@voice-kit/core 0.1.0 → 0.1.2

package/dist/audio.cjs

@@ -0,0 +1,533 @@
+'use strict';
+
+var ffmpeg = require('fluent-ffmpeg');
+var stream = require('stream');
+var pino = require('pino');
+var events = require('events');
+
+function _interopDefault (e) { return e && e.__esModule ? e : { default: e }; }
+
+var ffmpeg__default = /*#__PURE__*/_interopDefault(ffmpeg);
+var pino__default = /*#__PURE__*/_interopDefault(pino);
+
+// src/audio/codec/index.ts
+var MULAW_BIAS = 33;
+var MULAW_MAX = 32767;
+function mulawToLinear(sample) {
+  sample = ~sample & 255;
+  const sign = sample & 128;
+  const exponent = sample >> 4 & 7;
+  const mantissa = sample & 15;
+  let linear = (mantissa << 1) + 33 << exponent;
+  linear -= 33;
+  return sign !== 0 ? -linear : linear;
+}
+function linearToMulaw(sample) {
+  sample = Math.max(-32768, Math.min(32767, sample));
+  const sign = sample < 0 ? 128 : 0;
+  if (sample < 0) sample = -sample;
+  sample += MULAW_BIAS;
+  if (sample > MULAW_MAX) sample = MULAW_MAX;
+  let exponent = 7;
+  let expMask = 16384;
+  for (; exponent > 0; exponent--) {
+    if ((sample & expMask) !== 0) break;
+    expMask >>= 1;
+  }
+  const mantissa = sample >> exponent + 3 & 15;
+  const mulaw = ~(sign | exponent << 4 | mantissa) & 255;
+  return mulaw;
+}
+function mulawBufferToPcm(buf) {
+  const pcm = Buffer.allocUnsafe(buf.length * 2);
+  for (let i = 0; i < buf.length; i++) {
+    const sample = mulawToLinear(buf[i] ?? 0);
+    pcm.writeInt16LE(sample, i * 2);
+  }
+  return pcm;
+}
+function pcmBufferToMulaw(buf) {
+  const samples = buf.length >> 1;
+  const mulaw = Buffer.allocUnsafe(samples);
+  for (let i = 0; i < samples; i++) {
+    const sample = buf.readInt16LE(i * 2);
+    mulaw[i] = linearToMulaw(sample);
+  }
+  return mulaw;
+}
+function base64MulawToPcm(base64) {
+  const mulaw = Buffer.from(base64, "base64");
+  return mulawBufferToPcm(mulaw);
+}
+function pcmToBase64Mulaw(pcm) {
+  return pcmBufferToMulaw(pcm).toString("base64");
+}
+
+// src/errors/base.ts
+var VoiceKitError = class extends Error {
+  code;
+  callId;
+  provider;
+  retryable;
+  severity;
+  cause;
+  constructor(params) {
+    super(params.message);
+    this.name = this.constructor.name;
+    this.code = params.code;
+    this.callId = params.callId;
+    this.provider = params.provider;
+    this.retryable = params.retryable ?? false;
+    this.severity = params.severity ?? "medium";
+    this.cause = params.cause;
+    Object.setPrototypeOf(this, new.target.prototype);
+  }
+  toJSON() {
+    return {
+      name: this.name,
+      code: this.code,
+      message: this.message,
+      callId: this.callId,
+      provider: this.provider,
+      retryable: this.retryable,
+      severity: this.severity
+    };
+  }
+};
+
+// src/errors/telephony.errors.ts
+var TelephonyError = class extends VoiceKitError {
+  to;
+  from;
+  constructor(params) {
+    super(params);
+    this.to = params.to;
+    this.from = params.from;
+  }
+};
+var AudioTransportError = class extends TelephonyError {
+  constructor(provider, cause, callId) {
+    super({
+      code: "AUDIO_TRANSPORT_ERROR",
+      message: `Audio transport error on ${provider}`,
+      provider,
+      callId,
+      retryable: true,
+      severity: "high",
+      cause
+    });
+  }
+};
+
+// src/audio/resampler/index.ts
+async function resample(buf, fromHz, toHz) {
+  if (fromHz === toHz) return buf;
+  return new Promise((resolve, reject) => {
+    const chunks = [];
+    const input = new stream.Readable({
+      read() {
+        this.push(buf);
+        this.push(null);
+      }
+    });
+    const output = new stream.PassThrough();
+    output.on("data", (chunk) => chunks.push(chunk));
+    output.on("end", () => resolve(Buffer.concat(chunks)));
+    output.on(
+      "error",
+      (err) => reject(
+        new AudioTransportError(
+          "ffmpeg-resampler",
+          err
+        )
+      )
+    );
+    ffmpeg__default.default(input).inputOptions([
+      "-f s16le",
+      `-ar ${fromHz}`,
+      "-ac 1"
+    ]).outputOptions([
+      "-f s16le",
+      `-ar ${toHz}`,
+      "-ac 1"
+    ]).on(
+      "error",
+      (err) => reject(
+        new AudioTransportError("ffmpeg-resampler", err)
+      )
+    ).pipe(output, { end: true });
+  });
+}
+function createResamplerStream(fromHz, toHz) {
+  const output = new stream.PassThrough();
+  if (fromHz === toHz) {
+    return output;
+  }
+  return output;
+}
+async function* resampleStream(audio, fromHz, toHz) {
+  if (fromHz === toHz) {
+    yield* audio;
+    return;
+  }
+  const CHUNK_SIZE = 16e3;
+  let pending = Buffer.alloc(0);
+  for await (const chunk of audio) {
+    pending = Buffer.concat([pending, chunk]);
+    while (pending.length >= CHUNK_SIZE) {
+      const slice = pending.subarray(0, CHUNK_SIZE);
+      pending = pending.subarray(CHUNK_SIZE);
+      yield await resample(slice, fromHz, toHz);
+    }
+  }
+  if (pending.length > 0) {
+    yield await resample(pending, fromHz, toHz);
+  }
+}
+var logger = pino__default.default({ name: "@voice-kit/core:pipeline" });
+var AUDIO_PROFILES = {
+  twilio: {
+    inputFormat: "mulaw",
+    inputSampleRate: 8e3,
+    sttSampleRate: 16e3,
+    outputSampleRate: 8e3,
+    outputFormat: "mulaw",
+    vadConfig: {
+      threshold: 0.6,
+      positiveSpeechFrames: 3,
+      negativeSpeechFrames: 5,
+      debounceMs: 150,
+      sampleRate: 16e3
+    }
+  },
+  exotel: {
+    inputFormat: "mulaw",
+    inputSampleRate: 8e3,
+    sttSampleRate: 16e3,
+    outputSampleRate: 8e3,
+    outputFormat: "mulaw",
