@happyvertical/speech 0.102.4 → 0.102.5

package/dist/chunks/{wav-B0Djpj2r.js → wav-vzrhhZbu.js}

@@ -48,6 +48,255 @@ function resampleMono(samples, fromRate, toRate) {
 	return out;
 }
 //#endregion
+//#region src/shared/vad.ts
+var VAD_DEFAULTS = {
+	silenceMs: 800,
+	minSpeechMs: 150,
+	preRollMs: 300,
+	maxUtteranceMs: 3e4,
+	sensitivity: .5,
+	calibrateMs: 300
+};
+/** Length of one analysed frame. */
+var VAD_FRAME_MS = 20;
+/** The quietest noise floor assumed (digital silence must not make everything "speech"). */
+var MIN_FLOOR = .0015;
+/** Floor assumed before the room has been heard (a conservative, quiet-room value). */
+var DEFAULT_FLOOR = .003;
+/**
+* Highest floor calibration may settle on. If the opening frames are louder
+* than this they are taken to be speech (or a very noisy room) and the floor
+* stays conservative instead of climbing to speech level.
+*/
+var CALIBRATION_FLOOR_CAP = .006;
+/** Calibration uses this low percentile of the opening energies, not the mean. */
+var CALIBRATION_PERCENTILE = .2;
+/** Highest floor slow tracking may reach between utterances. */
+var MAX_FLOOR = .02;
+/** Time constant of the slow noise-floor tracking, in ms. */
+var FLOOR_TRACK_MS = 2e3;
+/** Silence kept at the end of an utterance when it is trimmed, in ms. */
+var TRAILING_KEEP_MS = 250;
+function clamp(value, min, max) {
+	return Math.min(max, Math.max(min, value));
+}
+/** Resolve options against the defaults, clamping silly values. */
+function resolveVadOptions(options = {}) {
+	const pick = (value, fallback, min) => typeof value === "number" && Number.isFinite(value) ? Math.max(min, value) : fallback;
+	return {
+		silenceMs: pick(options.silenceMs, VAD_DEFAULTS.silenceMs, 20),
+		minSpeechMs: pick(options.minSpeechMs, VAD_DEFAULTS.minSpeechMs, 0),
+		preRollMs: pick(options.preRollMs, VAD_DEFAULTS.preRollMs, 0),
+		maxUtteranceMs: pick(options.maxUtteranceMs, VAD_DEFAULTS.maxUtteranceMs, 1e3),
+		sensitivity: clamp(typeof options.sensitivity === "number" && Number.isFinite(options.sensitivity) ? options.sensitivity : VAD_DEFAULTS.sensitivity, 0, 1),
+		calibrateMs: pick(options.calibrateMs, VAD_DEFAULTS.calibrateMs, 0)
+	};
+}
+/** Root-mean-square loudness of a block of samples (-1 to 1). */
+function frameEnergy(samples) {
+	const n = samples.length;
+	if (n === 0) return 0;
+	let sum = 0;
+	for (let i = 0; i < n; i++) {
+		const s = samples[i];
+		sum += s * s;
+	}
+	return Math.sqrt(sum / n);
+}
+function createVadSegmenter(options = {}) {
+	const o = resolveVadOptions(options);
+	const ratio = 6 - 4 * o.sensitivity;
+	const absMin = .02 - .015 * o.sensitivity;
+	let floor = DEFAULT_FLOOR;
+	let calibratedMs = 0;
+	const calibrationEnergies = [];
+	let speaking = false;
+	let candidateMs = 0;
+	let silenceMs = 0;
+	let utteranceMs = 0;
+	const onThreshold = () => Math.max(floor * ratio, absMin);
+	const offThreshold = () => Math.max(onThreshold() * .6, floor * 1.5);
+	const reset = () => {
+		speaking = false;
+		candidateMs = 0;
+		silenceMs = 0;
+		utteranceMs = 0;
+	};
+	return {
+		get speaking() {
+			return speaking;
+		},
+		get noiseFloor() {
+			return floor;
+		},
+		get threshold() {
+			return onThreshold();
+		},
+		push(energy, frameMs = 20) {
+			const events = [];
+			const e = Number.isFinite(energy) ? Math.max(0, energy) : 0;
+			const level = clamp(e / (onThreshold() * 3), 0, 1);
+			const calibrating = calibratedMs < o.calibrateMs;
+			if (calibrating) {
+				calibratedMs += frameMs;
+				if (!speaking) {
+					calibrationEnergies.push(e);
+					const sorted = [...calibrationEnergies].sort((a, b) => a - b);
+					floor = clamp(sorted[Math.min(sorted.length - 1, Math.floor(sorted.length * CALIBRATION_PERCENTILE))] ?? DEFAULT_FLOOR, MIN_FLOOR, CALIBRATION_FLOOR_CAP);
+				}
+			}
+			if (!speaking) {
+				if (e > onThreshold()) {
+					candidateMs += frameMs;
+					if (candidateMs >= o.minSpeechMs) {
+						speaking = true;
+						silenceMs = 0;
+						utteranceMs = candidateMs;
+						events.push({
+							type: "start",
+							leadMs: o.preRollMs + candidateMs
+						});
+						candidateMs = 0;
+					}
+				} else if (!calibrating) {
+					candidateMs = 0;
+					const alpha = frameMs / FLOOR_TRACK_MS * (e < floor ? 4 : 1);
+					floor = clamp(floor + (e - floor) * Math.min(1, alpha), MIN_FLOOR, MAX_FLOOR);
+				} else candidateMs = 0;
+				return {
+					events,
+					level,
+					speaking
+				};
+			}
+			utteranceMs += frameMs;
+			if (e > offThreshold()) silenceMs = 0;
+			else silenceMs += frameMs;
+			if (silenceMs >= o.silenceMs) {
+				events.push({
+					type: "end",
+					reason: "silence",
+					trailingSilenceMs: silenceMs
+				});
+				reset();
