@tryhamster/gerbil 1.0.0-rc.0 → 1.0.0-rc.10

package/dist/stt-CG_7KB_0.mjs.map

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+{"version":3,"file":"stt-CG_7KB_0.mjs","names":["WHISPER_MODELS: STTModelConfig[]","tfDevice: \"webgpu\" | \"cpu\" | \"wasm\"","audioData: Float32Array","pipelineOptions: any","transcribeResult: TranscribeResult","audioBuffer: Float32Array[]","intervalId: ReturnType<typeof setInterval> | null","e: any"],"sources":["../src/core/stt.ts"],"sourcesContent":["/**\n * Speech-to-Text with Whisper\n *\n * Provides local speech recognition using Whisper ONNX models via transformers.js.\n * Supports multiple model sizes and languages.\n *\n * @example\n * ```ts\n * const stt = new WhisperSTT();\n * await stt.load({ onProgress: (p) => console.log(p.status) });\n *\n * // Transcribe audio (Float32Array at 16kHz)\n * const result = await stt.transcribe(audioData);\n * console.log(result.text);\n *\n * // With timestamps\n * const result = await stt.transcribe(audioData, { timestamps: true });\n * for (const seg of result.segments) {\n *   console.log(`[${seg.start.toFixed(1)}s] ${seg.text}`);\n * }\n * ```\n */\n\nimport type {\n  LoadSTTOptions,\n  ProgressInfo,\n  STTModelConfig,\n  StreamingTranscriptionOptions,\n  StreamingTranscriptionSession,\n  TranscribeOptions,\n  TranscribeResult,\n  TranscribeSegment,\n} from \"./types.js\";\n\n// ============================================\n// Model Registry\n// ============================================\n\n/**\n * Available Whisper models\n * Ordered by size (smallest first)\n */\nexport const WHISPER_MODELS: STTModelConfig[] = [\n  {\n    id: \"whisper-tiny.en\",\n    repo: \"onnx-community/whisper-tiny.en\",\n    description: \"Tiny English-only model, fastest\",\n    size: \"39M\",\n    multilingual: false,\n    languages: [\"en\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-tiny\",\n    repo: \"onnx-community/whisper-tiny\",\n    description: \"Tiny multilingual model\",\n    size: \"39M\",\n    multilingual: true,\n    languages: [\"en\", \"es\", \"fr\", \"de\", \"it\", \"pt\", \"nl\", \"ru\", \"zh\", \"ja\", \"ko\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-base.en\",\n    repo: \"onnx-community/whisper-base.en\",\n    description: \"Base English-only model, good balance\",\n    size: \"74M\",\n    multilingual: false,\n    languages: [\"en\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-base\",\n    repo: \"onnx-community/whisper-base\",\n    description: \"Base multilingual model\",\n    size: \"74M\",\n    multilingual: true,\n    languages: [\"en\", \"es\", \"fr\", \"de\", \"it\", \"pt\", \"nl\", \"ru\", \"zh\", \"ja\", \"ko\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-small.en\",\n    repo: \"onnx-community/whisper-small.en\",\n    description: \"Small English-only model, high quality\",\n    size: \"244M\",\n    multilingual: false,\n    languages: [\"en\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-small\",\n    repo: \"onnx-community/whisper-small\",\n    description: \"Small multilingual model\",\n    size: \"244M\",\n    multilingual: true,\n    languages: [\"en\", \"es\", \"fr\", \"de\", \"it\", \"pt\", \"nl\", \"ru\", \"zh\", \"ja\", \"ko\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-large-v3-turbo\",\n    repo: \"onnx-community/whisper-large-v3-turbo\",\n    description: \"Large Turbo model, 5.4x faster, 80+ languages\",\n    size: \"809M\",\n    multilingual: true,\n    languages: [\n      \"en\",\n      \"es\",\n      \"fr\",\n      \"de\",\n      \"it\",\n      \"pt\",\n      \"nl\",\n      \"ru\",\n      \"zh\",\n      \"ja\",\n      \"ko\",\n      \"ar\",\n      \"hi\",\n      \"vi\",\n      \"th\",\n    ],\n    sampleRate: 16000,\n  },\n];\n\n// Default model\nconst DEFAULT_MODEL = \"whisper-tiny.en\";\n\n// ============================================\n// Audio Utilities\n// ============================================\n\n/**\n * Decode WAV file to Float32Array\n * Handles stereo to mono conversion\n */\nexport function decodeWav(buffer: Uint8Array): { audio: Float32Array; sampleRate: number } {\n  const view = new DataView(buffer.buffer, buffer.byteOffset, buffer.byteLength);\n\n  // Validate RIFF header\n  const riff = String.fromCharCode(buffer[0], buffer[1], buffer[2], buffer[3]);\n  if (riff !== \"RIFF\") {\n    throw new Error(\"Invalid WAV file: missing RIFF header\");\n  }\n\n  // Get format details\n  const numChannels = view.getUint16(22, true);\n  const sampleRate = view.getUint32(24, true);\n  const bitsPerSample = view.getUint16(34, true);\n\n  if (bitsPerSample !== 16) {\n    throw new Error(`Unsupported bit depth: ${bitsPerSample}. Only 16-bit WAV is supported.`);\n  }\n\n  // Find data chunk\n  let dataOffset = 12;\n  while (dataOffset < buffer.length - 8) {\n    const chunkId = String.fromCharCode(\n      buffer[dataOffset],\n      buffer[dataOffset + 1],\n      buffer[dataOffset + 2],\n      buffer[dataOffset + 3],\n    );\n    const chunkSize = view.getUint32(dataOffset + 4, true);\n    if (chunkId === \"data\") {\n      dataOffset += 8;\n      break;\n    }\n    dataOffset += 8 + chunkSize;\n  }\n\n  const dataSize = buffer.length - dataOffset;\n  const bytesPerSample = bitsPerSample / 8;\n  const totalSamples = Math.floor(dataSize / bytesPerSample);\n  const samplesPerChannel = Math.floor(totalSamples / numChannels);\n\n  // Convert to mono Float32Array\n  const audio = new Float32Array(samplesPerChannel);\n\n  for (let i = 0; i < samplesPerChannel; i++) {\n    if (numChannels === 2) {\n      const left = view.getInt16(dataOffset + i * 4, true);\n      const right = view.getInt16(dataOffset + i * 4 + 2, true);\n      audio[i] = (left + right) / 2 / 32768;\n    } else {\n      const sample = view.getInt16(dataOffset + i * 2, true);\n      audio[i] = sample / 32768;\n    }\n  }\n\n  return { audio, sampleRate };\n}\n\n/**\n * Resample audio to target sample rate using linear interpolation\n */\nexport function resampleAudio(audio: Float32Array, fromRate: number, toRate: number): Float32Array {\n  if (fromRate === toRate) return audio;\n\n  const