@onjmin/koe 1.0.3 → 1.0.4

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Files changed (6) hide show
  1. package/dist/index.d.ts +115 -109
  2. package/dist/index.js +482 -393
  3. package/dist/index.js.map +1 -1
  4. package/dist/koe-convert.js +182 -154
  5. package/dist/koe-worklet.js +1 -1
  6. package/package.json +1 -1
package/dist/index.js.map CHANGED
@@ -1 +1 @@
1
- {"version":3,"sources":["../src/koe.ts","../src/engine/voice-bank.ts","../src/engine/index.ts","../src/engine/worldline.ts","../src/converter/parse-oto.ts","../src/converter/wav.ts","../src/converter/pitch.ts","../src/converter/pack.ts","../src/converter/frq.ts"],"sourcesContent":["import type { Manifest } from \"./types.js\";\n\n/** Magic number 'KOE\\0' (stored big-endian as 0x4B 0x4F 0x45 0x00). */\nconst MAGIC = 0x4b4f4500;\n\n/**\n * Koe Archive Format (.koe)\n * [4B] magic 'KOE\\0' (big-endian)\n * [4B] JSON length (little-endian)\n * [N ] manifest JSON (UTF-8)\n * [M ] raw PCM (Int16 / 48kHz / mono); phoneme offsets are relative to here\n */\nexport function packKoe(manifest: Manifest, pcmParts: BlobPart[]): Blob {\n\tconst json = new TextEncoder().encode(JSON.stringify(manifest));\n\tconst header = new ArrayBuffer(8);\n\tconst view = new DataView(header);\n\tview.setUint32(0, MAGIC, false); // magic, big-endian\n\tview.setUint32(4, json.byteLength, true); // json length, little-endian\n\treturn new Blob([header, json, ...pcmParts]);\n}\n\n/** Read the 8-byte header → JSON length. Throws on bad magic. */\nexport function parseKoeHeader(headerBytes: ArrayBuffer): {\n\tjsonLength: number;\n} {\n\tconst view = new DataView(headerBytes);\n\tif (view.byteLength < 8 || view.getUint32(0, false) !== MAGIC) {\n\t\tthrow new Error(\"Not a .koe file (bad magic)\");\n\t}\n\treturn { jsonLength: view.getUint32(4, true) };\n}\n\n/** Byte offset where PCM data begins, given the JSON length. */\nexport const pcmBase = (jsonLength: number): number => 8 + jsonLength;\n","import type { Manifest } from \"../types.js\";\nimport { parseKoeHeader, pcmBase } from \"../koe.js\";\n\n/**\n * Supplies raw bytes from the PCM section of a .koe archive on demand.\n * - BlobVoiceSource: slices an in-memory Blob / File (no full-buffer copy)\n * - RangeVoiceSource: HTTP Range requests against a URL (mobile-friendly)\n * `base` is the byte offset where PCM data starts (8 + jsonLength).\n */\ninterface VoiceSource {\n\treadBytes(offset: number, length: number): Promise<ArrayBuffer>;\n}\n\nclass BlobVoiceSource implements VoiceSource {\n\tconstructor(\n\t\tprivate blob: Blob,\n\t\tprivate base: number,\n\t) {}\n\treadBytes(offset: number, length: number): Promise<ArrayBuffer> {\n\t\tconst start = this.base + offset;\n\t\treturn this.blob.slice(start, start + length).arrayBuffer();\n\t}\n}\n\nclass RangeVoiceSource implements VoiceSource {\n\tconstructor(\n\t\tprivate url: string,\n\t\tprivate base: number,\n\t) {}\n\tasync readBytes(offset: number, length: number): Promise<ArrayBuffer> {\n\t\tconst start = this.base + offset;\n\t\tconst res = await fetch(this.url, {\n\t\t\theaders: { Range: `bytes=${start}-${start + length - 1}` },\n\t\t});\n\t\tif (!res.ok && res.status !== 206) {\n\t\t\tthrow new Error(`.koe range request failed: ${res.status}`);\n\t\t}\n\t\treturn res.arrayBuffer();\n\t}\n}\n\nasync function rangeFetch(\n\turl: string,\n\tstart: number,\n\tlength: number,\n): Promise<ArrayBuffer> {\n\tconst res = await fetch(url, {\n\t\theaders: { Range: `bytes=${start}-${start + length - 1}` },\n\t});\n\tif (!res.ok && res.status !== 206)\n\t\tthrow new Error(`.koe fetch failed: ${res.status}`);\n\treturn res.arrayBuffer();\n}\n\n/**\n * Read-only access to a .koe voice bank: its manifest plus per-phoneme PCM,\n * fetched on demand (Blob slice or HTTP Range). The full bank is never held in\n * memory.\n *\n * Pure data — no AudioContext, no AudioWorklet, no DOM. Use this when you only\n * need the source samples (e.g. to feed the {@link Worldline} renderer or any\n * other vocoder). {@link KoeEngine} builds its concatenative playback on top of\n * this same class.\n *\n * const bank = await VoiceBank.load(koeBlobOrUrl);\n * const pcm = await bank.getPcm('a'); // Float64 [-1, 1]\n */\nexport class VoiceBank {\n\tprivate constructor(\n\t\t/** The voice bank manifest (sample rate, reference pitch, phoneme table). */\n\t\treadonly manifest: Manifest,\n\t\tprivate source: VoiceSource,\n\t) {}\n\n\t/**\n\t * Parse a .koe archive header + manifest and bind a lazy PCM source.