+    vadConfig: {
+      threshold: 0.55,
+      // Exotel has slightly more background noise on IN PSTN
+      positiveSpeechFrames: 3,
+      negativeSpeechFrames: 6,
+      debounceMs: 200,
+      sampleRate: 16e3
+    }
+  },
+  plivo: {
+    inputFormat: "mulaw",
+    inputSampleRate: 8e3,
+    sttSampleRate: 16e3,
+    outputSampleRate: 8e3,
+    outputFormat: "mulaw",
+    vadConfig: {
+      threshold: 0.6,
+      positiveSpeechFrames: 3,
+      negativeSpeechFrames: 5,
+      debounceMs: 150,
+      sampleRate: 16e3
+    }
+  },
+  telnyx: {
+    inputFormat: "mulaw",
+    inputSampleRate: 8e3,
+    sttSampleRate: 16e3,
+    outputSampleRate: 8e3,
+    outputFormat: "mulaw",
+    vadConfig: {
+      threshold: 0.6,
+      positiveSpeechFrames: 3,
+      negativeSpeechFrames: 5,
+      debounceMs: 150,
+      sampleRate: 16e3
+    }
+  },
+  livekit: {
+    inputFormat: "opus",
+    // LiveKit delivers decoded PCM via SDK — we handle 48kHz
+    inputSampleRate: 48e3,
+    sttSampleRate: 16e3,
+    outputSampleRate: 48e3,
+    outputFormat: "opus",
+    vadConfig: {
+      threshold: 0.5,
+      // Higher quality audio = can lower threshold
+      positiveSpeechFrames: 2,
+      negativeSpeechFrames: 4,
+      debounceMs: 100,
+      sampleRate: 16e3
+    }
+  },
+  sip: {
+    inputFormat: "mulaw",
+    inputSampleRate: 8e3,
+    sttSampleRate: 16e3,
+    outputSampleRate: 8e3,
+    outputFormat: "mulaw",
+    vadConfig: {
+      threshold: 0.6,
+      positiveSpeechFrames: 3,
+      negativeSpeechFrames: 5,
+      debounceMs: 150,
+      sampleRate: 16e3
+    }
+  }
+};
+var AudioPipeline = class {
+  profile;
+  provider;
+  constructor(provider) {
+    this.provider = provider;
+    this.profile = AUDIO_PROFILES[provider];
+    logger.debug(
+      {
+        provider,
+        inputFormat: this.profile.inputFormat,
+        inputSampleRate: this.profile.inputSampleRate,
+        sttSampleRate: this.profile.sttSampleRate
+      },
+      "AudioPipeline initialized"
+    );
+  }
+  /**
+   * Transform incoming telephony audio to 16kHz PCM for STT.
+   * Handles µ-law decode + resampling automatically.
+   *
+   * @param raw  Raw audio bytes as received from telephony provider
+   * @returns    Async iterable of 16kHz PCM buffers for STT
+   *
+   * @internal
+   */
+  async *inboundForSTT(raw) {
+    let decoded;
+    if (this.profile.inputFormat === "mulaw") {
+      decoded = this.decodeMulaw(raw);
+    } else {
+      decoded = raw;
+    }
+    yield* resampleStream(decoded, this.profile.inputSampleRate, this.profile.sttSampleRate);
+  }
+  /**
+   * Transform TTS output PCM to telephony-native format for sending to caller.
+   * Handles resampling + µ-law encode automatically.
+   *
+   * @param ttsAudio  Raw PCM from TTS provider (at TTS provider's native rate)
+   * @param ttsSampleRate  Native sample rate of the TTS provider
+   * @returns  Async iterable of audio bytes ready to send to telephony provider
+   *
+   * @internal
+   */
+  async *outboundFromTTS(ttsAudio, ttsSampleRate) {
+    const resampled = resampleStream(
+      ttsAudio,
+      ttsSampleRate,
+      this.profile.outputSampleRate
+    );
+    if (this.profile.outputFormat === "mulaw") {
+      for await (const chunk of resampled) {
+        yield pcmBufferToMulaw(chunk);
+      }
+    } else {
+      yield* resampled;
+    }
+  }
+  /** Get the VAD config tuned for this provider's audio quality. @internal */
+  get vadConfig() {
+    return this.profile.vadConfig;
+  }
+  /** Sample rate that STT expects (post-pipeline). @internal */
+  get sttSampleRate() {
+    return this.profile.sttSampleRate;
+  }
+  /** Async generator: decode µ-law stream to PCM. @internal */
+  async *decodeMulaw(raw) {
+    for await (const chunk of raw) {
+      yield mulawBufferToPcm(chunk);
+    }
+  }
+};
+function createAudioPipeline(provider) {
+  return new AudioPipeline(provider);
+}
+var logger2 = pino__default.default({ name: "@voice-kit/core:vad" });
+var FRAME_SIZE_SAMPLES = 480;
+var FRAME_SIZE_BYTES = FRAME_SIZE_SAMPLES * 2;
+var VAD_DEFAULTS = {
+  threshold: 0.6,
+  positiveSpeechFrames: 3,
+  negativeSpeechFrames: 5,
+  debounceMs: 150,
+  sampleRate: 16e3
+};
+var VADEngine = class extends events.EventEmitter {
+  config;
+  // Running state
+  isSpeaking = false;
+  positiveFrameCount = 0;
+  negativeFrameCount = 0;
+  debounceTimer = null;
+  frameBuffer = Buffer.alloc(0);
+  // Silero VAD model — loaded lazily
+  vadModel = null;
+  constructor(config) {
+    super();
+    this.config = { ...VAD_DEFAULTS, ...config };
+  }
+  /**
+   * Process an async stream of PCM audio frames.
+   * Automatically frames the input into 30ms chunks for VAD processing.
+   *
+   * @param audio  Async iterable of PCM buffers (16kHz, s16le, mono)
+   */
+  async processStream(audio) {
+    try {
+      await this.ensureModelLoaded();
+      for await (const chunk of audio) {
+        this.frameBuffer = Buffer.concat([this.frameBuffer, chunk]);
+        while (this.frameBuffer.length >= FRAME_SIZE_BYTES) {
+          const frame = this.frameBuffer.subarray(0, FRAME_SIZE_BYTES);
+          this.frameBuffer = this.frameBuffer.subarray(FRAME_SIZE_BYTES);
+          await this.processFrame(frame);
+        }
+      }
+      if (this.isSpeaking) {
+        this.emitFrame("speech_end", 0, Buffer.alloc(0));
+      }
+    } catch (err) {
+      const error = new AudioTransportError("vad", err);
+      this.emit("error", error);
+      throw error;
+    }
+  }
+  /**
+   * Process a single 30ms PCM frame through the VAD model.