ratio = toRate / fromRate;\n  const newLength = Math.round(audio.length * ratio);\n  const result = new Float32Array(newLength);\n\n  for (let i = 0; i < newLength; i++) {\n    const srcIndex = i / ratio;\n    const floor = Math.floor(srcIndex);\n    const ceil = Math.min(floor + 1, audio.length - 1);\n    const t = srcIndex - floor;\n    result[i] = audio[floor] * (1 - t) + audio[ceil] * t;\n  }\n\n  return result;\n}\n\n// ============================================\n// WhisperSTT Class\n// ============================================\n\n/**\n * Speech-to-Text using Whisper ONNX models\n */\nexport class WhisperSTT {\n  private modelConfig: STTModelConfig;\n  private pipeline: any = null;\n  private loadPromise: Promise<void> | null = null;\n  private _isLoaded = false;\n  private _deviceMode: \"webgpu\" | \"cpu\" = \"cpu\";\n\n  constructor(modelId: string = DEFAULT_MODEL) {\n    const config = WHISPER_MODELS.find((m) => m.id === modelId);\n    if (!config) {\n      const available = WHISPER_MODELS.map((m) => m.id).join(\", \");\n      throw new Error(`Unknown STT model: ${modelId}. Available: ${available}`);\n    }\n    this.modelConfig = config;\n  }\n\n  /**\n   * Check if model is loaded\n   */\n  isLoaded(): boolean {\n    return this._isLoaded;\n  }\n\n  /**\n   * Get model configuration\n   */\n  getModelConfig(): STTModelConfig {\n    return this.modelConfig;\n  }\n\n  /**\n   * Get model info (alias for getModelConfig)\n   */\n  getModelInfo(): STTModelConfig {\n    return this.modelConfig;\n  }\n\n  /**\n   * Get current device mode\n   */\n  getDeviceMode(): \"webgpu\" | \"cpu\" {\n    return this._deviceMode;\n  }\n\n  /**\n   * List available models\n   */\n  static listModels(): STTModelConfig[] {\n    return [...WHISPER_MODELS];\n  }\n\n  /**\n   * Load the STT model\n   */\n  async load(options: LoadSTTOptions = {}): Promise<void> {\n    if (this._isLoaded) return;\n    if (this.loadPromise) {\n      await this.loadPromise;\n      return;\n    }\n\n    this.loadPromise = this._load(options);\n    await this.loadPromise;\n  }\n\n  private async _load(options: LoadSTTOptions = {}): Promise<void> {\n    const { onProgress, device = \"auto\" } = options;\n\n    onProgress?.({ status: \"Loading transformers.js...\" });\n\n    // Check if we're in Node.js or browser\n    const isNode = typeof process !== \"undefined\" && process.versions?.node;\n\n    // Import transformers.js dynamically\n    // tsdown handles resolution: Node.js = external, Browser = bundled\n    const { pipeline, env } = await import(\"@huggingface/transformers\");\n\n    // Configure environment based on runtime\n    if (isNode) {\n      // Node.js: allow local models (for CLI/server use)\n      env.allowLocalModels = true;\n      env.allowRemoteModels = true;\n    } else {\n      // Browser: use IndexedDB cache, fetch from HuggingFace CDN\n      env.useBrowserCache = true;\n      env.allowLocalModels = false;\n      // Load WASM files from CDN (not bundled with gerbil)\n      if (env.backends?.onnx?.wasm) {\n        env.backends.onnx.wasm.wasmPaths =\n          \"https://cdn.jsdelivr.net/npm/onnxruntime-web@1.21.0/dist/\";\n      }\n    }\n\n    // Determine device\n    // Note: Whisper ONNX models work best with fp32 on CPU/WASM\n    // WebGPU support for ASR is limited, so we use CPU for reliability\n    let tfDevice: \"webgpu\" | \"cpu\" | \"wasm\" = \"cpu\";\n\n    // In browser, use WASM for better compatibility\n    if (!isNode) {\n      tfDevice = \"wasm\";\n    }\n\n    // Store device mode\n    this._deviceMode = \"cpu\"; // STT always reports as CPU since WASM is CPU-based\n\n    onProgress?.({ status: `Loading ${this.modelConfig.id}...` });\n\n    // Load the ASR pipeline\n    // Always use fp32 for Whisper models (fp16 not available for ONNX ASR)\n    this.pipeline = await pipeline(\"automatic-speech-recognition\", this.modelConfig.repo, {\n      dtype: \"fp32\",\n      device: tfDevice,\n      progress_callback: (progress: any) => {\n        if (progress.status === \"progress\" && progress.file) {\n          onProgress?.({\n            status: `Downloading ${progress.file}`,\n            progress: Math.round(progress.progress || 0),\n            file: progress.file,\n          });\n        }\n      },\n    });\n\n    this._isLoaded = true;\n    onProgress?.({ status: `Ready (${tfDevice.toUpperCase()})!` });\n  }\n\n  /**\n   * Transcribe audio to text\n   *\n   * @param audio - Audio data as Float32Array (mono, 16kHz preferred) or Uint8Array (WAV file)\n   * @param options - Transcription options\n   * @returns Transcription result with text and optional timestamps\n   */\n  async transcribe(\n    audio: Float32Array | Uint8Array,\n    options: TranscribeOptions = {},\n  ): Promise<TranscribeResult> {\n    if (!this._isLoaded) {\n      throw new Error(\"STT model not loaded. Call load() first.\");\n    }\n\n    const { language, timestamps = false, onProgress } = options;\n    const startTime = performance.now();\n\n    // Convert Uint8Array (WAV) to Float32Array\n    let audioData: Float32Array;\n    let inputSampleRate = 16000;\n\n    if (audio instanceof Uint8Array) {\n      onProgress?.({ status: \"Decoding audio...\" });\n      const decoded = decodeWav(audio);\n      audioData = decoded.audio;\n      inputSampleRate = decoded.sampleRate;\n    } else {\n      audioData = audio;\n    }\n\n    // Resample to 16kHz if needed\n    if (inputSampleRate !== 16000) {\n      onProgress?.({ status: \"Resampling to 16kHz...\" });\n      audioData = resampleAudio(audioData, inputSampleRate, 16000);\n    }\n\n    const audioDuration = audioData.length / 16000;\n    onProgress?.({ status: `Transcribing ${audioDuration.toFixed(1)}s of audio...