+			} else if (utteranceMs >= o.maxUtteranceMs) {
+				events.push({ type: "split" });
+				utteranceMs = 0;
+				silenceMs = 0;
+			}
+			return {
+				events,
+				level,
+				speaking
+			};
+		},
+		flush() {
+			const events = [];
+			if (speaking) events.push({
+				type: "end",
+				reason: "flush",
+				trailingSilenceMs: silenceMs
+			});
+			reset();
+			return events;
+		}
+	};
+}
+function createUtteranceCollector(options) {
+	const o = resolveVadOptions(options);
+	const segmenter = createVadSegmenter(o);
+	const frameSize = Math.max(1, Math.round(options.sampleRate * 20 / 1e3));
+	const frameMs = frameSize / options.sampleRate * 1e3;
+	const ringFrames = Math.ceil((o.preRollMs + o.minSpeechMs) / frameMs) + 1;
+	const ring = [];
+	let carry = /* @__PURE__ */ new Float32Array(0);
+	let utterance = null;
+	let wasSpeaking = false;
+	const emit = (frames, reason, trailMs) => {
+		const dropFrames = Math.max(0, Math.floor((trailMs - TRAILING_KEEP_MS) / frameMs));
+		const kept = frames.slice(0, Math.max(1, frames.length - dropFrames));
+		let total = 0;
+		for (const f of kept) total += f.length;
+		const pcm = new Float32Array(total);
+		let at = 0;
+		for (const f of kept) {
+			pcm.set(f, at);
+			at += f.length;
+		}
+		options.onUtterance({
+			pcm,
+			sampleRate: options.sampleRate,
+			durationMs: total / options.sampleRate * 1e3,
+			reason
+		});
+	};
+	const setSpeaking = (speaking) => {
+		if (speaking === wasSpeaking) return;
+		wasSpeaking = speaking;
+		options.onSpeaking?.(speaking);
+	};
+	const handleFrame = (frame) => {
+		const step = segmenter.push(frameEnergy(frame), frameMs);
+		options.onLevel?.(step.level);
+		ring.push(frame);
+		if (ring.length > ringFrames) ring.shift();
+		if (utterance) utterance.push(frame);
+		for (const event of step.events) if (event.type === "start") {
+			utterance = [...ring];
+			const wantLead = Math.ceil(o.preRollMs / frameMs);
+			const haveLead = Math.max(0, ring.length - Math.ceil(o.minSpeechMs / frameMs));
+			for (let i = haveLead; i < wantLead; i++) utterance.unshift(new Float32Array(frameSize));
+		} else if (event.type === "end") {
+			const frames = utterance ?? [];
+			utterance = null;
+			ring.length = 0;
+			if (frames.length > 0) emit(frames, event.reason, event.trailingSilenceMs);
+		} else if (event.type === "split") {
+			const frames = utterance ?? [];
+			utterance = [];
+			if (frames.length > 0) emit(frames, "max", 0);
+		}
+		setSpeaking(segmenter.speaking);
+	};
+	return {
+		get speaking() {
+			return segmenter.speaking;
+		},
+		segmenter,
+		push(samples) {
+			let data = samples;
+			if (carry.length > 0) {
+				data = new Float32Array(carry.length + samples.length);
+				data.set(carry, 0);
+				data.set(samples, carry.length);
+			}
+			let at = 0;
+			while (at + frameSize <= data.length) {
+				handleFrame(data.slice(at, at + frameSize));
+				at += frameSize;
+			}
+			carry = data.slice(at);
+		},
+		flush() {
+			const events = segmenter.flush();
+			for (const event of events) if (event.type === "end" && utterance && utterance.length > 0) emit(utterance, "flush", event.trailingSilenceMs);
+			utterance = null;
+			ring.length = 0;
+			carry = /* @__PURE__ */ new Float32Array(0);
+			setSpeaking(false);
+			options.onLevel?.(0);
+		},
+		discard() {
+			segmenter.flush();
+			utterance = null;
+			ring.length = 0;
+			carry = /* @__PURE__ */ new Float32Array(0);
+			setSpeaking(false);
+			options.onLevel?.(0);
+		}
+	};
+}
+//#endregion
 //#region src/shared/wav.ts
 /**
 * Pure, dependency-free 16-bit PCM WAV helpers. No Node APIs and no DOM: they
@@ -227,6 +476,6 @@ function parseWavPcm16(bytes, expect = {}) {
 	};
 }
 //#endregion
-export { MAX_RESAMPLE_OUTPUT_SAMPLES as a, pcm16ToFloat32 as i, float32ToPcm16 as n, resampleMono as o, parseWavPcm16 as r, encodeWavPcm16 as t };
+export { VAD_DEFAULTS as a, createVadSegmenter as c, MAX_RESAMPLE_OUTPUT_SAMPLES as d, resampleMono as f, pcm16ToFloat32 as i, frameEnergy as l, float32ToPcm16 as n, VAD_FRAME_MS as o, parseWavPcm16 as r, createUtteranceCollector as s, encodeWavPcm16 as t, resolveVadOptions as u };
 
-//# sourceMappingURL=wav-B0Djpj2r.js.map
+//# sourceMappingURL=wav-vzrhhZbu.js.map