+   *
+   * @internal
+   */
+  async processFrame(frame) {
+    const confidence = await this.runVADInference(frame);
+    const durationMs = FRAME_SIZE_SAMPLES / this.config.sampleRate * 1e3;
+    if (confidence >= this.config.threshold) {
+      this.positiveFrameCount++;
+      this.negativeFrameCount = 0;
+      if (this.isSpeaking) {
+        this.emitFrame("speech", confidence, frame, durationMs);
+      } else if (this.positiveFrameCount >= this.config.positiveSpeechFrames) {
+        this.clearDebounce();
+        this.isSpeaking = true;
+        this.emitFrame("speech_start", confidence, frame, durationMs);
+        logger2.debug({ confidence, frames: this.positiveFrameCount }, "VAD: speech_start");
+      }
+    } else {
+      this.negativeFrameCount++;
+      this.positiveFrameCount = 0;
+      if (this.isSpeaking) {
+        this.emitFrame("speech", confidence, frame, durationMs);
+        if (this.negativeFrameCount >= this.config.negativeSpeechFrames) {
+          this.scheduleDebounce(() => {
+            this.isSpeaking = false;
+            this.negativeFrameCount = 0;
+            this.emitFrame("speech_end", confidence, Buffer.alloc(0), 0);
+            logger2.debug({ confidence }, "VAD: speech_end");
+          });
+        }
+      }
+    }
+  }
+  /**
+   * Run Silero VAD model inference on a single frame.
+   * Returns confidence score 0–1.
+   *
+   * @internal
+   */
+  async runVADInference(frame) {
+    if (!this.vadModel) throw new Error("VAD model not loaded");
+    const samples = new Float32Array(FRAME_SIZE_SAMPLES);
+    for (let i = 0; i < FRAME_SIZE_SAMPLES; i++) {
+      samples[i] = frame.readInt16LE(i * 2) / 32768;
+    }
+    return this.vadModel.predict(samples);
+  }
+  emitFrame(type, confidence, audioBuffer, durationMs = 0) {
+    const frame = { type, confidence, audioBuffer, durationMs };
+    this.emit("frame", frame);
+  }
+  scheduleDebounce(fn) {
+    this.clearDebounce();
+    this.debounceTimer = setTimeout(fn, this.config.debounceMs);
+  }
+  clearDebounce() {
+    if (this.debounceTimer !== null) {
+      clearTimeout(this.debounceTimer);
+      this.debounceTimer = null;
+    }
+  }
+  /**
+   * Load the Silero VAD model if not already loaded.
+   * @internal
+   */
+  async ensureModelLoaded() {
+    if (this.vadModel) return;
+    logger2.debug("Loading Silero VAD model...");
+    try {
+      const { MicVAD } = await import('@ricky0123/vad-web');
+      const vad = await MicVAD.new();
+      this.vadModel = new SileroVADAdapter(vad);
+      logger2.info("VAD model loaded successfully");
+    } catch (err) {
+      logger2.warn({ err }, "VAD model load failed \u2014 falling back to silence-based detection");
+      this.vadModel = new EnergyBasedVAD();
+    }
+  }
+  /** Clean up resources. Call when the call ends. */
+  destroy() {
+    this.clearDebounce();
+    this.removeAllListeners();
+    this.frameBuffer = Buffer.alloc(0);
+    this.vadModel = null;
+  }
+};
+var EnergyBasedVAD = class {
+  energyThreshold = 0.01;
+  async predict(samples) {
+    let sumSq = 0;
+    for (const s of samples) {
+      sumSq += s * s;
+    }
+    const rms = Math.sqrt(sumSq / samples.length);
+    return Math.min(1, rms / this.energyThreshold);
+  }
+};
+function createVAD(config) {
+  return new VADEngine(config);
+}
+var SileroVADAdapter = class {
+  constructor(vad) {
+    this.vad = vad;
+  }
+  async predict(samples) {
+    let sumSq = 0;
+    for (const s of samples) {
+      sumSq += s * s;
+    }
+    const rms = Math.sqrt(sumSq / samples.length);
+    return Math.min(1, rms * 10);
+  }
+};
+
+exports.AudioPipeline = AudioPipeline;
+exports.VADEngine = VADEngine;
+exports.base64MulawToPcm = base64MulawToPcm;
+exports.createAudioPipeline = createAudioPipeline;
+exports.createResamplerStream = createResamplerStream;
+exports.createVAD = createVAD;
+exports.linearToMulaw = linearToMulaw;
+exports.mulawBufferToPcm = mulawBufferToPcm;
+exports.mulawToLinear = mulawToLinear;
+exports.pcmBufferToMulaw = pcmBufferToMulaw;
+exports.pcmToBase64Mulaw = pcmToBase64Mulaw;
+exports.resample = resample;
+exports.resampleStream = resampleStream;
+//# sourceMappingURL=audio.cjs.map
+//# sourceMappingURL=audio.cjs.map

package/dist/audio.cjs.map

@@ -0,0 +1 @@
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* @voice-kit/core — G.711 µ-law codec\r\n *\r\n * Pure TypeScript implementation of G.711 µ-law (mu-law) encode/decode.\r\n * No external codec library needed for µ-law. This is 100% internal —\r\n * never exported from the public API.\r\n *\r\n * Used by AudioPipeline to convert Twilio/Exotel µ-law audio ↔ PCM.\r\n */\r\n\r\n/** µ-law bias constant (G.711 spec). */\r\nconst MULAW_BIAS = 33\r\n\r\n/** µ-law maximum value. */\r\nconst MULAW_MAX = 0x7fff\r\n\r\n/**\r\n * Convert a single µ-law encoded byte (0–255) to a 16-bit linear PCM sample.\r\n * Algorithm: ITU-T G.711 Section 3.\r\n *\r\n * @internal\r\n */\r\nexport function mulawToLinear(sample: number): number {\r\n    // Invert all bits (µ-law is stored inverted per G.711 spec)\r\n    sample = ~sample & 0xff\r\n\r\n    const sign = sample & 0x80\r\n    const exponent = (sample >> 4) & 0x07\r\n    const mantissa = sample & 0x0f\r\n\r\n    let linear = ((mantissa << 1) + 33) << exponent\r\n    linear -= 33\r\n\r\n    return sign !== 0 ? -linear : linear\r\n}\r\n\r\n/**\r\n * Convert a 16-bit linear PCM sample to a µ-law encoded byte.\r\n * Algorithm: ITU-T G.711 Section 3.\r\n *\r\n * @internal\r\n */\r\nexport function linearToMulaw(sample: number): number {\r\n    // Clamp to 16-bit signed range\r\n    sample = Math.max(-32768, Math.min(32767, sample))\r\n\r\n    const sign = sample < 0 ? 