` });\n\n    // Build pipeline options\n    const pipelineOptions: any = {};\n\n    // Only set language for multilingual models\n    if (language && this.modelConfig.multilingual) {\n      pipelineOptions.language = language;\n      pipelineOptions.task = \"transcribe\";\n    }\n\n    // Enable timestamps if requested\n    if (timestamps) {\n      pipelineOptions.return_timestamps = true;\n    }\n\n    // Run transcription\n    const result = await this.pipeline(audioData, pipelineOptions);\n\n    const totalTime = performance.now() - startTime;\n\n    // Build result\n    let text = result.text?.trim() || \"\";\n\n    // Filter out Whisper artifacts\n    if (text === \"[BLANK_AUDIO]\" || text === \"(blank audio)\" || text === \"[BLANK AUDIO]\") {\n      text = \"\";\n    }\n\n    const transcribeResult: TranscribeResult = {\n      text,\n      language: language || (this.modelConfig.multilingual ? \"auto\" : \"en\"),\n      duration: audioDuration,\n      totalTime,\n    };\n\n    // Add segments if timestamps were requested\n    if (timestamps && result.chunks) {\n      transcribeResult.segments = result.chunks.map(\n        (chunk: any): TranscribeSegment => ({\n          text: chunk.text?.trim() || \"\",\n          start: chunk.timestamp?.[0] || 0,\n          end: chunk.timestamp?.[1] || 0,\n        }),\n      );\n    }\n\n    onProgress?.({ status: \"Done!\" });\n\n    return transcribeResult;\n  }\n\n  /**\n   * Create a streaming transcription session\n   *\n   * Transcribes audio in real-time by processing chunks at regular intervals.\n   * Perfect for live captioning, call transcription, or real-time subtitles.\n   *\n   * @param options - Streaming options\n   * @returns Streaming session controller\n   *\n   * @example\n   * ```ts\n   * const session = stt.createStreamingSession({\n   *   chunkDuration: 3000,  // Transcribe every 3 seconds\n   *   onChunk: (text, idx) => console.log(`Chunk ${idx}: ${text}`),\n   *   onTranscript: (fullText) => console.log(\"Full:\", fullText),\n   * });\n   *\n   * // Feed audio data as it comes in (Float32Array at 16kHz)\n   * session.feedAudio(audioChunk);\n   *\n   * // Or manually trigger transcription\n   * await session.flush();\n   *\n   * // Stop and get final transcript\n   * const finalText = await session.stop();\n   * ```\n   */\n  createStreamingSession(\n    options: StreamingTranscriptionOptions = {},\n  ): StreamingTranscriptionSession {\n    const {\n      chunkDuration = 3000,\n      minChunkSize = 8000, // ~0.5 seconds at 16kHz\n      onChunk,\n      onTranscript,\n      onError,\n      language,\n    } = options;\n\n    let audioBuffer: Float32Array[] = [];\n    let fullTranscript = \"\";\n    let chunkIndex = 0;\n    let intervalId: ReturnType<typeof setInterval> | null = null;\n    let isRunning = false;\n\n    const getBufferSize = (): number => {\n      return audioBuffer.reduce((sum, chunk) => sum + chunk.length, 0);\n    };\n\n    const mergeBuffer = (): Float32Array => {\n      const totalLength = getBufferSize();\n      const merged = new Float32Array(totalLength);\n      let offset = 0;\n      for (const chunk of audioBuffer) {\n        merged.set(chunk, offset);\n        offset += chunk.length;\n      }\n      return merged;\n    };\n\n    const transcribeBuffer = async (): Promise<string> => {\n      if (!this._isLoaded || getBufferSize() < minChunkSize) {\n        return \"\";\n      }\n\n      const audio = mergeBuffer();\n      audioBuffer = []; // Clear buffer\n\n      try {\n        const result = await this.transcribe(audio, { language });\n        const text = result.text.trim();\n\n        if (text) {\n          chunkIndex++;\n          onChunk?.(text, chunkIndex);\n\n          // Append to full transcript\n          fullTranscript = fullTranscript + (fullTranscript ? \" \" : \"\") + text;\n          onTranscript?.(fullTranscript);\n        }\n\n        return text;\n      } catch (e: any) {\n        onError?.(e.message || \"Transcription failed\");\n        return \"\";\n      }\n    };\n\n    let aborted = false;\n\n    const session: StreamingTranscriptionSession = {\n      feedAudio: (audio: Float32Array) => {\n        if (!aborted) {\n          audioBuffer.push(audio);\n        }\n      },\n\n      flush: async () => {\n        if (aborted) return \"\";\n        return transcribeBuffer();\n      },\n\n      start: () => {\n        if (isRunning || aborted) return;\n        isRunning = true;\n\n        intervalId = setInterval(async () => {\n          if (isRunning && !aborted) {\n            await transcribeBuffer();\n          }\n        }, chunkDuration);\n      },\n\n      stop: async () => {\n        isRunning = false;\n\n        if (intervalId) {\n          clearInterval(intervalId);\n          intervalId = null;\n        }\n\n        // Transcribe any remaining audio (unless aborted)\n        if (!aborted && getBufferSize() >= minChunkSize) {\n          await transcribeBuffer();\n        }\n\n        return fullTranscript;\n      },\n\n      abort: () => {\n        // Immediately stop without final transcription\n        aborted = true;\n        isRunning = false;\n\n        if (intervalId) {\n          clearInterval(intervalId);\n          intervalId = null;\n        }\n\n        audioBuffer = [];\n      },\n\n      isRunning: () => isRunning,\n\n      getTranscript: () => fullTranscript,\n\n      getChunkCount: () => chunkIndex,\n\n      reset: () => {\n        audioBuffer = [];\n        fullTranscript = \"\";\n        chunkIndex = 0;\n      },\n    };\n\n    return session;\n  }\n\n  /**\n   * Dispose of resources\n   */\n  dispose(): void {\n    this.pipeline = null;\n    this._isLoaded = false;\n    this.loadPromise = null;\n  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package/dist/stt-Dne6SENv.js

@@ -0,0 +1,434 @@
+//#region src/core/stt.ts
+/**
+* Available Whisper models
+* Ordered by size (smallest first)
+*/
+const WHISPER_MODELS = [
+	{
+		id: "whisper-tiny.en",
+		repo: "onnx-community/whisper-tiny.en",
+		description: "Tiny English-only model, fastest",
+		size: "39M",
+		multilingual: false,
+		languages: ["en"],
+		sampleRate: 16e3
+	},
+	{
+		id: "whisper-tiny",
+		repo: "onnx-community/whisper-tiny",
+		description: "Tiny multilingual model",
+		size: "39M",
+		multilingual: true,
+		languages: [
+			"en",
+			"es",
+			"fr",
+			"de",
+			"it",
+			"pt",
+			"nl",
+			"ru",
+			"zh",
+			"ja",
+			"ko"
+		],
+		sampleRate: 16e3
+	},
+	{