\n\t * @param koe a Blob/File of the .koe archive, or a URL (served with Range support)\n\t */\n\tstatic async load(koe: Blob | string): Promise<VoiceBank> {\n\t\tif (typeof koe === \"string\") {\n\t\t\tconst header = await rangeFetch(koe, 0, 8);\n\t\t\tconst { jsonLength } = parseKoeHeader(header);\n\t\t\tconst json = await rangeFetch(koe, 8, jsonLength);\n\t\t\tconst manifest = JSON.parse(new TextDecoder().decode(json)) as Manifest;\n\t\t\treturn new VoiceBank(\n\t\t\t\tmanifest,\n\t\t\t\tnew RangeVoiceSource(koe, pcmBase(jsonLength)),\n\t\t\t);\n\t\t}\n\t\tconst header = await koe.slice(0, 8).arrayBuffer();\n\t\tconst { jsonLength } = parseKoeHeader(header);\n\t\tconst json = await koe.slice(8, 8 + jsonLength).arrayBuffer();\n\t\tconst manifest = JSON.parse(new TextDecoder().decode(json)) as Manifest;\n\t\treturn new VoiceBank(\n\t\t\tmanifest,\n\t\t\tnew BlobVoiceSource(koe, pcmBase(jsonLength)),\n\t\t);\n\t}\n\n\t/** True if the bank contains a phoneme under this alias. */\n\thas(phoneme: string): boolean {\n\t\treturn this.manifest.phonemes[phoneme] !== undefined;\n\t}\n\n\t/**\n\t * Raw Int16 PCM bytes (48 kHz / mono) for a phoneme, or null if unknown.\n\t * The returned ArrayBuffer is freshly allocated and safe to transfer to a\n\t * worker / AudioWorklet.\n\t */\n\tasync readPcmBytes(phoneme: string): Promise<ArrayBuffer | null> {\n\t\tconst entry = this.manifest.phonemes[phoneme];\n\t\tif (!entry) return null;\n\t\treturn this.source.readBytes(entry.offset, entry.length * 2); // Int16 = 2 bytes/sample\n\t}\n\n\t/**\n\t * A phoneme's PCM as a Float64Array normalised to [-1, 1], or null if unknown.\n\t * Intended for external analysis / resynthesis such as the WORLD vocoder.\n\t */\n\tasync getPcm(phoneme: string): Promise<Float64Array | null> {\n\t\tconst buf = await this.readPcmBytes(phoneme);\n\t\tif (!buf) return null;\n\t\tconst int16 = new Int16Array(buf);\n\t\tconst f64 = new Float64Array(int16.length);\n\t\tfor (let i = 0; i < int16.length; i++) f64[i] = int16[i] / 32768;\n\t\treturn f64;\n\t}\n}\n","import type { Manifest, NoteEvent } from \"../types.js\";\nimport { VoiceBank } from \"./voice-bank.js\";\n\nexport type { NoteEvent };\n\nexport interface KoeEngineOptions {\n\t/** URL to koe-worklet.js. Defaults to './koe-worklet.js'. */\n\tworkletUrl?: string;\n}\n\n/**\n * Main-thread API for the koe concatenative synthesis engine.\n *\n * Loads a single .koe archive. The full voice bank is never held in memory:\n * phonemes are fetched on demand (via {@link VoiceBank}) and cached in the\n * audio thread.\n *\n * const engine = new KoeEngine();\n * await engine.load(koeBlobOrUrl);\n * await engine.play([{ phoneme: 'a', pitch: 440, duration: 48000 }]);\n *\n * For analysis / resynthesis (e.g. {@link Worldline}) you usually only need the\n * raw samples — use {@link VoiceBank} directly instead, which needs no\n * AudioContext or worklet.\n */\nexport class KoeEngine {\n\tprivate ctx: AudioContext;\n\tprivate workletUrl: string;\n\tprivate node: AudioWorkletNode | null = null;\n\tprivate bank: VoiceBank | null = null;\n\tprivate delivered = new Set<string>();\n\tprivate pending = new Map<string, Promise<void>>();\n\n\tconstructor(options: KoeEngineOptions = {}) {\n\t\tthis.ctx = new AudioContext({ sampleRate: 48000 });\n\t\tthis.workletUrl = options.workletUrl ?? \"./koe-worklet.js\";\n\t}\n\n\tget audioContext(): AudioContext {\n\t\treturn this.ctx;\n\t}\n\n\tget manifest(): Manifest | null {\n\t\treturn this.bank?.manifest ?? null;\n\t}\n\n\t/** The underlying voice bank (manifest + on-demand PCM), or null before load(). */\n\tget voiceBank(): VoiceBank | null {\n\t\treturn this.bank;\n\t}\n\n\t/**\n\t * Register the worklet and bind a .koe voice bank.\n\t * @param koe a Blob/File of the .koe archive, or a URL (served with Range support)\n\t */\n\tasync load(koe: Blob | string): Promise<void> {\n\t\tawait this.ctx.audioWorklet.addModule(this.workletUrl);\n\n\t\tthis.bank = await VoiceBank.load(koe);\n\t\tthis.delivered.clear();\n\t\tthis.pending.clear();\n\n\t\tthis.node = new AudioWorkletNode(this.ctx, \"koe-processor\", {\n\t\t\tnumberOfInputs: 0,\n\t\t\tnumberOfOutputs: 1,\n\t\t\toutputChannelCount: [1],\n\t\t});\n\t\tthis.node.port.postMessage({ type: \"init\", manifest: this.bank.manifest });\n\t\tthis.node.connect(this.ctx.destination);\n\n\t\tconsole.log(\n\t\t\t\"[koe] ready —\",\n\t\t\tObject.keys(this.bank.manifest.phonemes).length,\n\t\t\t\"phonemes (on-demand)\",\n\t\t);\n\t}\n\n\t/** Fetch one phoneme's PCM and deliver it to the worklet (deduped, cached). */\n\tprivate ensurePhoneme(name: string): Promise<void> {\n\t\tif (this.delivered.has(name)) return Promise.resolve();\n\t\tconst existing = this.pending.get(name);\n\t\tif (existing) return existing;\n\t\tif (!this.bank || !this.node) return Promise.resolve();\n\n\t\tconst load = this.bank.readPcmBytes(name).then((buf) => {\n\t\t\tif (!buf) return;\n\t\t\tthis.node!.port.postMessage({ type: \"phoneme\", name, buffer: buf }, [\n\t\t\t\tbuf,\n\t\t\t]);\n\t\t\tthis.delivered.add(name);\n\t\t\tthis.pending.delete(name);\n\t\t});\n\n\t\tthis.pending.set(name, load);\n\t\treturn load;\n\t}\n\n\t/** Stop current playback, preload the phonemes for `notes`, then queue them. */\n\tasync play(notes: NoteEvent[]): Promise<void> {\n\t\tif (!this.node) throw new Error(\"KoeEngine: call load() before play()\");\n\t\tthis.node.port.postMessage({ type: \"stop\" });\n\t\tconst names = [...new Set(notes.map((n) => n.phoneme))].filter(Boolean);\n\t\tawait Promise.all(names.map((n) => this.ensurePhoneme(n)));\n\t\tthis.node.port.postMessage({ type: \"play\", notes });\n\t}\n\n\t/** Stop playback and clear the queue. */\n\tstop(): void {\n\t\tthis.node?.port.postMessage({ type: \"stop\" });\n\t}\n\n\t/** Resume the AudioContext if suspended (e.g. after autoplay block). */\n\tasync resume(): Promise<void> {\n\t\tif (this.ctx.state === \"suspended\") await this.ctx.resume();\n\t}\n\n\t/**\n\t * Read a phoneme's raw PCM and return it as a Float64Array normalised to\n\t * [-1, 1]. Convenience that forwards to the underlying {@link VoiceBank}.