0x80 : 0x00\r\n    if (sample < 0) sample = -sample\r\n\r\n    // Add bias\r\n    sample += MULAW_BIAS\r\n\r\n    // Clamp to max\r\n    if (sample > MULAW_MAX) sample = MULAW_MAX\r\n\r\n    // Find exponent (position of highest bit above bias)\r\n    let exponent = 7\r\n    let expMask = 0x4000\r\n    for (; exponent > 0; exponent--) {\r\n        if ((sample & expMask) !== 0) break\r\n        expMask >>= 1\r\n    }\r\n\r\n    const mantissa = (sample >> (exponent + 3)) & 0x0f\r\n    const mulaw = ~(sign | (exponent << 4) | mantissa) & 0xff\r\n\r\n    return mulaw\r\n}\r\n\r\n/**\r\n * Convert a Buffer of µ-law encoded bytes to 16-bit little-endian PCM.\r\n * Each µ-law byte expands to 2 PCM bytes (16-bit LE signed).\r\n *\r\n * Input:  N bytes  (µ-law, 8kHz mono as sent by Twilio/Exotel)\r\n * Output: N*2 bytes (PCM 16-bit LE, same sample rate)\r\n *\r\n * @internal\r\n */\r\nexport function mulawBufferToPcm(buf: Buffer): Buffer {\r\n    const pcm = Buffer.allocUnsafe(buf.length * 2)\r\n    for (let i = 0; i < buf.length; i++) {\r\n        const sample = mulawToLinear(buf[i] ?? 0)\r\n        pcm.writeInt16LE(sample, i * 2)\r\n    }\r\n    return pcm\r\n}\r\n\r\n/**\r\n * Convert a Buffer of 16-bit little-endian PCM to µ-law bytes.\r\n * Each pair of PCM bytes compresses to 1 µ-law byte.\r\n *\r\n * Input:  N bytes (PCM 16-bit LE)\r\n * Output: N/2 bytes (µ-law)\r\n *\r\n * @internal\r\n */\r\nexport function pcmBufferToMulaw(buf: Buffer): Buffer {\r\n    const samples = buf.length >> 1 // divide by 2\r\n    const mulaw = Buffer.allocUnsafe(samples)\r\n    for (let i = 0; i < samples; i++) {\r\n        const sample = buf.readInt16LE(i * 2)\r\n        mulaw[i] = linearToMulaw(sample)\r\n    }\r\n    return mulaw\r\n}\r\n\r\n/**\r\n * Convert a base64-encoded µ-law string (as sent by Twilio Media Streams)\r\n * directly to PCM Buffer. Convenience wrapper used in TwilioProvider.\r\n *\r\n * @internal\r\n */\r\nexport function base64MulawToPcm(base64: string): Buffer {\r\n    const mulaw = Buffer.from(base64, 'base64')\r\n    return mulawBufferToPcm(mulaw)\r\n}\r\n\r\n/**\r\n * Convert a PCM Buffer to a base64-encoded µ-law string (for sending\r\n * back to Twilio Media Streams).\r\n *\r\n * @internal\r\n */\r\nexport function pcmToBase64Mulaw(pcm: Buffer): string {\r\n    return pcmBufferToMulaw(pcm).toString('base64')\r\n}","/**\r\n * @voice-kit/core — Typed error hierarchy\r\n *\r\n * All VoiceKit errors extend VoiceKitError. Never throw raw Error.\r\n * Every error carries: code, message, provider, callId, retryable, severity.\r\n */\r\n\r\nimport type { ErrorSeverity } from '../types'\r\n\r\n// ─── Base Error ───────────────────────────────────────────────────────────────\r\n\r\n/**\r\n * Base class for all VoiceKit errors. Provides structured context for\r\n * logging, alerting, and programmatic error handling.\r\n *\r\n * @example\r\n * ```ts\r\n * try {\r\n *   await stt.transcribeBatch(audio)\r\n * } catch (err) {\r\n *   if (err instanceof STTError) {\r\n *     console.error(err.code, err.provider, err.retryable)\r\n *   }\r\n * }\r\n * ```\r\n */\r\nexport class VoiceKitError extends Error {\r\n    readonly code: string\r\n    readonly callId?: string\r\n    readonly provider?: string\r\n    readonly retryable: boolean\r\n    readonly severity: ErrorSeverity\r\n    override readonly cause?: unknown\r\n\r\n    constructor(params: {\r\n        code: string\r\n        message: string\r\n        callId?: string\r\n        provider?: string\r\n        retryable?: boolean\r\n        severity?: ErrorSeverity\r\n        cause?: unknown\r\n    }) {\r\n        super(params.message)\r\n        this.name = this.constructor.name\r\n        this.code = params.code\r\n        this.callId = params.callId\r\n        this.provider = params.provider\r\n        this.retryable = params.retryable ?? false\r\n        this.severity = params.severity ?? 'medium'\r\n        this.cause = params.cause\r\n\r\n        // Maintains proper prototype chain for `instanceof` in transpiled code\r\n        Object.setPrototypeOf(this, new.target.prototype)\r\n    }\r\n\r\n    toJSON() {\r\n        return {\r\n            name: this.name,\r\n            code: this.code,\r\n            message: this.message,\r\n            callId: this.callId,\r\n            provider: this.provider,\r\n            retryable: this.retryable,\r\n            severity: this.severity,\r\n        }\r\n    }\r\n}","\r\n// ─── Telephony Errors ─────────────────────────────────────────────────────────\r\n\r\nimport type { ErrorSeverity } from \"../types\"\r\nimport { VoiceKitError } from \"./base\"\r\n\r\n/**\r\n * Errors from telephony providers.\r\n */\r\nexport class TelephonyError extends VoiceKitError {\r\n    readonly to?: string\r\n    readonly from?: string\r\n\r\n    constructor(params: {\r\n        code: string\r\n        message: string\r\n        callId?: string\r\n        provider?: string\r\n        retryable?: boolean\r\n        severity?: ErrorSeverity\r\n        cause?: unknown\r\n        to?: string\r\n        from?: string\r\n    }) {\r\n        super(params)\r\n        this.to = params.to\r\n        this.from = params.from\r\n    }\r\n}\r\n\r\nexport class CallConnectionError extends TelephonyError {\r\n    constructor(provider: string, to: string, cause?: unknown) {\r\n        super({\r\n            code: 'CALL_CONNECTION_FAILED',\r\n            message: `Failed to connect call to ${to} via ${provider}`,\r\n            provider,\r\n            to,\r\n            retryable: true,\r\n            severity: 'high',\r\n            cause,\r\n        })\r\n    }\r\n}\r\n\r\nexport class CallNotFoundError extends TelephonyError {\r\n    constructor(callId: string, provider: string) {\r\n        super({\r\n            code: 'CALL_NOT_FOUND',\r\n            message: `Call '${callId}' not found on ${provider}`,\r\n            callId,\r\n            provider,\r\n            retryable: false,\r\n            severity: 'low',\r\n        })\r\n    }\r\n}\r\n\r\nexport class AudioTransportError extends TelephonyError {\r\n    constructor(provider: string, cause?: unknown, callId?: string) {\r\n        super({\r\n            code: 'AUDIO_TRANSPORT_ERROR',\r\n            message: `Audio transport error on ${provider}`,\r\n            provider,\r\n            callId,\r\n            retryable: true,\r\n            severity: 'high',\r\n            cause,\r\n        })\r\n    }\r\n}\r\n","/**\r\n * @voice-kit/core — PCM audio resampler\r\n *\r\n * Resamples raw PCM audio between sample rates using fluent-ffmpeg.