+		id: "whisper-base.en",
+		repo: "onnx-community/whisper-base.en",
+		description: "Base English-only model, good balance",
+		size: "74M",
+		multilingual: false,
+		languages: ["en"],
+		sampleRate: 16e3
+	},
+	{
+		id: "whisper-base",
+		repo: "onnx-community/whisper-base",
+		description: "Base multilingual model",
+		size: "74M",
+		multilingual: true,
+		languages: [
+			"en",
+			"es",
+			"fr",
+			"de",
+			"it",
+			"pt",
+			"nl",
+			"ru",
+			"zh",
+			"ja",
+			"ko"
+		],
+		sampleRate: 16e3
+	},
+	{
+		id: "whisper-small.en",
+		repo: "onnx-community/whisper-small.en",
+		description: "Small English-only model, high quality",
+		size: "244M",
+		multilingual: false,
+		languages: ["en"],
+		sampleRate: 16e3
+	},
+	{
+		id: "whisper-small",
+		repo: "onnx-community/whisper-small",
+		description: "Small multilingual model",
+		size: "244M",
+		multilingual: true,
+		languages: [
+			"en",
+			"es",
+			"fr",
+			"de",
+			"it",
+			"pt",
+			"nl",
+			"ru",
+			"zh",
+			"ja",
+			"ko"
+		],
+		sampleRate: 16e3
+	},
+	{
+		id: "whisper-large-v3-turbo",
+		repo: "onnx-community/whisper-large-v3-turbo",
+		description: "Large Turbo model, 5.4x faster, 80+ languages",
+		size: "809M",
+		multilingual: true,
+		languages: [
+			"en",
+			"es",
+			"fr",
+			"de",
+			"it",
+			"pt",
+			"nl",
+			"ru",
+			"zh",
+			"ja",
+			"ko",
+			"ar",
+			"hi",
+			"vi",
+			"th"
+		],
+		sampleRate: 16e3
+	}
+];
+const DEFAULT_MODEL = "whisper-tiny.en";
+/**
+* Decode WAV file to Float32Array
+* Handles stereo to mono conversion
+*/
+function decodeWav(buffer) {
+	const view = new DataView(buffer.buffer, buffer.byteOffset, buffer.byteLength);
+	if (String.fromCharCode(buffer[0], buffer[1], buffer[2], buffer[3]) !== "RIFF") throw new Error("Invalid WAV file: missing RIFF header");
+	const numChannels = view.getUint16(22, true);
+	const sampleRate = view.getUint32(24, true);
+	const bitsPerSample = view.getUint16(34, true);
+	if (bitsPerSample !== 16) throw new Error(`Unsupported bit depth: ${bitsPerSample}. Only 16-bit WAV is supported.`);
+	let dataOffset = 12;
+	while (dataOffset < buffer.length - 8) {
+		const chunkId = String.fromCharCode(buffer[dataOffset], buffer[dataOffset + 1], buffer[dataOffset + 2], buffer[dataOffset + 3]);
+		const chunkSize = view.getUint32(dataOffset + 4, true);
+		if (chunkId === "data") {
+			dataOffset += 8;
+			break;
+		}
+		dataOffset += 8 + chunkSize;
+	}
+	const dataSize = buffer.length - dataOffset;
+	const bytesPerSample = bitsPerSample / 8;
+	const totalSamples = Math.floor(dataSize / bytesPerSample);
+	const samplesPerChannel = Math.floor(totalSamples / numChannels);
+	const audio = new Float32Array(samplesPerChannel);
+	for (let i = 0; i < samplesPerChannel; i++) if (numChannels === 2) audio[i] = (view.getInt16(dataOffset + i * 4, true) + view.getInt16(dataOffset + i * 4 + 2, true)) / 2 / 32768;
+	else audio[i] = view.getInt16(dataOffset + i * 2, true) / 32768;
+	return {
+		audio,
+		sampleRate
+	};
+}
+/**
+* Resample audio to target sample rate using linear interpolation
+*/
+function resampleAudio(audio, fromRate, toRate) {
+	if (fromRate === toRate) return audio;
+	const ratio = toRate / fromRate;
+	const newLength = Math.round(audio.length * ratio);
+	const result = new Float32Array(newLength);
+	for (let i = 0; i < newLength; i++) {
+		const srcIndex = i / ratio;
+		const floor = Math.floor(srcIndex);
+		const ceil = Math.min(floor + 1, audio.length - 1);
+		const t = srcIndex - floor;
+		result[i] = audio[floor] * (1 - t) + audio[ceil] * t;
+	}
+	return result;
+}
+/**
+* Speech-to-Text using Whisper ONNX models
+*/
+var WhisperSTT = class {
+	modelConfig;
+	pipeline = null;
+	loadPromise = null;
+	_isLoaded = false;
+	_deviceMode = "cpu";
+	constructor(modelId = DEFAULT_MODEL) {
+		const config = WHISPER_MODELS.find((m) => m.id === modelId);
+		if (!config) {
+			const available = WHISPER_MODELS.map((m) => m.id).join(", ");
+			throw new Error(`Unknown STT model: ${modelId}. Available: ${available}`);
+		}
+		this.modelConfig = config;
+	}
+	/**
+	* Check if model is loaded
+	*/
+	isLoaded() {
+		return this._isLoaded;
+	}
+	/**
+	* Get model configuration
+	*/
+	getModelConfig() {
+		return this.modelConfig;
+	}
+	/**
+	* Get model info (alias for getModelConfig)
+	*/
+	getModelInfo() {
+		return this.modelConfig;
+	}
+	/**
+	* Get current device mode
+	*/
+	getDeviceMode() {
+		return this._deviceMode;
+	}
+	/**
+	* List available models
+	*/
+	static listModels() {
+		return [...WHISPER_MODELS];
+	}
+	/**
+	* Load the STT model
+	*/
+	async load(options = {}) {
+		if (this._isLoaded) return;
+		if (this.loadPromise) {
+			await this.loadPromise;
+			return;
+		}
+		this.loadPromise = this._load(options);
+		await this.loadPromise;
+	}
+	async _load(options = {}) {
+		const { onProgress, device = "auto" } = options;
+		onProgress?.({ status: "Loading transformers.js..." });
+		const isNode = typeof process !== "undefined" && process.versions?.node;
+		const { pipeline, env } = await import("./transformers.web-u34VxRFM.js");
+		if (isNode) {
+			env.allowLocalModels = true;
+			env.allowRemoteModels = true;
+		} else {
+			env.useBrowserCache = true;
+			env.allowLocalModels = false;
+			if (env.backends?.onnx?.wasm) env.backends.onnx.wasm.wasmPaths = "https://cdn.jsdelivr.net/npm/onnxruntime-web@1.21.0/dist/";
+		}
+		let tfDevice = "cpu";
+		if (!isNode) tfDevice = "wasm";
+		this._deviceMode = "cpu";
+		onProgress?.({ status: `Loading ${this.modelConfig.id}...` });
+		this.pipeline = await pipeline("automatic-speech-recognition", this.modelConfig.repo, {
+			dtype: "fp32",
+			device: tfDevice,
+			progress_callback: (progress) => {
+				if (progress.status === "progress" && progress.file) onProgress?.({
+					status: `Downloading ${progress.file}`,
+					progress: Math.round(progress.progress || 0),
+					file: progress.file
+				});
+			}
+		});
+		this._isLoaded = true;
+		onProgress?.({ status: `Ready (${tfDevice.toUpperCase()})!` });
+	}
+	/**
+	* Transcribe audio to text
+	*
+	* @param audio - Audio data as Float32Array (mono, 16kHz preferred) or Uint8Array (WAV file)