\n\t * Intended for external analysis such as the WORLD vocoder.\n\t */\n\tasync getPcm(phoneme: string): Promise<Float64Array | null> {\n\t\treturn this.bank?.getPcm(phoneme) ?? null;\n\t}\n}\n","import type { PhonemeEntry } from \"../types.js\";\n\n/** Output sample rate of the worldline synthesizer. */\nexport const WORLDLINE_SAMPLE_RATE = 48000;\n\n/**\n * WORLD needs roughly 85 ms of audio for stable F0 analysis. Phonemes with\n * fewer samples than this are rejected (renderNote returns null).\n */\nexport const MIN_WORLDLINE_SAMPLES = 4096;\n\n/** SynthRequest struct layout — 120 bytes, WASM32. */\nconst SYNTH_REQ_SIZE = 120;\n/** Frame period hardcoded in worldline's phrase_synth.cpp. */\nconst WL_FRAME_MS = 10;\n\n/**\n * The subset of the Emscripten module surface that we call. worldline.js is an\n * MODULARIZE=1 / EXPORT_NAME=WorldlineModule build of OpenUtau's worldline.\n */\ninterface WorldlineWasm {\n\t_PhraseSynthNew(): number;\n\t_PhraseSynthDelete(ps: number): void;\n\t_PhraseSynthAddRequest(\n\t\tps: number,\n\t\treq: number,\n\t\tposMs: number,\n\t\tskipMs: number,\n\t\tlengthMs: number,\n\t\tfadeInMs: number,\n\t\tfadeOutMs: number,\n\t\tflag: number,\n\t): void;\n\t_PhraseSynthSetCurves(\n\t\tps: number,\n\t\tf0: number,\n\t\tgender: number,\n\t\ttension: number,\n\t\tbreathiness: number,\n\t\tvoicing: number,\n\t\tlength: number,\n\t\tframeMs: number,\n\t): void;\n\t_PhraseSynthSynth(ps: number, yPtrPtr: number, flag: number): number;\n\t_malloc(size: number): number;\n\t_free(ptr: number): void;\n\tsetValue(ptr: number, value: number, type: string): void;\n\tgetValue(ptr: number, type: string): number;\n\tHEAPF32: Float32Array;\n\tHEAPF64: Float64Array;\n}\n\ntype WorldlineFactory = (opts?: {\n\tlocateFile?: (path: string) => string;\n}) => Promise<WorldlineWasm>;\n\ndeclare global {\n\t// Injected by worldline.js when loaded as a classic <script> or via importScripts.\n\t// eslint-disable-next-line no-var\n\tvar WorldlineModule: WorldlineFactory | undefined;\n\t// Present in classic Web Workers (used to load worldline.js off the main thread).\n\t// eslint-disable-next-line no-var\n\tvar importScripts: ((...urls: string[]) => void) | undefined;\n}\n\nexport interface WorldlineLoadOptions {\n\t/**\n\t * URL of `worldline.js` (the Emscripten loader). The matching\n\t * `worldline.wasm` must sit next to it — it is resolved relative to this URL.\n\t *\n\t * When consuming koe from npm + a CDN this is typically e.g.\n\t * `https://cdn.jsdelivr.net/npm/@onjmin/koe/dist/world/worldline.js`, or your\n\t * own hosted copy of `dist/world/`.\n\t */\n\tscriptUrl: string;\n}\n\nexport interface RenderNoteParams {\n\t/**\n\t * Source phoneme PCM normalised to [-1, 1] (e.g. from\n\t * `VoiceBank.getPcm()` / `KoeEngine.getPcm()`).\n\t */\n\tpcm: Float64Array;\n\t/** Target output pitch in Hz. */\n\tpitch: number;\n\t/** Sustain / vowel duration in ms (the lead-in below is rendered on top). */\n\tdurationMs: number;\n\t/** Preutterance / lead-in in ms — convert from {@link PhonemeEntry.pre}. */\n\tpreMs: number;\n\t/** Consonant length in ms — convert from {@link PhonemeEntry.consonant}. */\n\tconsonantMs: number;\n\t/** Reference tempo in BPM for worldline's internal timing. Default 120. */\n\ttempo?: number;\n}\n\n/** Convert a sample count at 48 kHz to milliseconds. */\nexport const samplesToMs = (samples: number): number =>\n\t(samples / WORLDLINE_SAMPLE_RATE) * 1000;\n\n/**\n * Derive {@link RenderNoteParams} lead-in / consonant fields from a manifest\n * entry, so callers don't repeat the sample→ms conversion.\n *\n * const params = { pcm, pitch, durationMs, ...leadInFromEntry(entry) };\n */\nexport function leadInFromEntry(entry: PhonemeEntry): {\n\tpreMs: number;\n\tconsonantMs: number;\n} {\n\treturn {\n\t\tpreMs: samplesToMs(entry.pre || 0),\n\t\tconsonantMs: samplesToMs(entry.consonant || 0),\n\t};\n}\n\n// Cache module factories by script URL so repeated loads share one WASM instance.