package/dist/chunks/wav-vzrhhZbu.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"wav-vzrhhZbu.js","names":[],"sources":["../../src/shared/resample.ts","../../src/shared/vad.ts","../../src/shared/wav.ts"],"sourcesContent":["/**\n * Mono resampler: windowed-sinc (Hann, 16 zero crossings per side)\n * interpolation. When downsampling, the kernel's cutoff is scaled to 95% of\n * the target Nyquist frequency so content above it is attenuated before it can\n * alias; when upsampling it is a plain band-limited interpolator. Edges treat\n * missing samples as absent and renormalise the kernel, so DC gain stays 1.\n * Quality is suited to speech recognition (passband flat to roughly 90% of\n * Nyquist, stopband well below -40 dB), not mastering. Cost is\n * O(output length x taps), taps ~ 32 x max(1, fromRate / toRate).\n */\n\nimport { WavFormatError } from './errors.js';\n\n/** Largest output a single call may produce (samples). */\nexport const MAX_RESAMPLE_OUTPUT_SAMPLES = 2 ** 28;\n\nconst ZERO_CROSSINGS = 16;\nconst DOWNSAMPLE_CUTOFF = 0.95;\n\nfunction sinc(x: number): number {\n  if (x === 0) return 1;\n  const px = Math.PI * x;\n  return Math.sin(px) / px;\n}\n\n/** Resamples mono float samples from `fromRate` to `toRate` (Hz). */\nexport function resampleMono(\n  samples: Float32Array,\n  fromRate: number,\n  toRate: number,\n): Float32Array {\n  for (const [name, rate] of [\n    ['fromRate', fromRate],\n    ['toRate', toRate],\n  ] as const) {\n    if (!Number.isFinite(rate) || rate <= 0) {\n      throw new WavFormatError(\n        'invalid_argument',\n        `${name} must be a positive finite number (got ${String(rate)})`,\n      );\n    }\n  }\n  const length = Math.floor((samples.length * toRate) / fromRate);\n  if (length > MAX_RESAMPLE_OUTPUT_SAMPLES) {\n    throw new WavFormatError(\n      'invalid_argument',\n      `resampled output would be ${length} samples; the limit is ${MAX_RESAMPLE_OUTPUT_SAMPLES}`,\n    );\n  }\n  if (fromRate === toRate) return samples.slice();\n  const out = new Float32Array(length);\n  const step = fromRate / toRate;\n  // Cutoff relative to the input Nyquist; 1 means no low-pass.\n  const cutoff = step > 1 ? DOWNSAMPLE_CUTOFF / step : 1;\n  const halfWidth = ZERO_CROSSINGS / cutoff;\n  const last = samples.length - 1;\n\n  for (let index = 0; index < length; index += 1) {\n    const center = index * step;\n    const first = Math.max(0, Math.ceil(center - halfWidth));\n    const end = Math.min(last, Math.floor(center + halfWidth));\n    let sum = 0;\n    let weights = 0;\n    for (let tap = first; tap <= end; tap += 1) {\n      const distance = tap - center;\n      const window = 0.5 + 0.5 * Math.cos((Math.PI * distance) / halfWidth);\n      const weight = cutoff * sinc(cutoff * distance) * window;\n      sum += weight * samples[tap];\n      weights += weight;\n    }\n    out[index] = weights !== 0 ? sum / weights : 0;\n  }\n  return out;\n}\n","/**\n * Voice activity detection for hands-free capture.\n *\n * Energy based, no model and no download: the loudness (RMS) of short audio\n * frames is compared with a noise floor learned from the room. Speech onset\n * starts an utterance and a pause ends it. No Node APIs, no DOM; safe in\n * browsers, workers and Node.\n *\n * Two pure pieces:\n * - `createVadSegmenter`: fed one frame energy at a time, answers with\n *   `start` / `end` / `split` events and a live level;\n * - `createUtteranceCollector`: fed raw mono PCM, keeps a pre-roll so the\n *   first syllable is not clipped and hands over each finished utterance.\n *\n * `createVadCapture` (`@happyvertical/speech/browser`) drives the collector\n * from a microphone.\n */\n\n/** Tuning for the detector. Every field is optional. */\nexport interface VadOptions {\n  /** Silence that ends an utterance, in ms. Default 800. */\n  silenceMs?: number;\n  /** Speech shorter than this is a click or a cough, not an utterance. Default 150. */\n  minSpeechMs?: number;\n  /** Audio kept from before the onset so the first syllable is not clipped. Default 300. */\n  preRollMs?: number;\n  /** An utterance this long is cut and the next one starts at once. Default 30000. */\n  maxUtteranceMs?: number;\n  /**\n   * 0 (needs loud, close speech) to 1 (picks up quiet speech, and more\n   * noise). Default 0.5.\n   */\n  sensitivity?: number;\n  /**\n   * How long the room is listened to before the floor is settled, in ms.\n   * Default 300. Speech is still detected during this window (with a\n   * conservative floor), and calibration only uses a low percentile of the\n   * energies, so speaking from the first frame does not poison the floor.\n   */\n  calibrateMs?: number;\n}\n\nexport const VAD_DEFAULTS = {\n  silenceMs: 800,\n  minSpeechMs: 150,\n  preRollMs: 300,\n  maxUtteranceMs: 30_000,\n  sensitivity: 0.5,\n  calibrateMs: 300,\n} as const satisfies Required<VadOptions>;\n\n/** Length of one analysed frame. */\nexport const VAD_FRAME_MS = 20;\n\n/** The quietest noise floor assumed (digital silence must not make everything \"speech\"). */\nconst MIN_FLOOR = 0.0015;\n/** Floor assumed before the room has been heard (a conservative, quiet-room value). */\nconst DEFAULT_FLOOR = 0.003;\n/**\n * Highest floor calibration may settle on. If the opening frames are louder\n * than this they are taken to be speech (or a very noisy room) and the floor\n * stays conservative instead of climbing to speech level.