+	* @param options - Transcription options
+	* @returns Transcription result with text and optional timestamps
+	*/
+	async transcribe(audio, options = {}) {
+		if (!this._isLoaded) throw new Error("STT model not loaded. Call load() first.");
+		const { language, timestamps = false, onProgress } = options;
+		const startTime = performance.now();
+		let audioData;
+		let inputSampleRate = 16e3;
+		if (audio instanceof Uint8Array) {
+			onProgress?.({ status: "Decoding audio..." });
+			const decoded = decodeWav(audio);
+			audioData = decoded.audio;
+			inputSampleRate = decoded.sampleRate;
+		} else audioData = audio;
+		if (inputSampleRate !== 16e3) {
+			onProgress?.({ status: "Resampling to 16kHz..." });
+			audioData = resampleAudio(audioData, inputSampleRate, 16e3);
+		}
+		const audioDuration = audioData.length / 16e3;
+		onProgress?.({ status: `Transcribing ${audioDuration.toFixed(1)}s of audio...` });
+		const pipelineOptions = {};
+		if (language && this.modelConfig.multilingual) {
+			pipelineOptions.language = language;
+			pipelineOptions.task = "transcribe";
+		}
+		if (timestamps) pipelineOptions.return_timestamps = true;
+		const result = await this.pipeline(audioData, pipelineOptions);
+		const totalTime = performance.now() - startTime;
+		let text = result.text?.trim() || "";
+		if (text === "[BLANK_AUDIO]" || text === "(blank audio)" || text === "[BLANK AUDIO]") text = "";
+		const transcribeResult = {
+			text,
+			language: language || (this.modelConfig.multilingual ? "auto" : "en"),
+			duration: audioDuration,
+			totalTime
+		};
+		if (timestamps && result.chunks) transcribeResult.segments = result.chunks.map((chunk) => ({
+			text: chunk.text?.trim() || "",
+			start: chunk.timestamp?.[0] || 0,
+			end: chunk.timestamp?.[1] || 0
+		}));
+		onProgress?.({ status: "Done!" });
+		return transcribeResult;
+	}
+	/**
+	* Create a streaming transcription session
+	*
+	* Transcribes audio in real-time by processing chunks at regular intervals.
+	* Perfect for live captioning, call transcription, or real-time subtitles.
+	*
+	* @param options - Streaming options
+	* @returns Streaming session controller
+	*
+	* @example
+	* ```ts
+	* const session = stt.createStreamingSession({
+	*   chunkDuration: 3000,  // Transcribe every 3 seconds
+	*   onChunk: (text, idx) => console.log(`Chunk ${idx}: ${text}`),
+	*   onTranscript: (fullText) => console.log("Full:", fullText),
+	* });
+	*
+	* // Feed audio data as it comes in (Float32Array at 16kHz)
+	* session.feedAudio(audioChunk);
+	*
+	* // Or manually trigger transcription
+	* await session.flush();
+	*
+	* // Stop and get final transcript
+	* const finalText = await session.stop();
+	* ```
+	*/
+	createStreamingSession(options = {}) {
+		const { chunkDuration = 3e3, minChunkSize = 8e3, onChunk, onTranscript, onError, language } = options;
+		let audioBuffer = [];
+		let fullTranscript = "";
+		let chunkIndex = 0;
+		let intervalId = null;
+		let isRunning = false;
+		const getBufferSize = () => {
+			return audioBuffer.reduce((sum, chunk) => sum + chunk.length, 0);
+		};
+		const mergeBuffer = () => {
+			const totalLength = getBufferSize();
+			const merged = new Float32Array(totalLength);
+			let offset = 0;
+			for (const chunk of audioBuffer) {
+				merged.set(chunk, offset);
+				offset += chunk.length;
+			}
+			return merged;
+		};
+		const transcribeBuffer = async () => {
+			if (!this._isLoaded || getBufferSize() < minChunkSize) return "";
+			const audio = mergeBuffer();
+			audioBuffer = [];
+			try {
+				const text = (await this.transcribe(audio, { language })).text.trim();
+				if (text) {
+					chunkIndex++;
+					onChunk?.(text, chunkIndex);
+					fullTranscript = fullTranscript + (fullTranscript ? " " : "") + text;
+					onTranscript?.(fullTranscript);
+				}
+				return text;
+			} catch (e) {
+				onError?.(e.message || "Transcription failed");
+				return "";
+			}
+		};
+		let aborted = false;
+		return {
+			feedAudio: (audio) => {
+				if (!aborted) audioBuffer.push(audio);
+			},
+			flush: async () => {
+				if (aborted) return "";
+				return transcribeBuffer();
+			},
+			start: () => {
+				if (isRunning || aborted) return;
+				isRunning = true;
+				intervalId = setInterval(async () => {
+					if (isRunning && !aborted) await transcribeBuffer();
+				}, chunkDuration);
+			},
+			stop: async () => {
+				isRunning = false;
+				if (intervalId) {
+					clearInterval(intervalId);
+					intervalId = null;
+				}
+				if (!aborted && getBufferSize() >= minChunkSize) await transcribeBuffer();
+				return fullTranscript;
+			},
+			abort: () => {
+				aborted = true;
+				isRunning = false;
+				if (intervalId) {
+					clearInterval(intervalId);
+					intervalId = null;
+				}
+				audioBuffer = [];
+			},
+			isRunning: () => isRunning,
+			getTranscript: () => fullTranscript,
+			getChunkCount: () => chunkIndex,
+			reset: () => {
+				audioBuffer = [];
+				fullTranscript = "";
+				chunkIndex = 0;
+			}
+		};
+	}
+	/**
+	* Dispose of resources
+	*/
+	dispose() {
+		this.pipeline = null;
+		this._isLoaded = false;
+		this.loadPromise = null;
+	}
+};
+
+//#endregion
+export { WhisperSTT };
+//# sourceMappingURL=stt-Dne6SENv.js.map

