULT_TIMEOUT_MS);\r\n }\r\n\r\n recognition.onresult = (e: SpeechRecognitionEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.receivedResult = true;\r\n this.clearNoResultTimer();\r\n\r\n // Discard results while muted (keeping warm between user sessions)\r\n if (!this.active) return;\r\n\r\n let final_ = '';\r\n let interim = '';\r\n for (let i = e.resultIndex; i < e.results.length; i++) {\r\n const t = e.results[i][0].transcript;\r\n if (e.results[i].isFinal) {\r\n if (i > this.lastFinalIndex) {\r\n final_ += t;\r\n this.lastFinalIndex = i;\r\n }\r\n } else {\r\n interim += t;\r\n }\r\n }\r\n\r\n this.log(\r\n `[SSM] onresult — finals: \"${final_}\", interim: \"${interim}\", accumulated: \"${this.accumulated}\"`,\r\n );\r\n\r\n if (final_ && final_.trim() !== this.lastFinalText) {\r\n this.lastFinalText = final_.trim();\r\n this.accumulated = this.accumulated\r\n ? this.accumulated + ' ' + final_.trim()\r\n : final_.trim();\r\n this.onTranscript?.(this.accumulated);\r\n } else if (interim) {\r\n const trimmed = interim.trimStart();\r\n const full = this.accumulated ? this.accumulated + ' ' + trimmed : trimmed;\r\n this.onTranscript?.(full);\r\n }\r\n };\r\n\r\n recognition.onerror = (e: SpeechRecognitionErrorEvent) => {\r\n if (this.recognition !== recognition) return;\r\n this.log(`[SSM] onerror — ${e.error}`);\r\n // Suppress errors while muted — they're not relevant to the user\r\n if (this.active) this.onError?.(e.error);\r\n };\r\n\r\n recognition.onend = () => {\r\n if (this.recognition !== recognition) return;\r\n this.log(\r\n `[SSM] onend — active: ${this.active}, keepingWarm: ${this.keepingWarm}, receivedResult: ${this.receivedResult}`,\r\n );\r\n\r\n if (this.active) {\r\n // User is recording — trigger correction and restart for continued streaming\r\n this.onPause?.();\r\n try {\r\n recognition.start();\r\n this.log('[SSM] restarted after pause');\r\n } catch (err) {\r\n this.log(`[SSM] restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n this.onError?.('Speech recognition failed to restart after pause.');\r\n }\r\n } else if (this.keepingWarm) {\r\n // Between user sessions — restart silently to keep the session alive\r\n try {\r\n recognition.start();\r\n this.log('[SSM] warm restart');\r\n } catch (err) {\r\n this.log(`[SSM] warm restart THREW: ${err}`);\r\n this.recognition = null;\r\n this.keepingWarm = false;\r\n this.currentLang = '';\r\n }\r\n } else {\r\n this.recognition = null;\r\n this.currentLang = '';\r\n }\r\n };\r\n\r\n this.recognition = recognition;\r\n try {\r\n recognition.start();\r\n this.log('[SSM] recognition.start() succeeded');\r\n } catch (err) {\r\n this.log(`[SSM] recognition.start() THREW: ${err}`);\r\n this.recognition = null;\r\n this.currentLang = '';\r\n const wasActive = this.active;\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n if (wasActive) {\r\n this.onError?.(\r\n `Speech recognition failed to start: ${err instanceof Error ? err.message : String(err)}`,\r\n );\r\n }\r\n return Promise.resolve();\r\n }\r\n\r\n // preWarm path or skipMicWait — no need to wait for mic claim\r\n if (!this.active \|\| skipMicWait) {\r\n if (skipMicWait) this.log('[SSM] skipMicWait — warm restart, returning immediately');\r\n return Promise.resolve();\r\n }\r\n\r\n return micClaimPromise;\r\n }\r\n\r\n private clearNoResultTimer(): void {\r\n if (this.noResultTimer) {\r\n clearTimeout(this.noResultTimer);\r\n this.noResultTimer = null;\r\n }\r\n }\r\n\r\n private clearIdleTimer(): void {\r\n if (this.idleTimer) {\r\n clearTimeout(this.idleTimer);\r\n this.idleTimer = null;\r\n }\r\n }\r\n\r\n private startIdleTimer(): void {\r\n this.clearIdleTimer();\r\n this.idleTimer = setTimeout(() => {\r\n this.idleTimer = null;\r\n this.keepingWarm = false;\r\n this.log('[SSM] idle timeout — stopping recognition');\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n try { rec.stop(); } catch { / already stopped / }\r\n }\r\n }, IDLE_TIMEOUT_MS);\r\n }\r\n\r\n /* Stop streaming recognition and return accumulated text.