\n */\nconst CALIBRATION_FLOOR_CAP = 0.006;\n/** Calibration uses this low percentile of the opening energies, not the mean. */\nconst CALIBRATION_PERCENTILE = 0.2;\n/** Highest floor slow tracking may reach between utterances. */\nconst MAX_FLOOR = 0.02;\n/** Time constant of the slow noise-floor tracking, in ms. */\nconst FLOOR_TRACK_MS = 2000;\n/** Silence kept at the end of an utterance when it is trimmed, in ms. */\nconst TRAILING_KEEP_MS = 250;\n\nexport type VadEndReason = 'silence' | 'max' | 'flush';\n\nexport type VadEvent =\n  /** Speech began; `leadMs` is how much of the recent past belongs to it. */\n  | { type: 'start'; leadMs: number }\n  /** The utterance ended; `trailingSilenceMs` of it is silence. */\n  | { type: 'end'; reason: VadEndReason; trailingSilenceMs: number }\n  /** `maxUtteranceMs` reached while speaking: end this one, start the next. */\n  | { type: 'split' };\n\nexport interface VadStep {\n  events: VadEvent[];\n  /** 0 to 1, how loud this frame is against what counts as speech. */\n  level: number;\n  /** Currently inside an utterance. */\n  speaking: boolean;\n}\n\nexport interface VadSegmenter {\n  /** Feed the RMS energy (0 to 1) of the next frame of `frameMs`. */\n  push(energy: number, frameMs?: number): VadStep;\n  /** Finish: ends an utterance in progress (`end` with reason `flush`). */\n  flush(): VadEvent[];\n  readonly speaking: boolean;\n  /** The learned noise floor (RMS). */\n  readonly noiseFloor: number;\n  /** The energy above which a frame counts as speech. */\n  readonly threshold: number;\n}\n\nfunction clamp(value: number, min: number, max: number): number {\n  return Math.min(max, Math.max(min, value));\n}\n\n/** Resolve options against the defaults, clamping silly values. */\nexport function resolveVadOptions(\n  options: VadOptions = {},\n): Required<VadOptions> {\n  const pick = (value: number | undefined, fallback: number, min: number) =>\n    typeof value === 'number' && Number.isFinite(value)\n      ? Math.max(min, value)\n      : fallback;\n  return {\n    silenceMs: pick(options.silenceMs, VAD_DEFAULTS.silenceMs, VAD_FRAME_MS),\n    minSpeechMs: pick(options.minSpeechMs, VAD_DEFAULTS.minSpeechMs, 0),\n    preRollMs: pick(options.preRollMs, VAD_DEFAULTS.preRollMs, 0),\n    maxUtteranceMs: pick(\n      options.maxUtteranceMs,\n      VAD_DEFAULTS.maxUtteranceMs,\n      1000,\n    ),\n    sensitivity: clamp(\n      typeof options.sensitivity === 'number' &&\n        Number.isFinite(options.sensitivity)\n        ? options.sensitivity\n        : VAD_DEFAULTS.sensitivity,\n      0,\n      1,\n    ),\n    calibrateMs: pick(options.calibrateMs, VAD_DEFAULTS.calibrateMs, 0),\n  };\n}\n\n/** Root-mean-square loudness of a block of samples (-1 to 1). */\nexport function frameEnergy(samples: ArrayLike<number>): number {\n  const n = samples.length;\n  if (n === 0) return 0;\n  let sum = 0;\n  for (let i = 0; i < n; i++) {\n    const s = samples[i] as number;\n    sum += s * s;\n  }\n  return Math.sqrt(sum / n);\n}\n\nexport function createVadSegmenter(options: VadOptions = {}): VadSegmenter {\n  const o = resolveVadOptions(options);\n  // Higher sensitivity: a lower multiple of the floor and a lower absolute minimum.\n  const ratio = 6 - 4 * o.sensitivity;\n  const absMin = 0.02 - 0.015 * o.sensitivity;\n\n  let floor = DEFAULT_FLOOR;\n  let calibratedMs = 0;\n  const calibrationEnergies: number[] = [];\n  let speaking = false;\n  let candidateMs = 0;\n  let silenceMs = 0;\n  let utteranceMs = 0;\n\n  const onThreshold = () => Math.max(floor * ratio, absMin);\n  // Hysteresis: once speaking, only clearly lower energy counts as a pause.\n  const offThreshold = () => Math.max(onThreshold() * 0.6, floor * 1.5);\n\n  const reset = () => {\n    speaking = false;\n    candidateMs = 0;\n    silenceMs = 0;\n    utteranceMs = 0;\n  };\n\n  return {\n    get speaking() {\n      return speaking;\n    },\n    get noiseFloor() {\n      return floor;\n    },\n    get threshold() {\n      return onThreshold();\n    },\n\n    push(energy, frameMs = VAD_FRAME_MS) {\n      const events: VadEvent[] = [];\n      const e = Number.isFinite(energy) ? Math.max(0, energy) : 0;\n      const level = clamp(e / (onThreshold() * 3), 0, 1);\n\n      // Learn the room from the quietest frames of the opening window. Frames\n      // inside an utterance never count, and the estimate is capped so speech\n      // at t=0 cannot raise the floor to speech level. Detection stays live.