\nconst moduleCache = new Map<string, Promise<WorldlineWasm>>();\n\nfunction injectScript(src: string): Promise<void> {\n\treturn new Promise((resolve, reject) => {\n\t\tconst existing = document.querySelector<HTMLScriptElement>(\n\t\t\t`script[data-koe-worldline=\"${src}\"]`,\n\t\t);\n\t\tif (existing) {\n\t\t\tresolve();\n\t\t\treturn;\n\t\t}\n\t\tconst s = document.createElement(\"script\");\n\t\ts.src = src;\n\t\ts.dataset.koeWorldline = src;\n\t\ts.onload = () => resolve();\n\t\ts.onerror = () => reject(new Error(`worldline: failed to load ${src}`));\n\t\tdocument.head.appendChild(s);\n\t});\n}\n\nfunction loadWasm(scriptUrl: string): Promise<WorldlineWasm> {\n\tconst cached = moduleCache.get(scriptUrl);\n\tif (cached) return cached;\n\n\tconst baseUrl = scriptUrl.slice(0, scriptUrl.lastIndexOf(\"/\") + 1);\n\tconst instantiate = (): Promise<WorldlineWasm> => {\n\t\tconst factory = globalThis.WorldlineModule;\n\t\tif (!factory)\n\t\t\tthrow new Error(\n\t\t\t\t\"worldline: WorldlineModule global was not defined by the script\",\n\t\t\t);\n\t\t// locateFile resolves worldline.wasm next to the script; fetch works in both\n\t\t// the DOM and Web Workers, so the WASM instantiates the same way off-thread.\n\t\treturn factory({ locateFile: (f: string) => baseUrl + f });\n\t};\n\n\tlet promise: Promise<WorldlineWasm>;\n\tif (typeof document !== \"undefined\") {\n\t\t// Main thread / DOM: load worldline.js via a <script> tag.\n\t\tpromise = injectScript(scriptUrl).then(instantiate);\n\t} else if (typeof globalThis.importScripts === \"function\") {\n\t\t// Classic Web Worker: importScripts runs worldline.js synchronously in the\n\t\t// worker's global scope, defining globalThis.WorldlineModule. This lets the\n\t\t// heavy WORLD synthesis run off the main thread.\n\t\tpromise = Promise.resolve().then(() => {\n\t\t\t(globalThis.importScripts as (...urls: string[]) => void)(scriptUrl);\n\t\t\treturn instantiate();\n\t\t});\n\t} else {\n\t\treturn Promise.reject(\n\t\t\tnew Error(\n\t\t\t\t\"Worldline.load requires a DOM or a classic Web Worker (importScripts) to load worldline.js\",\n\t\t\t),\n\t\t);\n\t}\n\tmoduleCache.set(scriptUrl, promise);\n\treturn promise;\n}\n\n/**\n * High-quality WORLD-vocoder note renderer (OpenUtau's worldline via WASM).\n *\n * Pure synthesis: PCM in → PCM out. No AudioContext, no scheduling — the caller\n * owns playback (schedule the returned buffer on its own timeline). Pair it with\n * {@link VoiceBank} for the source samples:\n *\n * const bank = await VoiceBank.load(koeUrl);\n * const wl = await Worldline.load({ scriptUrl: '.../world/worldline.js' });\n *\n * const entry = bank.manifest.phonemes[alias];\n * const pcm = await bank.getPcm(alias);\n * const audio = wl.renderNote({\n * pcm, pitch: 440, durationMs: 500, ...leadInFromEntry(entry),\n * });\n * // audio: Float32 @ 48 kHz, layout [lead-in/consonant ≈ preMs][vowel ≈ durationMs]\n */\nexport class Worldline {\n\treadonly sampleRate = WORLDLINE_SAMPLE_RATE;\n\n\tprivate constructor(private wasm: WorldlineWasm) {}\n\n\t/**\n\t * Load + instantiate the worldline WASM module (deduped per scriptUrl).\n\t *\n\t * Works on the main thread (loads via `<script>`) and inside a classic Web\n\t * Worker (loads via `importScripts`), so the heavy synthesis can run\n\t * off-thread. The matching `worldline.wasm` is fetched next to scriptUrl.\n\t */\n\tstatic async load(options: WorldlineLoadOptions): Promise<Worldline> {\n\t\treturn new Worldline(await loadWasm(options.scriptUrl));\n\t}\n\n\t/**\n\t * Render one note to Float32 PCM at 48 kHz.\n\t *\n\t * The output buffer is laid out as [lead-in/consonant ≈ preMs][vowel ≈\n\t * durationMs], rendered from sample offset 0 (no leading silence). The vowel\n\t * onset (the \"beat\") sits at ≈ preMs into the buffer, so a sequencer should\n\t * place the buffer at `beatTime − preMs` and may trim/crossfade the lead-in.\n\t *\n\t * No internal crossfade is applied — apply fades externally.\n\t *\n\t * @returns Float32 PCM, or null when `pcm` is shorter than\n\t * {@link MIN_WORLDLINE_SAMPLES} (too short for stable F0 analysis).\n\t */\n\trenderNote(params: RenderNoteParams): Float32Array | null {\n\t\tconst { pcm, pitch, durationMs, preMs, consonantMs, tempo = 120 } = params;\n\t\tif (!pcm || pcm.length < MIN_WORLDLINE_SAMPLES) return null;\n\n\t\tconst WL = this.wasm;\n\t\tconst FS = WORLDLINE_SAMPLE_RATE;\n\t\tconst midiNote = Math.round(69 + 12 * Math.log2(pitch / 440));\n\t\tconst posMs = 0; // no leading silence\n\t\tconst reqLen = preMs + durationMs; // render lead-in/consonant + vowel\n\n\t\t// cut_off must trim at least ~2 frames off the tail. PhraseSynth::AddRequest\n\t\t// computes the trim frame count as ceil(in_length_ms)/frame_ms, but\n\t\t// PyinEstimator yields floor(N/nhop)−1 frames — one fewer. With cut_off=0 the\n\t\t// requested frame count exceeds f0_.size() and Model::Trim's f0_.erase() runs\n\t\t// past end(), causing \"memory access out of bounds\". Trimming 2 frames keeps\n\t\t// it within bounds.