\r\n * Keeps the recognition session alive (muted) for instant restart. /\r\n stop(): string {\r\n this.active = false;\r\n this.keepingWarm = true;\r\n this.clearNoResultTimer();\r\n const result = this.accumulated;\r\n this.accumulated = '';\r\n // Keep recognition running silently; release after idle timeout\r\n this.startIdleTimer();\r\n return result;\r\n }\r\n\r\n /* Abort immediately and release all resources. /\r\n destroy(): void {\r\n this.active = false;\r\n this.keepingWarm = false;\r\n this.clearNoResultTimer();\r\n this.clearIdleTimer();\r\n if (this.recognition) {\r\n const rec = this.recognition;\r\n this.recognition = null;\r\n this.currentLang = '';\r\n rec.abort();\r\n }\r\n this.accumulated = '';\r\n this.onTranscript = null;\r\n this.onPause = null;\r\n this.onError = null;\r\n this.onDebug = null;\r\n }\r\n}\r\n","import type {\r\n STTConfig,\r\n STTState,\r\n STTEvents,\r\n STTStatus,\r\n ResolvedSTTConfig,\r\n AudioCaptureHandle,\r\n} from './types.js';\r\nimport { resolveConfig } from './types.js';\r\nimport { TypedEventEmitter } from './event-emitter.js';\r\nimport { startCapture, pauseCapture, resumeCapture, snapshotAudio, resampleAudio, trimAudioBuffer } from './audio-capture.js';\r\nimport { WorkerManager } from './worker-manager.js';\r\nimport { CorrectionOrchestrator } from './correction-orchestrator.js';\r\nimport { SpeechStreamingManager } from './speech-streaming.js';\r\n\r\n/\r\n Main STT engine — the public API for speech-to-text with Whisper correction.\r\n \r\n Usage:\r\n * ```typescript\r\n * const engine = new STTEngine({ model: 'tiny' });\r\n * engine.on('transcript', (text) => console.log(text));\r\n * engine.on('correction', (text) => console.log('corrected:', text));\r\n * await engine.init();\r\n * await engine.start();\r\n * const finalText = await engine.stop();\r\n * ```\r\n /\r\nexport class STTEngine extends TypedEventEmitter<STTEvents> {\r\n private config: ResolvedSTTConfig;\r\n private workerManager: WorkerManager;\r\n private correctionOrchestrator: CorrectionOrchestrator;\r\n private speechStreaming: SpeechStreamingManager;\r\n private capture: AudioCaptureHandle \| null = null;\r\n private state: STTState;\r\n private workerUrl?: URL;\r\n /* Prevents performCorrection from emitting while stop() is consuming the in-flight result. /\r\n private _stopping = false;\r\n\r\n /\r\n Create a new STT engine instance.\r\n * @param config - Optional configuration overrides (model, backend, language, etc.).\r\n * @param workerUrl - Optional custom URL for the Whisper Web Worker script.\r\n /\r\n constructor(config?: STTConfig, workerUrl?: URL) {\r\n super();\r\n this.config = resolveConfig(config);\r\n this.workerManager = new WorkerManager();\r\n this.correctionOrchestrator = new CorrectionOrchestrator(this.config.correction);\r\n this.speechStreaming = new SpeechStreamingManager();\r\n this.workerUrl = workerUrl;\r\n\r\n this.state = {\r\n status: 'idle',\r\n isModelLoaded: false,\r\n loadProgress: 0,\r\n backend: null,\r\n error: null,\r\n };\r\n\r\n this.correctionOrchestrator.setCorrectionFn(() => {\r\n this.performCorrection();\r\n });\r\n\r\n this.setupWorkerListeners();\r\n this.setupStreamingCallbacks();\r\n }\r\n\r\n /* Initialize the engine: spawn worker and load model. /\r\n async init(): Promise<void> {\r\n this.updateStatus('loading');\r\n this.workerManager.spawn(this.workerUrl);\r\n\r\n try {\r\n await this.workerManager.loadModel(this.config);\r\n this.state.isModelLoaded = true;\r\n this.updateStatus('ready');\r\n // Pre-warm recognition so the first click is instant (no Google handshake delay)\r\n if (this.config.streaming.enabled) {\r\n this.speechStreaming.preWarm(this.config.language);\r\n }\r\n } catch (err) {\r\n this.emitError('MODEL_LOAD_FAILED', err instanceof Error ? err.message : String(err));\r\n this.updateStatus('idle');\r\n throw err;\r\n }\r\n }\r\n\r\n /* Start recording audio and enable correction cycles. /\r\n async start(): Promise<void> {\r\n if (this.state.status !