\n      const calibrating = calibratedMs < o.calibrateMs;\n      if (calibrating) {\n        calibratedMs += frameMs;\n        if (!speaking) {\n          calibrationEnergies.push(e);\n          const sorted = [...calibrationEnergies].sort((a, b) => a - b);\n          const p =\n            sorted[\n              Math.min(\n                sorted.length - 1,\n                Math.floor(sorted.length * CALIBRATION_PERCENTILE),\n              )\n            ] ?? DEFAULT_FLOOR;\n          floor = clamp(p, MIN_FLOOR, CALIBRATION_FLOOR_CAP);\n        }\n      }\n\n      if (!speaking) {\n        if (e > onThreshold()) {\n          candidateMs += frameMs;\n          if (candidateMs >= o.minSpeechMs) {\n            speaking = true;\n            silenceMs = 0;\n            utteranceMs = candidateMs;\n            events.push({\n              type: 'start',\n              leadMs: o.preRollMs + candidateMs,\n            });\n            candidateMs = 0;\n          }\n        } else if (!calibrating) {\n          candidateMs = 0;\n          // Track the room slowly: down faster than up, so a hum that fades\n          // is forgotten sooner than a noise that creeps in is accepted.\n          const alpha = (frameMs / FLOOR_TRACK_MS) * (e < floor ? 4 : 1);\n          floor = clamp(\n            floor + (e - floor) * Math.min(1, alpha),\n            MIN_FLOOR,\n            MAX_FLOOR,\n          );\n        } else {\n          candidateMs = 0;\n        }\n        return { events, level, speaking };\n      }\n\n      utteranceMs += frameMs;\n      if (e > offThreshold()) {\n        silenceMs = 0;\n      } else {\n        silenceMs += frameMs;\n      }\n      if (silenceMs >= o.silenceMs) {\n        events.push({\n          type: 'end',\n          reason: 'silence',\n          trailingSilenceMs: silenceMs,\n        });\n        reset();\n      } else if (utteranceMs >= o.maxUtteranceMs) {\n        events.push({ type: 'split' });\n        utteranceMs = 0;\n        silenceMs = 0;\n      }\n      return { events, level, speaking };\n    },\n\n    flush() {\n      const events: VadEvent[] = [];\n      if (speaking) {\n        events.push({\n          type: 'end',\n          reason: 'flush',\n          trailingSilenceMs: silenceMs,\n        });\n      }\n      reset();\n      return events;\n    },\n  };\n}\n\n/** One finished utterance. */\nexport interface VadUtterance {\n  /** Mono samples (-1 to 1) at `sampleRate`. */\n  pcm: Float32Array;\n  sampleRate: number;\n  durationMs: number;\n  reason: VadEndReason;\n}\n\nexport interface UtteranceCollectorOptions extends VadOptions {\n  /** Sample rate of the PCM fed in. */\n  sampleRate: number;\n  onUtterance: (utterance: VadUtterance) => void;\n  /** An utterance began (`true`) or ended (`false`). */\n  onSpeaking?: (speaking: boolean) => void;\n  /** Loudness 0 to 1 for each frame, for a meter. */\n  onLevel?: (level: number) => void;\n}\n\nexport interface UtteranceCollector {\n  /** Feed any amount of mono PCM. */\n  push(samples: Float32Array): void;\n  /** Hand over an utterance in progress and reset. */\n  flush(): void;\n  /**\n   * Drop an utterance in progress and all buffered audio (pre-roll included)\n   * without emitting it. The learned noise floor is kept.\n   */\n  discard(): void;\n  readonly speaking: boolean;\n  readonly segmenter: VadSegmenter;\n}\n\nexport function createUtteranceCollector(\n  options: UtteranceCollectorOptions,\n): UtteranceCollector {\n  const o = resolveVadOptions(options);\n  const segmenter = createVadSegmenter(o);\n  const frameSize = Math.max(\n    1,\n    Math.round((options.sampleRate * VAD_FRAME_MS) / 1000),\n  );\n  const frameMs = (frameSize / options.sampleRate) * 1000;\n  // Past audio kept for the pre-roll, plus the frames that proved it was speech.\n  const ringFrames = Math.ceil((o.preRollMs + o.minSpeechMs) / frameMs) + 1;\n  const ring: Float32Array[] = [];\n  let carry = new Float32Array(0);\n  let utterance: Float32Array[] | null = null;\n  let wasSpeaking = false;\n\n  const emit = (\n    frames: Float32Array[],\n    reason: VadEndReason,\n    trailMs: number,\n  ) => {\n    // Drop silence beyond a short tail; the model gains nothing from it.\n    const dropFrames = Math.max(\n      0,\n      Math.floor((trailMs - TRAILING_KEEP_MS) / frameMs),\n    );\n    const kept = frames.slice(0, Math.max(1, frames.length - dropFrames));\n    let total = 0;\n    for (const f of kept) total += f.length;\n    const pcm = new Float32Array(total);\n    let at = 0;\n    for (const f of kept) {\n      pcm.set(f, at);\n      at += f.length;\n    }\n    options.onUtterance({\n      pcm,\n      sampleRate: options.sampleRate,\n      durationMs: (total / options.sampleRate) * 1000,\n      reason,\n    });\n  };\n\n  const setSpeaking = (speaking: boolean) => {\n    if (speaking === wasSpeaking) return;\n    wasSpeaking = speaking;\n    options.onSpeaking?.