\n\t\tconst cutMs = WL_FRAME_MS * 2;\n\n\t\tconst ps = WL._PhraseSynthNew();\n\t\tif (!ps) return null;\n\t\tconst reqPtr = WL._malloc(SYNTH_REQ_SIZE);\n\t\tif (!reqPtr) {\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tconst samplePtr = WL._malloc(pcm.length * 8);\n\t\tif (!samplePtr) {\n\t\t\tWL._free(reqPtr);\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tWL.HEAPF64.set(pcm, samplePtr >> 3);\n\n\t\tconst sv = (off: number, val: number, type: string) =>\n\t\t\tWL.setValue(reqPtr + off, val, type);\n\t\tsv(0, FS, \"i32\");\n\t\tsv(4, pcm.length, \"i32\");\n\t\tsv(8, samplePtr, \"*\");\n\t\tsv(12, 0, \"i32\");\n\t\tsv(16, 0, \"*\");\n\t\tsv(20, midiNote, \"i32\");\n\t\tsv(24, 100.0, \"double\");\n\t\tsv(32, 0.0, \"double\");\n\t\tsv(40, reqLen, \"double\");\n\t\tsv(48, consonantMs, \"double\");\n\t\tsv(56, cutMs, \"double\");\n\t\tsv(64, 100.0, \"double\");\n\t\tsv(72, 0.0, \"double\");\n\t\tsv(80, tempo, \"double\");\n\t\tsv(88, 0, \"i32\");\n\t\tsv(92, 0, \"*\");\n\t\tsv(96, 0, \"i32\");\n\t\tsv(100, 0, \"i32\");\n\t\tsv(104, 100, \"i32\");\n\t\tsv(108, 0, \"i32\");\n\t\tsv(112, 0, \"i32\");\n\t\tsv(116, 100, \"i32\");\n\n\t\t// skip/fadeIn/fadeOut all 0 — crossfade is handled externally.\n\t\tWL._PhraseSynthAddRequest(ps, reqPtr, posMs, 0.0, reqLen, 0.0, 0.0, 0);\n\t\tWL._free(samplePtr);\n\t\tWL._free(reqPtr);\n\n\t\tconst totalMs = posMs + reqLen + WL_FRAME_MS * 2;\n\t\tconst nFrames = Math.ceil(totalMs / WL_FRAME_MS) + 4;\n\t\tconst f0Arr = new Float64Array(nFrames).fill(pitch);\n\t\tconst gArr = new Float64Array(nFrames).fill(0.5);\n\t\tconst tArr = new Float64Array(nFrames).fill(0.5);\n\t\tconst bArr = new Float64Array(nFrames).fill(0.5);\n\t\tconst vArr = new Float64Array(nFrames).fill(1.0);\n\n\t\tconst f0Ptr = WL._malloc(nFrames * 8);\n\t\tconst gPtr = WL._malloc(nFrames * 8);\n\t\tconst tPtr = WL._malloc(nFrames * 8);\n\t\tconst bPtr = WL._malloc(nFrames * 8);\n\t\tconst vPtr = WL._malloc(nFrames * 8);\n\t\tif (!f0Ptr || !gPtr || !tPtr || !bPtr || !vPtr) {\n\t\t\tif (f0Ptr) WL._free(f0Ptr);\n\t\t\tif (gPtr) WL._free(gPtr);\n\t\t\tif (tPtr) WL._free(tPtr);\n\t\t\tif (bPtr) WL._free(bPtr);\n\t\t\tif (vPtr) WL._free(vPtr);\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tWL.HEAPF64.set(f0Arr, f0Ptr >> 3);\n\t\tWL.HEAPF64.set(gArr, gPtr >> 3);\n\t\tWL.HEAPF64.set(tArr, tPtr >> 3);\n\t\tWL.HEAPF64.set(bArr, bPtr >> 3);\n\t\tWL.HEAPF64.set(vArr, vPtr >> 3);\n\t\tWL._PhraseSynthSetCurves(\n\t\t\tps,\n\t\t\tf0Ptr,\n\t\t\tgPtr,\n\t\t\ttPtr,\n\t\t\tbPtr,\n\t\t\tvPtr,\n\t\t\tnFrames,\n\t\t\tWL_FRAME_MS,\n\t\t);\n\t\tWL._free(f0Ptr);\n\t\tWL._free(gPtr);\n\t\tWL._free(tPtr);\n\t\tWL._free(bPtr);\n\t\tWL._free(vPtr);\n\n\t\tconst yPtrPtr = WL._malloc(4);\n\t\tif (!yPtrPtr) {\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tconst outLen = WL._PhraseSynthSynth(ps, yPtrPtr, 0);\n\t\tconst yPtr = WL.getValue(yPtrPtr, \"*\");\n\t\tconst audio =\n\t\t\toutLen > 0\n\t\t\t\t? new Float32Array(WL.HEAPF32.buffer, yPtr, outLen).slice()\n\t\t\t\t: null;\n\t\tWL._free(yPtrPtr);\n\t\tWL._PhraseSynthDelete(ps);\n\t\treturn audio;\n\t}\n}\n","export interface OtoEntry {\n\t/** Source WAV filename */\n\twav: string;\n\t/** Phoneme alias */\n\talias: string;\n\t/** Left blank — offset from WAV start (ms) */\n\toffset: number;\n\t/** Consonant portion end from offset (ms) */\n\tconsonant: number;\n\t/** Right blank — negative = from WAV end, positive = from offset (ms) */\n\tcutoff: number;\n\t/** Preutterance from offset (ms) */\n\tpre: number;\n\t/** Overlap / crossfade region (ms) */\n\toverlap: number;\n}\n\n/**\n * Parse oto.ini content (already decoded to UTF-8 string).\n * Silently skips malformed lines.\n */\nexport function parseOto(content: string): OtoEntry[] {\n\tconst entries: OtoEntry[] = [];\n\n\tfor (const raw of content.split(/\\r?\\n/)) {\n\t\tconst line = raw.trim();\n\t\tif (!line || line.startsWith(\"#\")) continue;\n\n\t\tconst eq = line.indexOf(\"=\");\n\t\tif (eq === -1) continue;\n\n\t\tconst wav = line.slice(0, eq).trim();\n\t\tconst parts = line.slice(eq + 1).split(\",\");\n\t\tif (parts.length < 6) continue;\n\n\t\tconst [alias, offsetStr, consonantStr, cutoffStr, preStr, overlapStr] =\n\t\t\tparts;\n\t\t// Empty alias → use filename without extension (UTAU spec default)\n\t\tconst aliasStr = alias.trim() || wav.replace(/\\.[^.]+$/, \"\");\n\t\tconst entry: OtoEntry = {\n\t\t\twav,\n\t\t\talias: aliasStr,\n\t\t\toffset: parseFloat(offsetStr) || 0,\n\t\t\tconsonant: parseFloat(consonantStr) || 0,\n\t\t\tcutoff: parseFloat(cutoffStr) || 0,\n\t\t\tpre: parseFloat(preStr) || 0,\n\t\t\toverlap: parseFloat(overlapStr) || 0,\n\t\t};\n\n\t\tif (!entry.alias) continue;\n\t\tentries.push(entry);\n\t}\n\n\treturn entries;\n}\n","export interface WavData {\n\tsampleRate: number;\n\tchannels: number;\n\t/** Normalized samples in [-1, 1], interleaved if multi-channel */\n\tsamples: Float32Array;\n}\n\n/** Parse a WAV file from an ArrayBuffer. Supports PCM 8/16/24-bit and IEEE float 32-bit. */\nexport function parseWav(buf: ArrayBuffer): WavData {\n\tconst view = new DataView(buf);\n\n\tconst riff = readFourCC(view, 0);\n\tif (riff !