package/dist/stt-Dne6SENv.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"stt-Dne6SENv.js","names":["WHISPER_MODELS: STTModelConfig[]","tfDevice: \"webgpu\" | \"cpu\" | \"wasm\"","audioData: Float32Array","pipelineOptions: any","transcribeResult: TranscribeResult","audioBuffer: Float32Array[]","intervalId: ReturnType<typeof setInterval> | null","e: any"],"sources":["../src/core/stt.ts"],"sourcesContent":["/**\n * Speech-to-Text with Whisper\n *\n * Provides local speech recognition using Whisper ONNX models via transformers.js.\n * Supports multiple model sizes and languages.\n *\n * @example\n * ```ts\n * const stt = new WhisperSTT();\n * await stt.load({ onProgress: (p) => console.log(p.status) });\n *\n * // Transcribe audio (Float32Array at 16kHz)\n * const result = await stt.transcribe(audioData);\n * console.log(result.text);\n *\n * // With timestamps\n * const result = await stt.transcribe(audioData, { timestamps: true });\n * for (const seg of result.segments) {\n *   console.log(`[${seg.start.toFixed(1)}s] ${seg.text}`);\n * }\n * ```\n */\n\nimport type {\n  LoadSTTOptions,\n  ProgressInfo,\n  STTModelConfig,\n  StreamingTranscriptionOptions,\n  StreamingTranscriptionSession,\n  TranscribeOptions,\n  TranscribeResult,\n  TranscribeSegment,\n} from \"./types.js\";\n\n// ============================================\n// Model Registry\n// ============================================\n\n/**\n * Available Whisper models\n * Ordered by size (smallest first)\n */\nexport const WHISPER_MODELS: STTModelConfig[] = [\n  {\n    id: \"whisper-tiny.en\",\n    repo: \"onnx-community/whisper-tiny.en\",\n    description: \"Tiny English-only model, fastest\",\n    size: \"39M\",\n    multilingual: false,\n    languages: [\"en\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-tiny\",\n    repo: \"onnx-community/whisper-tiny\",\n    description: \"Tiny multilingual model\",\n    size: \"39M\",\n    multilingual: true,\n    languages: [\"en\", \"es\", \"fr\", \"de\", \"it\", \"pt\", \"nl\", \"ru\", \"zh\", \"ja\", \"ko\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-base.en\",\n    repo: \"onnx-community/whisper-base.en\",\n    description: \"Base English-only model, good balance\",\n    size: \"74M\",\n    multilingual: false,\n    languages: [\"en\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-base\",\n    repo: \"onnx-community/whisper-base\",\n    description: \"Base multilingual model\",\n    size: \"74M\",\n    multilingual: true,\n    languages: [\"en\", \"es\", \"fr\", \"de\", \"it\", \"pt\", \"nl\", \"ru\", \"zh\", \"ja\", \"ko\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-small.en\",\n    repo: \"onnx-community/whisper-small.en\",\n    description: \"Small English-only model, high quality\",\n    size: \"244M\",\n    multilingual: false,\n    languages: [\"en\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-small\",\n    repo: \"onnx-community/whisper-small\",\n    description: \"Small multilingual model\",\n    size: \"244M\",\n    multilingual: true,\n    languages: [\"en\", \"es\", \"fr\", \"de\", \"it\", \"pt\", \"nl\", \"ru\", \"zh\", \"ja\", \"ko\"],\n    sampleRate: 16000,\n  },\n  {\n    id: \"whisper-large-v3-turbo\",\n    repo: \"onnx-community/whisper-large-v3-turbo\",\n    description: \"Large Turbo model, 5.4x faster, 80+ languages\",\n    size: \"809M\",\n    multilingual: true,\n    languages: [\n      \"en\",\n      \"es\",\n      \"fr\",\n      \"de\",\n      \"it\",\n      \"pt\",\n      \"nl\",\n      \"ru\",\n      \"zh\",\n      \"ja\",\n      \"ko\",\n      \"ar\",\n      \"hi\",\n      \"vi\",\n      \"th\",\n    ],\n    sampleRate: 16000,\n  },\n];\n\n// Default model\nconst DEFAULT_MODEL = \"whisper-tiny.en\";\n\n// ============================================\n// Audio Utilities\n// ============================================\n\n/**\n * Decode WAV file to Float32Array\n * Handles stereo to mono conversion\n */\nexport function decodeWav(buffer: Uint8Array): { audio: Float32Array; sampleRate: number } {\n  const view = new DataView(buffer.buffer, buffer.byteOffset, buffer.byteLength);\n\n  // Validate RIFF header\n  const riff = String.fromCharCode(buffer[0], buffer[1], buffer[2], buffer[3]);\n  if (riff !== \"RIFF\") {\n    throw new Error(\"Invalid WAV file: missing RIFF header\");\n  }\n\n  // Get format details\n  const numChannels = view.getUint16(22, true);\n  const sampleRate = view.getUint32(24, true);\n  const bitsPerSample = view.getUint16(34, true);\n\n  if (bitsPerSample !== 16) {\n    throw new Error(`Unsupported bit depth: ${bitsPerSample}. Only 16-bit WAV is supported.`);\n  }\n\n  // Find data chunk\n  let dataOffset = 12;\n  while (dataOffset < buffer.length - 8) {\n    const chunkId = String.fromCharCode(\n      buffer[dataOffset],\n      buffer[dataOffset + 1],\n      buffer[dataOffset + 2],\n      buffer[dataOffset + 3],\n    );\n    const chunkSize = view.getUint32(dataOffset + 4, true);\n    if (chunkId === \"data\") {\n      dataOffset += 8;\n      break;\n    }\n    dataOffset += 8 + chunkSize;\n  }\n\n  const dataSize = buffer.length - dataOffset;\n  const bytesPerSample = bitsPerSample / 8;\n  const totalSamples = Math.floor(dataSize / bytesPerSample);\n  const samplesPerChannel = Math.floor(totalSamples / numChannels);\n\n  // Convert to mono Float32Array\n  const audio = new Float32Array(samplesPerChannel);\n\n  for (let i = 0; i < samplesPerChannel; i++) {\n    if (numChannels === 2) {\n      const left = view.getInt16(dataOffset + i * 4, true);\n      const right = view.getInt16(dataOffset + i * 4 + 2, true);\n      audio[i] = (left + right) / 2 / 32768;\n    } else {\n      const sample = view.getInt16(dataOffset + i * 2, true);\n      audio[i] = sample / 32768;\n    }\n  }\n\n  return { audio, sampleRate };\n}\n\n/**\n * Resample audio to target sample rate using linear interpolation\n */\nexport function resampleAudio(audio: Float32Array, fromRate: number, toRate: number): Float32Array {\n  if (fromRate === toRate) return audio;\n\n  const ratio = toRate / fromRate;\n  const newLength = Math.round(audio.length * ratio);\n  const result = new Float32Array(newLength);\n\n  for (let i = 0; i < newLength; i++) {\n    const srcIndex = i / ratio;\n    const floor = Math.floor(srcIndex);\n    const ceil = Math.min(floor + 1, audio.length - 1);\n    const t = srcIndex - floor;\n    result[i] = audio[floor] * (1 - t) + audio[ceil] * t;\n  }\n\n  return result;\n}\n\n// ============================================\n// WhisperSTT Class\n// ============================================\n\n/**\n * Speech-to-Text using Whisper ONNX models\n */\nexport class WhisperSTT {\n  private modelConfig: STTModelConfig;\n  private pipeline: any = null;\n  private loadPromise: Promise<void> | null = null;\n  private _isLoaded = false;\n  private _deviceMode: \"webgpu\" | \"cpu\" = \"cpu\";\n\n  constructor(modelId: string = DEFAULT_MODEL) {\n    const config = WHISPER_MODELS.find((m) => m.id === modelId);\n    if (!config) {\n      const available = WHISPER_MODELS.map((m) => m.id).join(\", \");\n      throw new Error(`Unknown STT model: ${modelId}. Available: ${available}`);\n    }\n    this.modelConfig = config;\n  }\n\n  /**\n   * Check if model is loaded\n   */\n  isLoaded(): boolean {\n    return this._isLoaded;\n  }\n\n  /**\n   * Get model configuration\n   */\n  getModelConfig(): STTModelConfig {\n    return this.modelConfig;\n  }\n\n  /**\n   * Get model info (alias for getModelConfig)\n   */\n  getModelInfo(): STTModelConfig {\n    return this.modelConfig;\n  }\n\n  /**\n   * Get current device mode\n   */\n  getDeviceMode(): \"webgpu\" | \"cpu\" {\n    return this._deviceMode;\n  }\n\n  /**\n   * List available models\n   */\n  static listModels(): STTModelConfig[] {\n    return [...WHISPER_MODELS];\n  }\n\n  /**\n   * Load the STT model\n   */\n  async load(options: LoadSTTOptions = {}): Promise<void> {\n    if (this._isLoaded) return;\n    if (this.loadPromise) {\n      await this.loadPromise;\n      return;\n    }\n\n    this.loadPromise = this._load(options);\n    await this.loadPromise;\n  }\n\n  private async _load(options: LoadSTTOptions = {}): Promise<void> {\n    const { onProgress, device = \"auto\" } = options;\n\n    onProgress?.({ status: \"Loading transformers.js...\" });\n\n    // Check if we're in Node.js or browser\n    const isNode = typeof process !== \"undefined\" && process.versions?.node;\n\n    // Import transformers.js dynamically\n    // tsdown handles resolution: Node.js = external, Browser = bundled\n    const { pipeline, env } = await import(\"@huggingface/transformers\");\n\n    // Configure environment based on runtime\n    if (isNode) {\n      // Node.js: allow local models (for CLI/server use)\n      env.allowLocalModels = true;\n      env.allowRemoteModels = true;\n    } else {\n      // Browser: use IndexedDB cache, fetch from HuggingFace CDN\n      env.useBrowserCache = true;\n      env.allowLocalModels = false;\n      // Load WASM files from CDN (not bundled with gerbil)\n      if (env.backends?.onnx?.wasm) {\n        env.backends.onnx.wasm.wasmPaths =\n          \"https://cdn.jsdelivr.net/npm/onnxruntime-web@1.21.0/dist/\";\n      }\n    }\n\n    // Determine device\n    // Note: Whisper ONNX models work best with fp32 on CPU/WASM\n    // WebGPU support for ASR is limited, so we use CPU for reliability\n    let tfDevice: \"webgpu\" | \"cpu\" | \"wasm\" = \"cpu\";\n\n    // In browser, use WASM for better compatibility\n    if (!isNode) {\n      tfDevice = \"wasm\";\n    }\n\n    // Store device mode\n    this._deviceMode = \"cpu\"; // STT always reports as CPU since WASM is CPU-based\n\n    onProgress?.({ status: `Loading ${this.modelConfig.id}...` });\n\n    // Load the ASR pipeline\n    // Always use fp32 for Whisper models (fp16 not available for ONNX ASR)\n    this.pipeline = await pipeline(\"automatic-speech-recognition\", this.modelConfig.repo, {\n      dtype: \"fp32\",\n      device: tfDevice,\n      progress_callback: (progress: any) => {\n        if (progress.status === \"progress\" && progress.file) {\n          onProgress?.({\n            status: `Downloading ${progress.file}`,\n            progress: Math.round(progress.progress || 0),\n            file: progress.file,\n          });\n        }\n      },\n    });\n\n    this._isLoaded = true;\n    onProgress?.({ status: `Ready (${tfDevice.toUpperCase()})!` });\n  }\n\n  /**\n   * Transcribe audio to text\n   *\n   * @param audio - Audio data as Float32Array (mono, 16kHz preferred) or Uint8Array (WAV file)\n   * @param options - Transcription options\n   * @returns Transcription result with text and optional timestamps\n   */\n  async transcribe(\n    audio: Float32Array | Uint8Array,\n    options: TranscribeOptions = {},\n  ): Promise<TranscribeResult> {\n    if (!this._isLoaded) {\n      throw new Error(\"STT model not loaded. Call load() first.\");\n    }\n\n    const { language, timestamps = false, onProgress } = options;\n    const startTime = performance.now();\n\n    // Convert Uint8Array (WAV) to Float32Array\n    let audioData: Float32Array;\n    let inputSampleRate = 16000;\n\n    if (audio instanceof Uint8Array) {\n      onProgress?.({ status: \"Decoding audio...\" });\n      const decoded = decodeWav(audio);\n      audioData = decoded.audio;\n      inputSampleRate = decoded.sampleRate;\n    } else {\n      audioData = audio;\n    }\n\n    // Resample to 16kHz if needed\n    if (inputSampleRate !== 16000) {\n      onProgress?.({ status: \"Resampling to 16kHz...\" });\n      audioData = resampleAudio(audioData, inputSampleRate, 16000);\n    }\n\n    const audioDuration = audioData.length / 16000;\n    onProgress?.({ status: `Transcribing ${audioDuration.toFixed(1)}s of audio...` });\n\n    // Build pipeline options\n    const pipelineOptions: any = {};\n\n    // Only set language for multilingual models\n    if (language && this.modelConfig.multilingual) {\n      pipelineOptions.language = language;\n      pipelineOptions.task = \"transcribe\";\n    }\n\n    // Enable timestamps if requested\n    if (timestamps) {\n      pipelineOptions.return_timestamps = true;\n    }\n\n    // Run transcription\n    const result = await this.pipeline(audioData, pipelineOptions);\n\n    const totalTime = performance.now() - startTime;\n\n    // Build result\n    let text = result.text?.trim() || \"\";\n\n    // Filter out Whisper artifacts\n    if (text === \"[BLANK_AUDIO]\" || text === \"(blank audio)\" || text === \"[BLANK AUDIO]\") {\n      text = \"\";\n    }\n\n    const transcribeResult: TranscribeResult = {\n      text,\n      language: language || (this.modelConfig.multilingual ? \"auto\" : \"en\"),\n      duration: audioDuration,\n      totalTime,\n    };\n\n    // Add segments if timestamps were requested\n    if (timestamps && result.chunks) {\n      transcribeResult.segments = result.chunks.map(\n        (chunk: any): TranscribeSegment => ({\n          text: chunk.text?.trim() || \"\",\n          start: chunk.timestamp?.[0] || 0,\n          end: chunk.timestamp?.