== 'ready') {\r\n throw new Error(`Cannot start: engine is \"${this.state.status}\", expected \"ready\"`);\r\n }\r\n\r\n try {\r\n // Check if mic is already warm from a previous recording session.\r\n // When warm, skip getUserMedia and the 300ms SR mic-claim wait.\r\n const warmCapture =\r\n this.capture &&\r\n this.capture.stream.getTracks().every((t) => t.readyState === 'live');\r\n\r\n this.emitDebug(\r\n `[STT] start() — streaming: ${this.config.streaming.enabled}, lang: \"${this.config.language}\", warm: ${!!warmCapture}`,\r\n );\r\n\r\n if (this.config.streaming.enabled) {\r\n // On warm restart, skip the mic-claim wait — no getUserMedia race.\r\n await this.speechStreaming.start(this.config.language, !!warmCapture);\r\n if (!warmCapture) {\r\n this.emitDebug('[STT] Speech API mic claim complete — starting getUserMedia');\r\n }\r\n }\r\n\r\n if (warmCapture) {\r\n await resumeCapture(this.capture!);\r\n this.emitDebug('[STT] warm mic resumed — skipped getUserMedia');\r\n } else {\r\n // First start or stale capture: full mic init with getUserMedia.\r\n this.capture = await startCapture();\r\n }\r\n\r\n this.updateStatus('recording');\r\n this.correctionOrchestrator.start();\r\n } catch (err) {\r\n this.emitError(\r\n 'MIC_DENIED',\r\n err instanceof Error ? err.message : 'Microphone access denied. Check browser permissions.',\r\n );\r\n }\r\n }\r\n\r\n /* Stop recording, run final transcription, return text.\r\n * Mic and AudioContext stay alive for fast restart — call destroy() to fully release. /\r\n async stop(): Promise<string> {\r\n if (!this.capture) return '';\r\n\r\n // Prevent any in-flight performCorrection from emitting — stop() will own the final emit.\r\n this._stopping = true;\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.stop();\r\n\r\n this.updateStatus('processing');\r\n\r\n // If a correction job is already running, reuse it: Whisper can't be interrupted mid-inference\r\n // anyway, so cancelling just wastes the work and causes a second full run.\r\n if (this.workerManager.isTranscribing) {\r\n try {\r\n // Resample audio and await the in-flight result in parallel.\r\n const [audio, inFlightText] = await Promise.all([\r\n pauseCapture(this.capture),\r\n this.workerManager.awaitCurrentTranscription(),\r\n ]);\r\n this._stopping = false;\r\n\r\n const text = inFlightText.trim();\r\n if (text) {\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n }\r\n\r\n // In-flight returned empty (cancelled or error) — fall through to fresh transcription.\r\n if (audio.length > 0) {\r\n const freshText = await this.workerManager.transcribe(audio);\r\n this.emit('correction', freshText);\r\n this.updateStatus('ready');\r\n return freshText;\r\n }\r\n\r\n this.updateStatus('ready');\r\n return '';\r\n } catch (err) {\r\n this._stopping = false;\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n // No job in-flight — cancel is a no-op, run fresh transcription.\r\n this.workerManager.cancel();\r\n this._stopping = false;\r\n\r\n try {\r\n // Soft pause — keeps stream and AudioContext alive for fast restart.