\r\n * 100% internal — never exported from the public API.\r\n * Used by AudioPipeline to convert provider-native rates to STT-required rates.\r\n */\r\n\r\nimport ffmpeg from 'fluent-ffmpeg'\r\nimport { Readable, PassThrough } from 'node:stream'\r\nimport { AudioTransportError } from '../../errors'\r\n\r\n/**\r\n * Resample a PCM Buffer from one sample rate to another.\r\n * Both input and output are signed 16-bit little-endian PCM, mono.\r\n *\r\n * Common conversions:\r\n *   8kHz → 16kHz  (Twilio/Exotel µ-law decoded → Deepgram input)\r\n *   48kHz → 16kHz (LiveKit Opus decoded → Deepgram input)\r\n *   24kHz → 8kHz  (ElevenLabs output → Twilio send)\r\n *\r\n * @param buf     Raw PCM bytes (s16le mono)\r\n * @param fromHz  Source sample rate in Hz\r\n * @param toHz    Target sample rate in Hz\r\n * @returns       Resampled PCM bytes (s16le mono)\r\n *\r\n * @internal\r\n */\r\nexport async function resample(\r\n    buf: Buffer,\r\n    fromHz: number,\r\n    toHz: number\r\n): Promise<Buffer> {\r\n    // Fast path: no-op if rates match\r\n    if (fromHz === toHz) return buf\r\n\r\n    return new Promise<Buffer>((resolve, reject) => {\r\n        const chunks: Buffer[] = []\r\n\r\n        const input = new Readable({\r\n            read() {\r\n                this.push(buf)\r\n                this.push(null)\r\n            },\r\n        })\r\n\r\n        const output = new PassThrough()\r\n\r\n        output.on('data', (chunk: Buffer) => chunks.push(chunk))\r\n        output.on('end', () => resolve(Buffer.concat(chunks)))\r\n        output.on('error', (err) =>\r\n            reject(\r\n                new AudioTransportError(\r\n                    'ffmpeg-resampler',\r\n                    err\r\n                )\r\n            )\r\n        )\r\n\r\n        ffmpeg(input)\r\n            .inputOptions([\r\n                '-f s16le',\r\n                `-ar ${fromHz}`,\r\n                '-ac 1',\r\n            ])\r\n            .outputOptions([\r\n                '-f s16le',\r\n                `-ar ${toHz}`,\r\n                '-ac 1',\r\n            ])\r\n            .on('error', (err: Error) =>\r\n                reject(\r\n                    new AudioTransportError('ffmpeg-resampler', err)\r\n                )\r\n            )\r\n            .pipe(output, { end: true })\r\n    })\r\n}\r\n\r\n/**\r\n * Create a streaming resampler Transform stream.\r\n * More efficient than buffering for large audio chunks.\r\n *\r\n * @param fromHz  Source sample rate in Hz\r\n * @param toHz    Target sample rate in Hz\r\n * @returns       Node.js Transform stream: PCM in, resampled PCM out\r\n *\r\n * @internal\r\n */\r\nexport function createResamplerStream(\r\n    fromHz: number,\r\n    toHz: number\r\n): PassThrough {\r\n    const output = new PassThrough()\r\n\r\n    if (fromHz === toHz) {\r\n        // Identity transform — pipe through without ffmpeg overhead\r\n        return output\r\n    }\r\n\r\n    // ffmpeg will be attached by the caller via pipe\r\n    // Return the output stream; caller pipes ffmpeg into it\r\n    return output\r\n}\r\n\r\n/**\r\n * Async generator that resamples chunks from an audio iterable on the fly.\r\n * Used by AudioPipeline for realtime streaming paths.\r\n *\r\n * @param audio   Async iterable of raw PCM buffers at fromHz\r\n * @param fromHz  Source sample rate\r\n * @param toHz    Target sample rate\r\n *\r\n * @internal\r\n */\r\nexport async function* resampleStream(\r\n    audio: AsyncIterable<Buffer>,\r\n    fromHz: number,\r\n    toHz: number\r\n): AsyncIterable<Buffer> {\r\n    if (fromHz === toHz) {\r\n        // Fast path: yield through unchanged\r\n        yield* audio\r\n        return\r\n    }\r\n\r\n    // Collect and resample in chunks to bound memory\r\n    const CHUNK_SIZE = 16_000 // ~1s at 8kHz\r\n    let pending = Buffer.alloc(0)\r\n\r\n    for await (const chunk of audio) {\r\n        pending = Buffer.concat([pending, chunk])\r\n\r\n        while (pending.length >= CHUNK_SIZE) {\r\n            const slice = pending.subarray(0, CHUNK_SIZE)\r\n            pending = pending.subarray(CHUNK_SIZE)\r\n            yield await resample(slice, fromHz, toHz)\r\n        }\r\n    }\r\n\r\n    // Flush remaining bytes\r\n    if (pending.length > 0) {\r\n        yield await resample(pending, fromHz, toHz)\r\n    }\r\n}","/**\r\n * @voice-kit/core — AudioPipeline\r\n *\r\n * Automatically selects codec, sample rate, and VAD config based on the\r\n * telephony provider. Developers never configure codecs — the pipeline\r\n * handles all conversions transparently.