(speaking);\n  };\n\n  const handleFrame = (frame: Float32Array) => {\n    const step = segmenter.push(frameEnergy(frame), frameMs);\n    options.onLevel?.(step.level);\n    ring.push(frame);\n    if (ring.length > ringFrames) ring.shift();\n    if (utterance) utterance.push(frame);\n    for (const event of step.events) {\n      if (event.type === 'start') {\n        // The ring already holds this frame; the utterance starts with it, and\n        // it is buffered from the very first frame of capture, so an utterance\n        // that begins during calibration keeps its first syllables.\n        utterance = [...ring];\n        // Near capture start there is less past than the pre-roll asks for.\n        // Transcribers do better with a little lead-in, so pad with silence.\n        const wantLead = Math.ceil(o.preRollMs / frameMs);\n        const haveLead = Math.max(\n          0,\n          ring.length - Math.ceil(o.minSpeechMs / frameMs),\n        );\n        for (let i = haveLead; i < wantLead; i++) {\n          utterance.unshift(new Float32Array(frameSize));\n        }\n      } else if (event.type === 'end') {\n        const frames = utterance ?? [];\n        utterance = null;\n        ring.length = 0;\n        if (frames.length > 0)\n          emit(frames, event.reason, event.trailingSilenceMs);\n      } else if (event.type === 'split') {\n        const frames = utterance ?? [];\n        utterance = [];\n        if (frames.length > 0) emit(frames, 'max', 0);\n      }\n    }\n    setSpeaking(segmenter.speaking);\n  };\n\n  return {\n    get speaking() {\n      return segmenter.speaking;\n    },\n    segmenter,\n    push(samples) {\n      let data = samples;\n      if (carry.length > 0) {\n        data = new Float32Array(carry.length + samples.length);\n        data.set(carry, 0);\n        data.set(samples, carry.length);\n      }\n      let at = 0;\n      while (at + frameSize <= data.length) {\n        // Copy: the caller reuses its buffer.\n        handleFrame(data.slice(at, at + frameSize));\n        at += frameSize;\n      }\n      carry = data.slice(at);\n    },\n    flush() {\n      const events = segmenter.flush();\n      for (const event of events) {\n        if (event.type === 'end' && utterance && utterance.length > 0) {\n          emit(utterance, 'flush', event.trailingSilenceMs);\n        }\n      }\n      utterance = null;\n      ring.length = 0;\n      carry = new Float32Array(0);\n      setSpeaking(false);\n      options.onLevel?.(0);\n    },\n    discard() {\n      segmenter.flush();\n      utterance = null;\n      ring.length = 0;\n      carry = new Float32Array(0);\n      setSpeaking(false);\n      options.onLevel?.(0);\n    },\n  };\n}\n","/**\n * Pure, dependency-free 16-bit PCM WAV helpers. No Node APIs and no DOM: they\n * run in the browser, in Node and in workers. Exposed as\n * `@happyvertical/speech/pcm`.\n *\n * `parseWavPcm16` is strict because it reads untrusted bytes: every declared\n * size is checked against the buffer before it is trusted, nothing is\n * allocated from a declared (rather than present) size, and work is linear in\n * the input length.\n */\n\nimport { WavFormatError } from './errors.js';\n\nconst HEADER_BYTES = 44;\nconst MAX_UINT32 = 0xffff_ffff;\nconst FORMAT_PCM = 1;\nconst FORMAT_EXTENSIBLE = 0xfffe;\n/** `KSDATAFORMAT_SUBTYPE_PCM` minus its leading format tag: 00000000-0010-8000-00AA-00389B71. */\nconst PCM_GUID_TAIL = [\n  0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b,\n  0x71,\n];\n\n/**\n * Converts float samples (-1..1) to 16-bit PCM (scale 32768, the inverse of\n * `pcm16ToFloat32`, so Int16 round-trips exactly). Clamps; NaN becomes 0.\n */\nexport function float32ToPcm16(samples: Float32Array): Int16Array {\n  const out = new Int16Array(samples.length);\n  for (let index = 0; index < samples.length; index += 1) {\n    const value = samples[index];\n    if (Number.isNaN(value)) continue; // stays 0\n    const clamped = value > 1 ? 1 : value < -1 ? -1 : value;\n    out[index] = Math.min(0x7fff, Math.round(clamped * 0x8000));\n  }\n  return out;\n}\n\n/** Converts 16-bit PCM to float samples in -1..1 (divides by 32768). */\nexport function pcm16ToFloat32(samples: Int16Array): Float32Array {\n  const out = new Float32Array(samples.length);\n  for (let index = 0; index < samples.length; index += 1) {\n    out[index] = samples[index] / 0x8000;\n  }\n  return out;\n}\n\nfunction assertSampleRate(sampleRate: number, what: string): void {\n  // byteRate = sampleRate * 2 must fit an unsigned 32-bit field.