== \"RIFF\") throw new Error(`Not a RIFF file (got \"${riff}\")`);\n\n\tlet sampleRate = 0;\n\tlet channels = 0;\n\tlet bitsPerSample = 0;\n\tlet audioFormat = 1;\n\tlet dataOffset = 0;\n\tlet dataLength = 0;\n\n\tlet pos = 12;\n\twhile (pos < view.byteLength - 8) {\n\t\tconst id = readFourCC(view, pos);\n\t\tconst size = view.getUint32(pos + 4, true);\n\t\tpos += 8;\n\n\t\tif (id === \"fmt \") {\n\t\t\taudioFormat = view.getUint16(pos, true);\n\t\t\tchannels = view.getUint16(pos + 2, true);\n\t\t\tsampleRate = view.getUint32(pos + 4, true);\n\t\t\tbitsPerSample = view.getUint16(pos + 14, true);\n\t\t} else if (id === \"data\") {\n\t\t\tdataOffset = pos;\n\t\t\tdataLength = size;\n\t\t\tbreak;\n\t\t}\n\n\t\tpos += size + (size & 1);\n\t}\n\n\tif (!dataOffset) throw new Error(\"WAV has no data chunk\");\n\tif (!channels || !sampleRate) throw new Error(\"WAV fmt chunk missing\");\n\n\tconst bytesPerSample = bitsPerSample >> 3;\n\tconst totalSamples = Math.floor(dataLength / bytesPerSample);\n\tconst samples = new Float32Array(totalSamples);\n\n\tfor (let i = 0; i < totalSamples; i++) {\n\t\tconst p = dataOffset + i * bytesPerSample;\n\t\tif (audioFormat === 3) {\n\t\t\tsamples[i] = view.getFloat32(p, true);\n\t\t} else if (bitsPerSample === 8) {\n\t\t\tsamples[i] = (view.getUint8(p) - 128) / 128;\n\t\t} else if (bitsPerSample === 16) {\n\t\t\tsamples[i] = view.getInt16(p, true) / 32768;\n\t\t} else if (bitsPerSample === 24) {\n\t\t\tconst lo = view.getUint8(p) | (view.getUint8(p + 1) << 8);\n\t\t\tlet hi = view.getUint8(p + 2);\n\t\t\tif (hi & 0x80) hi = hi | 0xffffff00;\n\t\t\tsamples[i] = ((hi << 16) | lo) / 8388608;\n\t\t}\n\t}\n\n\treturn { sampleRate, channels, samples };\n}\n\n/** Mix down to mono by averaging all channels. */\nexport function toMono(wav: WavData): WavData {\n\tif (wav.channels === 1) return wav;\n\tconst len = wav.samples.length / wav.channels;\n\tconst out = new Float32Array(len);\n\tfor (let i = 0; i < len; i++) {\n\t\tlet sum = 0;\n\t\tfor (let c = 0; c < wav.channels; c++)\n\t\t\tsum += wav.samples[i * wav.channels + c];\n\t\tout[i] = sum / wav.channels;\n\t}\n\treturn { sampleRate: wav.sampleRate, channels: 1, samples: out };\n}\n\n/** Linear interpolation resample to targetRate. Expects mono input. */\nexport function resample(wav: WavData, targetRate: number): WavData {\n\tif (wav.sampleRate === targetRate) return wav;\n\tconst ratio = wav.sampleRate / targetRate;\n\tconst outLen = Math.floor(wav.samples.length / ratio);\n\tconst out = new Float32Array(outLen);\n\tconst src = wav.samples;\n\tfor (let i = 0; i < outLen; i++) {\n\t\tconst x = i * ratio;\n\t\tconst xi = Math.floor(x);\n\t\tconst frac = x - xi;\n\t\tout[i] = (src[xi] ?? 0) + ((src[xi + 1] ?? 0) - (src[xi] ?? 0)) * frac;\n\t}\n\treturn { sampleRate: targetRate, channels: 1, samples: out };\n}\n\n/** Convert Float32 [-1,1] samples to Int16 PCM. */\nexport function toInt16(samples: Float32Array): Int16Array {\n\tconst out = new Int16Array(samples.length);\n\tfor (let i = 0; i < samples.length; i++) {\n\t\tout[i] = Math.round(Math.max(-1, Math.min(1, samples[i])) * 32767);\n\t}\n\treturn out;\n}\n\n/** Normalize then convert a WAV to 48kHz/16bit/mono Int16 PCM. */\nexport function normalizePcm(buf: ArrayBuffer): Int16Array {\n\tconst wav = parseWav(buf);\n\tconst mono = toMono(wav);\n\tconst resampled = resample(mono, 48000);\n\treturn toInt16(resampled.samples);\n}\n\nfunction readFourCC(view: DataView, pos: number): string {\n\treturn String.fromCharCode(\n\t\tview.getUint8(pos),\n\t\tview.getUint8(pos + 1),\n\t\tview.getUint8(pos + 2),\n\t\tview.getUint8(pos + 3),\n\t);\n}\n","const SAMPLE_RATE = 48000;\n\nconst NAME_SEMITONE: Record<string, number> = {\n\tc: 0,\n\td: 2,\n\te: 4,\n\tf: 5,\n\tg: 7,\n\ta: 9,\n\tb: 11,\n};\n\n/** Parse a note name like \"E4\", \"G#4\", \"Db5\" → frequency in Hz (null if invalid). */\nexport function noteNameToHz(name: string): number | null {\n\tconst m = /^([A-Ga-g])([#b]?)(-?\\d+)$/.exec(name);\n\tif (!m) return null;\n\tlet semi = NAME_SEMITONE[m[1].toLowerCase()];\n\tif (m[2] === \"#\") semi++;\n\telse if (m[2] === \"b\") semi--;\n\tconst midi = (parseInt(m[3], 10) + 1) * 12 + semi;\n\treturn 440 * 2 ** ((midi - 69) / 12);\n}\n\n/** Recorded pitch encoded in a multi-pitch alias suffix: \"a い_E4\" → 329.63 Hz. */\nexport function pitchFromAliasSuffix(alias: string): number | null {\n\tconst m = /_([A-Ga-g][#b]?-?\\d+)$/.exec(alias);\n\treturn m ? noteNameToHz(m[1]) : null;\n}\n\n/**\n * Estimate the fundamental frequency (Hz) of a voiced region by normalized\n * autocorrelation. Returns 0 when no clear pitch is found (unvoiced consonant,\n * silence, or a region too short to analyse).\n *\n * The signal is decimated to a lower analysis rate for speed; f0 below ~700 Hz\n * is well within the resulting Nyquist limit. A parabolic interpolation around\n * the best lag gives sub-sample (sub-semitone) accuracy.