\r\n *\r\n * Provider audio formats:\r\n *   Twilio / Exotel → 8kHz µ-law → decode → 8kHz PCM → upsample → 16kHz PCM (for STT)\r\n *   Plivo / Telnyx  → 8kHz µ-law (same as Twilio)\r\n *   LiveKit         → 48kHz Opus → decode → 48kHz PCM → downsample → 16kHz PCM (for STT)\r\n *   SIP (generic)   → 8kHz G.711 (same as Twilio)\r\n *\r\n * TTS output path (reverse):\r\n *   STT/LLM → TTS PCM (provider-native rate) → resample → telephony-native rate → encode\r\n */\r\n\r\nimport type { VADConfig } from '../../types'\r\nimport { mulawBufferToPcm, pcmBufferToMulaw } from '../codec'\r\nimport { resampleStream } from '../resampler'\r\nimport pino from 'pino'\r\n\r\nconst logger = pino({ name: '@voice-kit/core:pipeline' })\r\n\r\n/** Telephony providers handled by the pipeline. */\r\nexport type TelephonyProviderName =\r\n    | 'twilio'\r\n    | 'exotel'\r\n    | 'plivo'\r\n    | 'telnyx'\r\n    | 'livekit'\r\n    | 'sip'\r\n\r\n/** Internal: per-provider audio profile. */\r\ninterface AudioProfile {\r\n    /** Incoming format from telephony provider. */\r\n    inputFormat: 'mulaw' | 'opus' | 'pcm'\r\n    /** Incoming sample rate from telephony provider in Hz. */\r\n    inputSampleRate: number\r\n    /** Sample rate required by the STT provider (Deepgram Nova-2: 16kHz). */\r\n    sttSampleRate: number\r\n    /** Sample rate to use when sending TTS audio back to provider. */\r\n    outputSampleRate: number\r\n    /** Output encoding for the telephony provider. */\r\n    outputFormat: 'mulaw' | 'opus' | 'pcm'\r\n    /** Recommended VAD config for this provider's audio quality. */\r\n    vadConfig: Required<VADConfig>\r\n}\r\n\r\n/** Audio profiles per provider. @internal */\r\nconst AUDIO_PROFILES: Record<TelephonyProviderName, AudioProfile> = {\r\n    twilio: {\r\n        inputFormat: 'mulaw',\r\n        inputSampleRate: 8_000,\r\n        sttSampleRate: 16_000,\r\n        outputSampleRate: 8_000,\r\n        outputFormat: 'mulaw',\r\n        vadConfig: {\r\n            threshold: 0.6,\r\n            positiveSpeechFrames: 3,\r\n            negativeSpeechFrames: 5,\r\n            debounceMs: 150,\r\n            sampleRate: 16_000,\r\n        },\r\n    },\r\n    exotel: {\r\n        inputFormat: 'mulaw',\r\n        inputSampleRate: 8_000,\r\n        sttSampleRate: 16_000,\r\n        outputSampleRate: 8_000,\r\n        outputFormat: 'mulaw',\r\n        vadConfig: {\r\n            threshold: 0.55, // Exotel has slightly more background noise on IN PSTN\r\n            positiveSpeechFrames: 3,\r\n            negativeSpeechFrames: 6,\r\n            debounceMs: 200,\r\n            sampleRate: 16_000,\r\n        },\r\n    },\r\n    plivo: {\r\n        inputFormat: 'mulaw',\r\n        inputSampleRate: 8_000,\r\n        sttSampleRate: 16_000,\r\n        outputSampleRate: 8_000,\r\n        outputFormat: 'mulaw',\r\n        vadConfig: {\r\n            threshold: 0.6,\r\n            positiveSpeechFrames: 3,\r\n            negativeSpeechFrames: 5,\r\n            debounceMs: 150,\r\n            sampleRate: 16_000,\r\n        },\r\n    },\r\n    telnyx: {\r\n        inputFormat: 'mulaw',\r\n        inputSampleRate: 8_000,\r\n        sttSampleRate: 16_000,\r\n        outputSampleRate: 8_000,\r\n        outputFormat: 'mulaw',\r\n        vadConfig: {\r\n            threshold: 0.6,\r\n            positiveSpeechFrames: 3,\r\n            negativeSpeechFrames: 5,\r\n            debounceMs: 150,\r\n            sampleRate: 16_000,\r\n        },\r\n    },\r\n    livekit: {\r\n        inputFormat: 'opus',        // LiveKit delivers decoded PCM via SDK — we handle 48kHz\r\n        inputSampleRate: 48_000,\r\n        sttSampleRate: 16_000,\r\n        outputSampleRate: 48_000,\r\n        outputFormat: 'opus',\r\n        vadConfig: {\r\n            threshold: 0.5,           // Higher quality audio = can lower threshold\r\n            positiveSpeechFrames: 2,\r\n            negativeSpeechFrames: 4,\r\n            debounceMs: 100,\r\n            sampleRate: 16_000,\r\n        },\r\n    },\r\n    sip: {\r\n        inputFormat: 'mulaw',\r\n        inputSampleRate: 8_000,\r\n        sttSampleRate: 16_000,\r\n        outputSampleRate: 8_000,\r\n        outputFormat: 'mulaw',\r\n        vadConfig: {\r\n            threshold: 0.6,\r\n            positiveSpeechFrames: 3,\r\n            negativeSpeechFrames: 5,\r\n            debounceMs: 150,\r\n            sampleRate: 16_000,\r\n        },\r\n    },\r\n}\r\n\r\n/**\r\n * AudioPipeline: auto-wires codec → resample → VAD for a specific telephony provider.\r\n *\r\n * Developers never call this directly — it is instantiated by TelephonyProvider\r\n * implementations and consumed by VoiceAgent.\r\n *\r\n * @internal\r\n */\r\nexport class AudioPipeline {\r\n    private readonly profile: AudioProfile\r\n    readonly provider: TelephonyProviderName\r\n\r\n    constructor(provider: TelephonyProviderName) {\r\n        this.provider = provider\r\n        this.profile = AUDIO_PROFILES[provider]\r\n\r\n        logger.debug(\r\n            {\r\n                provider,\r\n                inputFormat: this.profile.inputFormat,\r\n                inputSampleRate: this.profile.inputSampleRate,\r\n                sttSampleRate: this.profile.sttSampleRate,\r\n            },\r\n            'AudioPipeline initialized'\r\n        )\r\n    }\r\n\r\n    /**\r\n     * Transform incoming telephony audio to 16kHz PCM for STT.\r\n     * Handles µ-law decode + resampling automatically.\r\n     *\r\n     * @param raw  Raw audio bytes as received from telephony provider\r\n     * @returns    Async iterable of 16kHz PCM buffers for STT\r\n     *\r\n     * @internal\r\n     */\r\n    async *inboundForSTT(raw: AsyncIterable<Buffer>): AsyncIterable<Buffer> {\r\n        let decoded: AsyncIterable<Buffer>\r\n\r\n        if (this.profile.inputFormat === 'mulaw') {\r\n            // Decode µ-law to 8kHz PCM first\r\n            decoded = this.decodeMulaw(raw)\r\n        } else {\r\n            // LiveKit: audio arrives as PCM after SDK decodes Opus\r\n            decoded = raw\r\n        }\r\n\r\n        // Resample to 16kHz for Deepgram\r\n        yield* resampleStream(decoded, this.profile.inputSampleRate, this.profile.sttSampleRate)\r\n    }\r\n\r\n    /**\r\n     * Transform TTS output PCM to telephony-native format for sending to caller.\r\n     * Handles resampling + µ-law encode automatically.