[1] || 0,\n        }),\n      );\n    }\n\n    onProgress?.({ status: \"Done!\" });\n\n    return transcribeResult;\n  }\n\n  /**\n   * Create a streaming transcription session\n   *\n   * Transcribes audio in real-time by processing chunks at regular intervals.\n   * Perfect for live captioning, call transcription, or real-time subtitles.\n   *\n   * @param options - Streaming options\n   * @returns Streaming session controller\n   *\n   * @example\n   * ```ts\n   * const session = stt.createStreamingSession({\n   *   chunkDuration: 3000,  // Transcribe every 3 seconds\n   *   onChunk: (text, idx) => console.log(`Chunk ${idx}: ${text}`),\n   *   onTranscript: (fullText) => console.log(\"Full:\", fullText),\n   * });\n   *\n   * // Feed audio data as it comes in (Float32Array at 16kHz)\n   * session.feedAudio(audioChunk);\n   *\n   * // Or manually trigger transcription\n   * await session.flush();\n   *\n   * // Stop and get final transcript\n   * const finalText = await session.stop();\n   * ```\n   */\n  createStreamingSession(\n    options: StreamingTranscriptionOptions = {},\n  ): StreamingTranscriptionSession {\n    const {\n      chunkDuration = 3000,\n      minChunkSize = 8000, // ~0.5 seconds at 16kHz\n      onChunk,\n      onTranscript,\n      onError,\n      language,\n    } = options;\n\n    let audioBuffer: Float32Array[] = [];\n    let fullTranscript = \"\";\n    let chunkIndex = 0;\n    let intervalId: ReturnType<typeof setInterval> | null = null;\n    let isRunning = false;\n\n    const getBufferSize = (): number => {\n      return audioBuffer.reduce((sum, chunk) => sum + chunk.length, 0);\n    };\n\n    const mergeBuffer = (): Float32Array => {\n      const totalLength = getBufferSize();\n      const merged = new Float32Array(totalLength);\n      let offset = 0;\n      for (const chunk of audioBuffer) {\n        merged.set(chunk, offset);\n        offset += chunk.length;\n      }\n      return merged;\n    };\n\n    const transcribeBuffer = async (): Promise<string> => {\n      if (!this._isLoaded || getBufferSize() < minChunkSize) {\n        return \"\";\n      }\n\n      const audio = mergeBuffer();\n      audioBuffer = []; // Clear buffer\n\n      try {\n        const result = await this.transcribe(audio, { language });\n        const text = result.text.trim();\n\n        if (text) {\n          chunkIndex++;\n          onChunk?.(text, chunkIndex);\n\n          // Append to full transcript\n          fullTranscript = fullTranscript + (fullTranscript ? \" \" : \"\") + text;\n          onTranscript?.(fullTranscript);\n        }\n\n        return text;\n      } catch (e: any) {\n        onError?.(e.message || \"Transcription failed\");\n        return \"\";\n      }\n    };\n\n    let aborted = false;\n\n    const session: StreamingTranscriptionSession = {\n      feedAudio: (audio: Float32Array) => {\n        if (!aborted) {\n          audioBuffer.push(audio);\n        }\n      },\n\n      flush: async () => {\n        if (aborted) return \"\";\n        return transcribeBuffer();\n      },\n\n      start: () => {\n        if (isRunning || aborted) return;\n        isRunning = true;\n\n        intervalId = setInterval(async () => {\n          if (isRunning && !aborted) {\n            await transcribeBuffer();\n          }\n        }, chunkDuration);\n      },\n\n      stop: async () => {\n        isRunning = false;\n\n        if (intervalId) {\n          clearInterval(intervalId);\n          intervalId = null;\n        }\n\n        // Transcribe any remaining audio (unless aborted)\n        if (!aborted && getBufferSize() >= minChunkSize) {\n          await transcribeBuffer();\n        }\n\n        return fullTranscript;\n      },\n\n      abort: () => {\n        // Immediately stop without final transcription\n        aborted = true;\n        isRunning = false;\n\n        if (intervalId) {\n          clearInterval(intervalId);\n          intervalId = null;\n        }\n\n        audioBuffer = [];\n      },\n\n      isRunning: () => isRunning,\n\n      getTranscript: () => fullTranscript,\n\n      getChunkCount: () => chunkIndex,\n\n      reset: () => {\n        audioBuffer = [];\n        fullTranscript = \"\";\n        chunkIndex = 0;\n      },\n    };\n\n    return session;\n  }\n\n  /**\n   * Dispose of resources\n   */\n  dispose(): void {\n    this.pipeline = null;\n    this._isLoaded = false;\n    this.loadPromise = null;\n  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package/dist/{tools-BsiEE6f2.mjs → tools-Bi1P7Xoy.mjs}

@@ -1,4 +1,4 @@
-import { n as zodToJsonSchema } from "./utils-7vXqtq2Q.mjs";
+import { n as zodToJsonSchema } from "./utils-CZBZ8dgR.mjs";
 import { z } from "zod";
 
 //#region src/core/docs.ts
@@ -397,8 +397,8 @@ function getToolDefinitions() {
 * No module imports - we eval the execute function directly
 */
 async function loadTools(dir) {
-	const fs = await import("fs");
-	const pathModule = await import("path");
+	const fs = await import("node:fs");
+	const pathModule = await import("node:path");
 	if (!fs.existsSync(dir)) return [];
 	const files = fs.readdirSync(dir).filter((f) => f.endsWith(".tool.ts") || f.endsWith(".tool.js"));
 	const results = [];
@@ -462,8 +462,7 @@ function parseToolFile(content, fallbackName) {
 			execute: config.execute
 		};
 		return null;
-	} catch (e) {
-		console.error("Failed to parse tool file:", e);
+	} catch (_e) {
 		return null;
 	}
 }
@@ -471,7 +470,7 @@ function parseToolFile(content, fallbackName) {
 * Load tools from project .gerbil/tools directory
 */
 async function loadProjectTools() {
-	return loadTools((await import("path")).join(process.cwd(), ".gerbil", "tools"));
+	return loadTools((await import("node:path")).join(process.cwd(), ".gerbil", "tools"));
 }
 /**
 * Format tools for Qwen3 prompt
@@ -564,4 +563,4 @@ const docsTool = defineTool({
 
 //#endregion
 export { getToolDefinitions as a, setToolContext as c, getTool as i, executeToolCall as n, loadProjectTools as o, formatToolsForPrompt as r, parseToolCall as s, defineTool as t };
-//# sourceMappingURL=tools-BsiEE6f2.mjs.map
+//# sourceMappingURL=tools-Bi1P7Xoy.mjs.map