\n  if (\n    !Number.isInteger(sampleRate) ||\n    sampleRate < 1 ||\n    sampleRate * 2 > MAX_UINT32\n  ) {\n    throw new WavFormatError(\n      'invalid_argument',\n      `${what} must be a positive integer sample rate (got ${String(sampleRate)})`,\n    );\n  }\n}\n\nfunction writeTag(view: DataView, offset: number, tag: string): void {\n  for (let index = 0; index < tag.length; index += 1) {\n    view.setUint8(offset + index, tag.charCodeAt(index));\n  }\n}\n\n/**\n * Encodes mono samples as a canonical 44-byte-header 16-bit PCM WAV. Float\n * input is clamped and converted; `Int16Array` is written as is.\n */\nexport function encodeWavPcm16(\n  samples: Float32Array | Int16Array,\n  sampleRate: number,\n): Uint8Array {\n  assertSampleRate(sampleRate, 'sampleRate');\n  const dataBytes = samples.length * 2;\n  if (dataBytes + HEADER_BYTES - 8 > MAX_UINT32) {\n    throw new WavFormatError(\n      'invalid_argument',\n      'too many samples for a single WAV (4 GiB limit)',\n    );\n  }\n  const pcm = samples instanceof Int16Array ? samples : float32ToPcm16(samples);\n  const bytes = new Uint8Array(HEADER_BYTES + dataBytes);\n  const view = new DataView(bytes.buffer);\n  writeTag(view, 0, 'RIFF');\n  view.setUint32(4, 36 + dataBytes, true);\n  writeTag(view, 8, 'WAVE');\n  writeTag(view, 12, 'fmt ');\n  view.setUint32(16, 16, true);\n  view.setUint16(20, FORMAT_PCM, true);\n  view.setUint16(22, 1, true);\n  view.setUint32(24, sampleRate, true);\n  view.setUint32(28, sampleRate * 2, true);\n  view.setUint16(32, 2, true);\n  view.setUint16(34, 16, true);\n  writeTag(view, 36, 'data');\n  view.setUint32(40, dataBytes, true);\n  for (let index = 0; index < pcm.length; index += 1) {\n    view.setInt16(HEADER_BYTES + index * 2, pcm[index], true);\n  }\n  return bytes;\n}\n\n/** Constraints `parseWavPcm16` enforces on the declared format. */\nexport interface WavExpectation {\n  /** Required sample rate in Hz. */\n  sampleRate?: number;\n  /** Required channel count. */\n  channels?: number;\n}\n\n/** A validated 16-bit PCM WAV. */\nexport interface ParsedWavPcm16 {\n  sampleRate: number;\n  channels: number;\n  /** Frames (one sample per channel). */\n  frames: number;\n  durationMs: number;\n  /** Interleaved samples, decoded to native endianness (a copy). */\n  samples: Int16Array;\n  /** The little-endian PCM bytes, a view into the input. */\n  data: Uint8Array;\n}\n\nfunction fail(\n  reason: ConstructorParameters<typeof WavFormatError>[0],\n  message: string,\n): never {\n  throw new WavFormatError(reason, message);\n}\n\nfunction tagAt(bytes: Uint8Array, offset: number): string {\n  return String.fromCharCode(\n    bytes[offset],\n    bytes[offset + 1],\n    bytes[offset + 2],\n    bytes[offset + 3],\n  );\n}\n\ninterface Fmt {\n  channels: number;\n  sampleRate: number;\n}\n\nfunction readFmt(\n  view: DataView,\n  bytes: Uint8Array,\n  body: number,\n  size: number,\n): Fmt {\n  if (size < 16) {\n    fail('malformed_chunk', `fmt chunk is ${size} bytes; at least 16 required`);\n  }\n  let format = view.getUint16(body, true);\n  const channels = view.getUint16(body + 2, true);\n  const sampleRate = view.getUint32(body + 4, true);\n  const byteRate = view.getUint32(body + 8, true);\n  const blockAlign = view.getUint16(body + 12, true);\n  const bits = view.getUint16(body + 14, true);\n\n  if (format === FORMAT_EXTENSIBLE) {\n    if (size < 40) {\n      fail(\n        'malformed_chunk',\n        `WAVE_FORMAT_EXTENSIBLE fmt chunk is ${size} bytes; at least 40 required`,\n      );\n    }\n    const cbSize = view.getUint16(body + 16, true);\n    if (cbSize < 22 || 18 + cbSize > size) {\n      fail(\n        'malformed_chunk',\n        `WAVE_FORMAT_EXTENSIBLE extension (${cbSize} bytes) is under 22 or runs past its fmt chunk`,\n      );\n    }\n    const validBits = view.getUint16(body + 18, true);\n    if (validBits !== 0 && validBits !== 16) {\n      fail(\n        'unsupported_bits',\n        `WAVE_FORMAT_EXTENSIBLE valid bits is ${validBits}; only 16 supported`,\n      );\n    }\n    const guid = body + 24;\n    const subFormat = view.getUint16(guid, true);\n    const tailMatches = PCM_GUID_TAIL.every(\n      (byte, index) => bytes[guid + 2 + index] === byte,\n    );\n    if (subFormat !== FORMAT_PCM || !tailMatches) {\n      fail('unsupported_format', 'WAVE_FORMAT_EXTENSIBLE subtype is not PCM');\n    }\n    format = FORMAT_PCM;\n  }\n\n  if (format !== FORMAT_PCM) {\n    fail('unsupported_format', `WAV format tag ${format} is not PCM`);\n  }\n  if (bits !