\n */\nexport function detectF0(pcm: Int16Array, start: number, end: number): number {\n\tconst DECIM = 4;\n\tconst sr = SAMPLE_RATE / DECIM; // 12 kHz analysis rate\n\tconst minLag = Math.floor(sr / 700); // highest f0 we look for\n\tconst maxLag = Math.floor(sr / 70); // lowest f0 we look for\n\n\tconst outLen = Math.floor((end - start) / DECIM);\n\tif (outLen < maxLag + 2) return 0;\n\tconst win = Math.min(outLen, 1500); // up to ~0.125 s of vowel\n\n\t// Decimate (box-average) + remove DC, and build a prefix sum of squares so\n\t// each lag's window energy is O(1).\n\tconst buf = new Float32Array(win);\n\tlet mean = 0;\n\tfor (let i = 0; i < win; i++) {\n\t\tlet s = 0;\n\t\tconst base = start + i * DECIM;\n\t\tfor (let j = 0; j < DECIM; j++) s += pcm[base + j];\n\t\tbuf[i] = s;\n\t\tmean += s;\n\t}\n\tmean /= win;\n\n\tconst sq = new Float64Array(win + 1);\n\tfor (let i = 0; i < win; i++) {\n\t\tbuf[i] -= mean;\n\t\tsq[i + 1] = sq[i] + buf[i] * buf[i];\n\t}\n\tif (sq[win] < 1) return 0; // effectively silent\n\n\t// Normalized autocorrelation: 2·r(lag) / (E[0..n] + E[lag..lag+n]) ∈ [-1, 1].\n\tconst norm = (lag: number): number => {\n\t\tconst n = win - lag;\n\t\tlet r = 0;\n\t\tfor (let i = 0; i < n; i++) r += buf[i] * buf[i + lag];\n\t\tconst e = sq[n] + (sq[lag + n] - sq[lag]);\n\t\treturn e > 0 ? (2 * r) / e : 0;\n\t};\n\n\tlet bestLag = -1;\n\tlet best = 0;\n\tfor (let lag = minLag; lag <= maxLag; lag++) {\n\t\tconst v = norm(lag);\n\t\tif (v > best) {\n\t\t\tbest = v;\n\t\t\tbestLag = lag;\n\t\t}\n\t}\n\tif (bestLag < 1 || best < 0.4) return 0; // unvoiced / no clear period\n\n\t// Parabolic interpolation around the peak\n\tconst y0 = norm(bestLag - 1);\n\tconst y1 = best;\n\tconst y2 = norm(bestLag + 1);\n\tconst denom = y0 - 2 * y1 + y2;\n\tconst shift = denom !== 0 ? (0.5 * (y0 - y2)) / denom : 0;\n\n\treturn sr / (bestLag + shift);\n}\n","import type { Manifest, PhonemeEntry } from \"../types.js\";\nimport type { OtoEntry } from \"./parse-oto.js\";\nimport { detectF0 } from \"./pitch.js\";\n\nconst TARGET_RATE = 48000;\n\nfunction msToSamples(ms: number): number {\n\treturn Math.round((ms / 1000) * TARGET_RATE);\n}\n\nconst clamp = (v: number, lo: number, hi: number) =>\n\tMath.max(lo, Math.min(hi, v));\n\nexport interface PackInput {\n\toto: OtoEntry;\n\t/** Full normalized PCM of the source WAV (48kHz / 16bit / mono) */\n\tpcm: Int16Array;\n\t/** Known recorded pitch in Hz (e.g. from the .frq file). 0/undefined → auto-detect. */\n\trecordedPitch?: number;\n}\n\nexport interface PackOutput {\n\tmanifest: Manifest;\n\t/** Raw PCM blob — Int16 / 48kHz / mono */\n\tbin: ArrayBuffer;\n}\n\nexport interface TrimmedPhoneme {\n\t/** PCM trimmed to the oto region [offset, cutoff] */\n\tpcm: Int16Array;\n\t/** Manifest params relative to the trimmed start (sample 0 = oto offset) */\n\tentry: Omit<PhonemeEntry, \"offset\">;\n}\n\n/**\n * Cut the full WAV PCM down to its usable oto region and recompute parameters\n * relative to the trimmed start.\n *\n * UTAU oto.ini values are all in ms and measured from `offset` (the left blank),\n * except `cutoff` (right blank):\n * - cutoff >= 0 : measured from the END of the file\n * - cutoff < 0 : region length from offset = |cutoff|\n *\n * After trimming, sample 0 == oto offset, so pre/overlap/consonant carry over\n * unchanged (just converted to samples), and the slice length is the region end.\n */\nexport function trimToOto(\n\tpcm: Int16Array,\n\toto: OtoEntry,\n\trecordedPitch = 0,\n): TrimmedPhoneme {\n\tconst full = pcm.length;\n\n\tconst start = clamp(msToSamples(oto.offset), 0, full);\n\tconst end =\n\t\toto.cutoff < 0\n\t\t\t? clamp(start + msToSamples(-oto.cutoff), start, full)\n\t\t\t: clamp(full - msToSamples(oto.cutoff), start, full);\n\n\tconst slice = pcm.subarray(start, end);\n\tconst length = slice.length;\n\n\tconst pre = clamp(msToSamples(oto.pre), 0, length);\n\tconst overlap = clamp(msToSamples(oto.overlap), 0, length);\n\tconst consonant = clamp(msToSamples(oto.consonant), 0, length);\n\n\t// Prefer the known recorded pitch (from the .frq file or alias suffix); only\n\t// fall back to autocorrelation on the sustained vowel when none is available.\n\tconst pitch =\n\t\trecordedPitch > 0\n\t\t\t? recordedPitch\n\t\t\t: detectF0(\n\t\t\t\t\tslice,\n\t\t\t\t\tMath.min(Math.max(pre, consonant), Math.max(0, length - 1)),\n\t\t\t\t\tlength,\n\t\t\t\t);\n\n\treturn {\n\t\tpcm: slice,\n\t\tentry: { length, pre, overlap, consonant, pitch },\n\t};\n}\n\n/**\n * Pack normalized PCM phonemes into voice.bin + manifest.json.\n * Each phoneme is trimmed to its oto region first.\n * Duplicate aliases are silently overwritten by the later entry.