\r\n const audio = await pauseCapture(this.capture);\r\n\r\n if (audio.length === 0) {\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n this.emit('correction', text);\r\n this.updateStatus('ready');\r\n return text;\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Final transcription failed.',\r\n );\r\n this.updateStatus('ready');\r\n return '';\r\n }\r\n }\r\n\r\n /* Destroy the engine: terminate worker, release all resources. /\r\n destroy(): void {\r\n this.correctionOrchestrator.stop();\r\n this.speechStreaming.destroy();\r\n\r\n if (this.capture) {\r\n try {\r\n this.capture._processor.disconnect();\r\n } catch {\r\n / already disconnected /\r\n }\r\n for (const track of this.capture.stream.getTracks()) {\r\n track.stop();\r\n }\r\n this.capture.audioCtx.close().catch(() => {});\r\n this.capture = null;\r\n }\r\n\r\n this.workerManager.destroy();\r\n this.updateStatus('idle');\r\n this.removeAllListeners();\r\n }\r\n\r\n /* Get current engine state. /\r\n getState(): Readonly<STTState> {\r\n return { ...this.state };\r\n }\r\n\r\n /* Notify the correction orchestrator of a speech pause. */\r\n notifyPause(): void {\r\n this.correctionOrchestrator.onPauseDetected();\r\n }\r\n\r\n private async performCorrection(): Promise<void> {\r\n if (!this.capture \|\| !this.state.isModelLoaded) return;\r\n\r\n this.workerManager.cancel();\r\n\r\n try {\r\n const samples = snapshotAudio(this.capture);\r\n const nativeSr = this.capture.audioCtx.sampleRate;\r\n const audio = await resampleAudio(samples, nativeSr);\r\n\r\n if (audio.length === 0) return;\r\n\r\n const text = await this.workerManager.transcribe(audio);\r\n if (text.trim() && this.capture && !this._stopping) {\r\n this.emit('correction', text);\r\n // Trim old audio — keep last 30s so the buffer doesn't grow unboundedly\r\n // during long sessions. Without this, a 15-min session accumulates ~158 MB\r\n // and makes each correction (and the final stop) process 15 min of audio.\r\n trimAudioBuffer(this.capture, 30);\r\n }\r\n } catch (err) {\r\n this.emitError(\r\n 'TRANSCRIPTION_FAILED',\r\n err instanceof Error ? err.message : 'Correction transcription failed.',\r\n );\r\n // Recording continues — error is non-fatal\r\n }\r\n }\r\n\r\n private setupStreamingCallbacks(): void {\r\n this.speechStreaming.setOnDebug((message) => {\r\n this.emit('debug', message);\r\n });\r\n\r\n this.speechStreaming.setOnTranscript((text) => {\r\n this.emitDebug(`[STT] transcript callback — \"${text}\"`);\r\n this.emit('transcript', text);\r\n });\r\n\r\n this.speechStreaming.setOnPause(() => {\r\n this.emitDebug('[STT] pause callback — triggering correction');\r\n this.correctionOrchestrator.onPauseDetected();\r\n });\r\n\r\n this.speechStreaming.setOnError((message) => {\r\n this.emitDebug(`[STT] streaming error — \"${message}\"`);\r\n this.emitError('STREAMING_ERROR', message);\r\n });\r\n }\r\n\r\n private setupWorkerListeners(): void {\r\n this.workerManager.on('progress', (percent) => {\r\n this.state.loadProgress = percent;\r\n this.emit('status', { ...this.state });\r\n });\r\n\r\n this.workerManager.on('error', (message) => {\r\n this.emitError('WORKER_ERROR', message);\r\n });\r\n }\r\n\r\n private updateStatus(status: STTStatus): void {\r\n this.state.status = status;\r\n this.state.error = null;\r\n this.emit('status', { ...this.state });\r\n }\r\n\r\n private emitError(code: string, message: string): void {\r\n this.state.error = message;\r\n this.emit('error', { code, message });\r\n }\r\n\r\n private emitDebug(message: string): void {\r\n console.warn(message);\r\n this.emit('debug', message);\r\n 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package/package.json CHANGED Viewed

@@ -1,6 +1,6 @@
 {
   "name": "@tekyzinc/stt-component",
-  "version": "0.3.2",
+  "version": "0.3.3",
   "description": "Framework-agnostic speech-to-text with real-time streaming transcription and mid-recording Whisper correction",
   "type": "module",
   "main": "dist/index.cjs",