== 16) {\n    fail(\n      'unsupported_bits',\n      `WAV is ${bits}-bit; only 16-bit PCM is supported`,\n    );\n  }\n  if (channels < 1) {\n    fail('inconsistent_header', 'WAV declares zero channels');\n  }\n  if (sampleRate < 1) {\n    fail('inconsistent_header', 'WAV declares a zero sample rate');\n  }\n  if (blockAlign !== channels * 2) {\n    fail(\n      'inconsistent_header',\n      `WAV block align ${blockAlign} is not channels x 2 (${channels * 2})`,\n    );\n  }\n  if (byteRate !== sampleRate * blockAlign) {\n    fail(\n      'inconsistent_header',\n      'WAV byte rate is not sample rate x block align',\n    );\n  }\n  return { channels, sampleRate };\n}\n\n/**\n * Strictly parses a 16-bit PCM WAV. Rejects (with `WavFormatError`) anything\n * that is not a well-formed RIFF/WAVE with a `fmt ` chunk before its `data`\n * chunk, format tag 1 (or WAVE_FORMAT_EXTENSIBLE with the PCM subtype), 16\n * bits, a consistent header, and a declared data length that lies within the\n * buffer, is non-zero and is a whole number of frames. Nothing is clamped or\n * guessed. Every chunk inside the declared RIFF is bounds-checked, including\n * those after `data`; bytes beyond the RIFF are ignored.\n */\nexport function parseWavPcm16(\n  bytes: Uint8Array,\n  expect: WavExpectation = {},\n): ParsedWavPcm16 {\n  if (\n    bytes.length < 12 ||\n    tagAt(bytes, 0) !== 'RIFF' ||\n    tagAt(bytes, 8) !== 'WAVE'\n  ) {\n    fail('not_riff', 'not a RIFF/WAVE file');\n  }\n  const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);\n  const riffEnd = 8 + view.getUint32(4, true);\n  if (riffEnd < 12) {\n    fail('malformed_chunk', 'RIFF size is too small');\n  }\n  if (riffEnd > bytes.length) {\n    fail(\n      'truncated',\n      `RIFF declares ${riffEnd} bytes but only ${bytes.length} are present`,\n    );\n  }\n\n  let fmt: Fmt | undefined;\n  let dataAt: number | undefined;\n  let dataSize = 0;\n  let offset = 12;\n  while (offset < riffEnd) {\n    if (offset + 8 > riffEnd) {\n      fail('malformed_chunk', 'chunk header runs past the end of the RIFF');\n    }\n    const id = tagAt(bytes, offset);\n    const size = view.getUint32(offset + 4, true);\n    const body = offset + 8;\n    if (body + size > riffEnd) {\n      fail(\n        'truncated',\n        `${JSON.stringify(id)} chunk declares ${size} bytes but only ${riffEnd - body} remain`,\n      );\n    }\n\n    if (id === 'fmt ') {\n      if (fmt) fail('duplicate_fmt', 'WAV has more than one fmt chunk');\n      fmt = readFmt(view, bytes, body, size);\n      if (\n        expect.sampleRate !== undefined &&\n        fmt.sampleRate !== expect.sampleRate\n      ) {\n        fail(\n          'rate_mismatch',\n          `WAV is ${fmt.sampleRate} Hz; expected ${expect.sampleRate} Hz`,\n        );\n      }\n      if (expect.channels !== undefined && fmt.channels !== expect.channels) {\n        fail(\n          'channel_mismatch',\n          `WAV has ${fmt.channels} channel(s); expected ${expect.channels}`,\n        );\n      }\n    } else if (id === 'data') {\n      if (dataAt !== undefined) {\n        fail('malformed_chunk', 'WAV has more than one data chunk');\n      }\n      if (!fmt)\n        fail('data_before_fmt', 'WAV data chunk precedes its fmt chunk');\n      if (size === 0) fail('empty_data', 'WAV data chunk is empty');\n      const frameBytes = fmt.channels * 2;\n      if (size % frameBytes !== 0) {\n        fail(\n          'misaligned_data',\n          `WAV data (${size} bytes) is not a whole number of ${frameBytes}-byte frames`,\n        );\n      }\n      dataAt = body;\n      dataSize = size;\n    }\n    // RIFF chunks are word aligned: an odd size is followed by one pad byte,\n    // which must also lie inside the RIFF.\n    const next = body + size + (size % 2);\n    if (next > riffEnd) {\n      fail('truncated', `${JSON.stringify(id)} chunk is missing its pad byte`);\n    }\n    offset = next;\n  }\n  if (!fmt || dataAt === undefined) {\n    return fail(\n      fmt ? 'missing_data' : 'missing_fmt',\n      fmt ? 'WAV has no data chunk' : 'WAV has no fmt chunk',\n    );\n  }\n  // Every chunk inside the RIFF has now been bounds-checked; decode.\n  const data = bytes.subarray(dataAt, dataAt + dataSize);\n  const samples = new Int16Array(dataSize / 2);\n  for (let index = 0; index < samples.length; index += 1) {\n    samples[index] = view.getInt16(dataAt + index * 2, true);\n  }\n  const frames = dataSize / (fmt.channels * 2);\n  return {\n    sampleRate: fmt.sampleRate,\n    channels: fmt.channels,\n    frames,\n    durationMs: Math.round((frames / fmt.sampleRate) * 1000),\n    samples,\n    data,\n  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