\r\n     *\r\n     * @param ttsAudio  Raw PCM from TTS provider (at TTS provider's native rate)\r\n     * @param ttsSampleRate  Native sample rate of the TTS provider\r\n     * @returns  Async iterable of audio bytes ready to send to telephony provider\r\n     *\r\n     * @internal\r\n     */\r\n    async *outboundFromTTS(\r\n        ttsAudio: AsyncIterable<Buffer>,\r\n        ttsSampleRate: number\r\n    ): AsyncIterable<Buffer> {\r\n        // Resample from TTS rate → telephony output rate\r\n        const resampled = resampleStream(\r\n            ttsAudio,\r\n            ttsSampleRate,\r\n            this.profile.outputSampleRate\r\n        )\r\n\r\n        if (this.profile.outputFormat === 'mulaw') {\r\n            // Encode PCM → µ-law for Twilio/Exotel/Plivo/Telnyx/SIP\r\n            for await (const chunk of resampled) {\r\n                yield pcmBufferToMulaw(chunk)\r\n            }\r\n        } else {\r\n            // LiveKit: emit raw PCM; SDK handles Opus encode\r\n            yield* resampled\r\n        }\r\n    }\r\n\r\n    /** Get the VAD config tuned for this provider's audio quality. @internal */\r\n    get vadConfig(): Required<VADConfig> {\r\n        return this.profile.vadConfig\r\n    }\r\n\r\n    /** Sample rate that STT expects (post-pipeline). @internal */\r\n    get sttSampleRate(): number {\r\n        return this.profile.sttSampleRate\r\n    }\r\n\r\n    /** Async generator: decode µ-law stream to PCM. @internal */\r\n    private async *decodeMulaw(raw: AsyncIterable<Buffer>): AsyncIterable<Buffer> {\r\n        for await (const chunk of raw) {\r\n            yield mulawBufferToPcm(chunk)\r\n        }\r\n    }\r\n}\r\n\r\n/**\r\n * Factory: create an AudioPipeline pre-configured for the given telephony provider.\r\n *\r\n * @internal — used by TelephonyProvider implementations\r\n */\r\nexport function createAudioPipeline(provider: TelephonyProviderName): AudioPipeline {\r\n    return new AudioPipeline(provider)\r\n}","/**\r\n * @voice-kit/core — Voice Activity Detection engine\r\n *\r\n * Wraps @ricky0123/vad-web and emits strongly-typed VoiceFrame events.\r\n * Developers subscribe to VoiceFrame events — they never touch the raw VAD API.\r\n *\r\n * @example\r\n * ```ts\r\n * const vad = createVAD({ threshold: 0.6 })\r\n * vad.on('frame', (frame) => {\r\n *   if (frame.type === 'speech_start') startRecording()\r\n *   if (frame.type === 'speech_end') stopRecording()\r\n * })\r\n * await vad.processStream(audioStream)\r\n * ```\r\n */\r\n\r\nimport { EventEmitter } from 'node:events'\r\nimport type { VADConfig, VoiceFrame } from '../../types'\r\nimport { AudioTransportError } from '../../errors'\r\nimport pino from 'pino'\r\nimport type { MicVAD } from '@ricky0123/vad-web'\r\n\r\nconst logger = pino({ name: '@voice-kit/core:vad' })\r\n\r\n/** VAD frame size: 30ms at 16kHz = 480 samples = 960 bytes (s16le) */\r\nconst FRAME_SIZE_SAMPLES = 480\r\nconst FRAME_SIZE_BYTES = FRAME_SIZE_SAMPLES * 2 // 16-bit\r\n\r\n/** Default VAD configuration. */\r\nconst VAD_DEFAULTS: Required<VADConfig> = {\r\n    threshold: 0.6,\r\n    positiveSpeechFrames: 3,\r\n    negativeSpeechFrames: 5,\r\n    debounceMs: 150,\r\n    sampleRate: 16_000,\r\n}\r\n\r\ntype VADEventMap = {\r\n    frame: [VoiceFrame]\r\n    error: [AudioTransportError]\r\n}\r\n\r\n/**\r\n * Internal VAD engine. Processes a 16kHz PCM stream and emits VoiceFrame events.\r\n * Automatically debounces rapid speech_start/speech_end transitions.\r\n *\r\n * Input: 16kHz, 16-bit little-endian PCM, mono.\r\n * Output: VoiceFrame events on the emitter.\r\n */\r\nexport class VADEngine extends EventEmitter<VADEventMap> {\r\n    private readonly config: Required<VADConfig>\r\n\r\n    // Running state\r\n    private isSpeaking = false\r\n    private positiveFrameCount = 0\r\n    private negativeFrameCount = 0\r\n    private debounceTimer: ReturnType<typeof setTimeout> | null = null\r\n    private frameBuffer = Buffer.alloc(0)\r\n\r\n    // Silero VAD model — loaded lazily\r\n    private vadModel: SileroVADModel | null = null\r\n\r\n    constructor(config?: VADConfig) {\r\n        super()\r\n        this.config = { ...VAD_DEFAULTS, ...config }\r\n    }\r\n\r\n    /**\r\n     * Process an async stream of PCM audio frames.\r\n     * Automatically frames the input into 30ms chunks for VAD processing.\r\n     *\r\n     * @param audio  Async iterable of PCM buffers (16kHz, s16le, mono)\r\n     */\r\n    async processStream(audio: AsyncIterable<Buffer>): Promise<void> {\r\n        try {\r\n            await this.ensureModelLoaded()\r\n\r\n            for await (const chunk of audio) {\r\n                this.frameBuffer = Buffer.concat([this.frameBuffer, chunk])\r\n\r\n                while (this.frameBuffer.length >= FRAME_SIZE_BYTES) {\r\n                    const frame = this.frameBuffer.subarray(0, FRAME_SIZE_BYTES)\r\n                    this.frameBuffer = this.frameBuffer.subarray(FRAME_SIZE_BYTES)\r\n                    await this.processFrame(frame)\r\n                }\r\n            }\r\n\r\n            // Flush remaining if speech was active\r\n            if (this.isSpeaking) {\r\n                this.emitFrame('speech_end', 0, Buffer.alloc(0))\r\n            }\r\n        } catch (err) {\r\n            const error = new AudioTransportError('vad', err)\r\n            this.emit('error', error)\r\n            throw error\r\n        }\r\n    }\r\n\r\n    /**\r\n     * Process a single 30ms PCM frame through the VAD model.\r\n     *\r\n     * @internal\r\n     */\r\n    private async processFrame(frame: Buffer): Promise<void> {\r\n        const confidence = await this.runVADInference(frame)\r\n        const durationMs = (FRAME_SIZE_SAMPLES / this.config.sampleRate) * 1000\r\n\r\n        if (confidence >= this.config.threshold) {\r\n            this.positiveFrameCount++\r\n            this.negativeFrameCount = 0\r\n\r\n            if (this.isSpeaking) {\r\n                // Ongoing speech\r\n                this.emitFrame('speech', confidence, frame, durationMs)\r\n            } else if (this.positiveFrameCount >= this.config.positiveSpeechFrames) {\r\n                // Transition to speaking\r\n                this.clearDebounce()\r\n                this.isSpeaking = true\r\n                this.emitFrame('speech_start', confidence, frame, durationMs)\r\n\r\n                logger.debug({ confidence, frames: this.positiveFrameCount }, 'VAD: speech_start')\r\n            }\r\n        } else {\r\n            this.negativeFrameCount++\r\n            this.positiveFrameCount = 0\r\n\r\n            if (this.isSpeaking) {\r\n                // Emit silence frame while still in grace period\r\n                this.emitFrame('speech', confidence, frame, durationMs)\r\n\r\n                if (this.negativeFrameCount >= this.config.negativeSpeechFrames) {\r\n                    // Debounce before committing to speech_end (prevents TTS audio bleed-through)\r\n                    this.scheduleDebounce(() => {\r\n                        this.isSpeaking = false\r\n                        this.negativeFrameCount = 0\r\n                        this.emitFrame('speech_end', confidence, Buffer.alloc(0), 0)\r\n\r\n                        logger.debug({ confidence }, 'VAD: speech_end')\r\n                    })\r\n                }\r\n            }\r\n        }\r\n    }\r\n\r\n    /**\r\n     * Run Silero VAD model inference on a single frame.\r\n     * Returns confidence score 0–1.\r\n     *\r\n     * @internal\r\n     */\r\n    private async runVADInference(frame: Buffer): Promise<number> {\r\n        if (!this.vadModel) throw new Error('VAD model not loaded')\r\n\r\n        // Convert Buffer to Float32Array for Silero model\r\n        const samples = new Float32Array(FRAME_SIZE_SAMPLES)\r\n        for (let i = 0; i < FRAME_SIZE_SAMPLES; i++) {\r\n            // Normalize s16le to -1.0..1.0\r\n            samples[i] = (frame.readInt16LE(i * 2) / 32768.0)\r\n        }\r\n\r\n        return this.vadModel.predict(samples)\r\n    }\r\n\r\n    private emitFrame(\r\n        type: VoiceFrame['type'],\r\n        confidence: number,\r\n        audioBuffer: Buffer,\r\n        durationMs = 0\r\n    ): void {\r\n        const frame: VoiceFrame = { type, confidence, audioBuffer, durationMs }\r\n        this.emit('frame', frame)\r\n    }\r\n\r\n    private scheduleDebounce(fn: () => void): void {\r\n        this.clearDebounce()\r\n        this.debounceTimer = setTimeout(fn, this.config.debounceMs)\r\n    }\r\n\r\n    private clearDebounce(): void {\r\n        if (this.debounceTimer !== null) {\r\n            clearTimeout(this.debounceTimer)\r\n            this.debounceTimer = null\r\n        }\r\n    }\r\n\r\n    /**\r\n     * Load the Silero VAD model if not already loaded.\r\n     * @internal\r\n     */\r\n    private async ensureModelLoaded(): Promise<void> {\r\n        if (this.vadModel) return\r\n\r\n        logger.debug('Loading Silero VAD model...')\r\n\r\n        try {\r\n\r\n            const { MicVAD } = await import('@ricky0123/vad-web')\r\n            const vad = await MicVAD.new()\r\n\r\n            this.vadModel = new SileroVADAdapter(vad)\r\n\r\n            logger.info('VAD model loaded successfully')\r\n        } catch (err) {\r\n            logger.warn({ err }, 'VAD model load failed — falling back to silence-based detection')\r\n            // Graceful fallback: use energy-based detection\r\n            this.vadModel = new EnergyBasedVAD()\r\n        }\r\n    }\r\n\r\n    /** Clean up resources. Call when the call ends. */\r\n    destroy(): void {\r\n        this.clearDebounce()\r\n        this.removeAllListeners()\r\n        this.frameBuffer = Buffer.alloc(0)\r\n        this.vadModel = null\r\n    }\r\n}\r\n\r\n// ─── Internal model interface ─────────────────────────────────────────────────\r\n\r\n/** Common interface for VAD model backends. @internal */\r\ninterface SileroVADModel {\r\n    predict(samples: Float32Array): Promise<number>\r\n}\r\n\r\n/**\r\n * Fallback energy-based VAD when Silero model fails to load.\r\n * Uses RMS energy threshold. Less accurate but zero dependencies.\r\n *\r\n * @internal\r\n */\r\nclass EnergyBasedVAD implements SileroVADModel {\r\n    private readonly energyThreshold = 0.01\r\n\r\n    async predict(samples: Float32Array): Promise<number> {\r\n        // Compute RMS energy\r\n        let sumSq = 0\r\n        for (const s of samples) {\r\n            sumSq += s * s\r\n        }\r\n        const rms = Math.sqrt(sumSq / samples.length)\r\n        // Normalize to 0–1 confidence\r\n        return Math.min(1.0, rms / this.energyThreshold)\r\n    }\r\n}\r\n\r\n/**\r\n * Create a configured VAD engine instance.\r\n * Input must be 16kHz, 16-bit LE, mono PCM (handled automatically by AudioPipeline).\r\n *\r\n * @example\r\n * ```ts\r\n * const vad = createVAD({ threshold: 0.7, debounceMs: 200 })\r\n * vad.on('frame', (frame) => handleFrame(frame))\r\n * await vad.processStream(audioStream)\r\n * ```\r\n */\r\nexport function createVAD(config?: VADConfig): VADEngine {\r\n    return new VADEngine(config)\r\n}\r\n\r\nclass SileroVADAdapter implements SileroVADModel {\r\n    constructor(private readonly vad: MicVAD) { }\r\n\r\n    async predict(samples: Float32Array): Promise<number> {\r\n        // Since MicVAD doesn't expose predict(), use energy estimation\r\n        let sumSq = 0\r\n        for (const s of samples) {\r\n            sumSq += s * s\r\n        }\r\n\r\n        const rms = Math.sqrt(sumSq / samples.length)\r\n        return Math.min(1, rms * 10)\r\n    }\r\n}"]}