\n */\nexport function pack(inputs: PackInput[], referencePitch = 220): PackOutput {\n\tconst phonemes: Record<string, PhonemeEntry> = {};\n\tconst chunks: Int16Array[] = [];\n\tlet byteOffset = 0;\n\n\tfor (const { oto, pcm, recordedPitch } of inputs) {\n\t\tconst { pcm: slice, entry } = trimToOto(pcm, oto, recordedPitch);\n\t\tif (slice.length === 0) continue;\n\n\t\tphonemes[oto.alias] = { offset: byteOffset, ...entry };\n\t\tbyteOffset += slice.byteLength;\n\t\tchunks.push(slice);\n\t}\n\n\tconst bin = new ArrayBuffer(byteOffset);\n\tconst view = new Uint8Array(bin);\n\tlet pos = 0;\n\tfor (const chunk of chunks) {\n\t\tview.set(\n\t\t\tnew Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength),\n\t\t\tpos,\n\t\t);\n\t\tpos += chunk.byteLength;\n\t}\n\n\tconst manifest: Manifest = {\n\t\tsampleRate: 48000,\n\t\treferencePitch,\n\t\tphonemes,\n\t};\n\n\treturn { manifest, bin };\n}\n","/**\n * UTAU `.frq` frequency-analysis files (FREQ0003). The same format OpenUtau\n * reads: an 8-byte header, hop size, then the average fundamental frequency of\n * the recording — exactly the reference pitch we need for correct resampling.\n *\n * Layout:\n * char[8] \"FREQ0003\"\n * int32 hopSize\n * float64 averageF0 ← the recorded pitch in Hz\n * byte[16] (blank)\n * int32 length\n * { float64 f0, float64 amp } × length\n */\nexport function parseFrqAverageF0(buffer: ArrayBuffer): number | null {\n\tif (buffer.byteLength < 20) return null;\n\tconst view = new DataView(buffer);\n\n\tlet header = \"\";\n\tfor (let i = 0; i < 8; i++) header += String.fromCharCode(view.getUint8(i));\n\tif (header !== \"FREQ0003\") return null;\n\n\tconst avg = view.getFloat64(12, true); // little-endian\n\treturn Number.isFinite(avg) && avg > 0 ? avg : null;\n}\n\n/** Map a WAV filename to its sibling frq filename: \"あ.wav\" → \"あ_wav.frq\". */\nexport function frqFileName(wavName: string): string {\n\tconst dot = wavName.lastIndexOf(\".\");\n\tconst base = dot >= 0 ? wavName.slice(0, dot) : wavName;\n\tconst ext = dot >= 0 ? wavName.slice(dot + 1) : \"wav\";\n\treturn 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1
+ {"version":3,"sources":["../src/converter/frq.ts","../src/converter/pitch.ts","../src/converter/pack.ts","../src/converter/parse-oto.ts","../src/converter/wav.ts","../src/koe.ts","../src/engine/voice-bank.ts","../src/engine/index.ts","../src/engine/worldline.ts"],"sourcesContent":["/**\n * UTAU `.frq` frequency-analysis files (FREQ0003). The same format OpenUtau\n * reads: an 8-byte header, hop size, then the average fundamental frequency of\n * the recording — exactly the reference pitch we need for correct resampling.\n *\n * Layout:\n * char[8] \"FREQ0003\"\n * int32 hopSize\n * float64 averageF0 ← the recorded pitch in Hz\n * byte[16] (blank)\n * int32 length\n * { float64 f0, float64 amp } × length\n */\nexport function parseFrqAverageF0(buffer: ArrayBuffer): number | null {\n\tif (buffer.byteLength < 20) return null;\n\tconst view = new DataView(buffer);\n\n\tlet header = \"\";\n\tfor (let i = 0; i < 8; i++) header += String.fromCharCode(view.getUint8(i));\n\tif (header !== \"FREQ0003\") return null;\n\n\tconst avg = view.getFloat64(12, true); // little-endian\n\treturn Number.isFinite(avg) && avg > 0 ? avg : null;\n}\n\n/** Map a WAV filename to its sibling frq filename: \"あ.wav\" → \"あ_wav.frq\". */\nexport function frqFileName(wavName: string): string {\n\tconst dot = wavName.lastIndexOf(\".\");\n\tconst base = dot >= 0 ? wavName.slice(0, dot) : wavName;\n\tconst ext = dot >= 0 ? wavName.slice(dot + 1) : \"wav\";\n\treturn `${base}_${ext}.frq`;\n}\n","const SAMPLE_RATE = 48000;\n\nconst NAME_SEMITONE: Record<string, number> = {\n\tc: 0,\n\td: 2,\n\te: 4,\n\tf: 5,\n\tg: 7,\n\ta: 9,\n\tb: 11,\n};\n\n/** Parse a note name like \"E4\", \"G#4\", \"Db5\" → frequency in Hz (null if invalid). */\nexport function noteNameToHz(name: string): number | null {\n\tconst m = /^([A-Ga-g])([#b]?)(-?\\d+)$/.exec(name);\n\tif (!m) return null;\n\tlet semi = NAME_SEMITONE[m[1].toLowerCase()];\n\tif (m[2] === \"#\") semi++;\n\telse if (m[2] === \"b\") semi--;\n\tconst midi = (parseInt(m[3], 10) + 1) * 12 + semi;\n\treturn 440 * 2 ** ((midi - 69) / 12);\n}\n\n/** Recorded pitch encoded in a multi-pitch alias suffix: \"a い_E4\" → 329.63 Hz. */\nexport function pitchFromAliasSuffix(alias: string): number | null {\n\tconst m = /_([A-Ga-g][#b]?-?\\d+)$/.exec(alias);\n\treturn m ? noteNameToHz(m[1]) : null;\n}\n\n/**\n * Estimate the fundamental frequency (Hz) of a voiced region by normalized\n * autocorrelation. Returns 0 when no clear pitch is found (unvoiced consonant,\n * silence, or a region too short to analyse).\n *\n * The signal is decimated to a lower analysis rate for speed; f0 below ~700 Hz\n * is well within the resulting Nyquist limit. A parabolic interpolation around\n * the best lag gives sub-sample (sub-semitone) accuracy.\n */\nexport function detectF0(pcm: Int16Array, start: number, end: number): number {\n\tconst DECIM = 4;\n\tconst sr = SAMPLE_RATE / DECIM; // 12 kHz analysis rate\n\tconst minLag = Math.floor(sr / 700); // highest f0 we look for\n\tconst maxLag = Math.floor(sr / 70); // lowest f0 we look for\n\n\tconst outLen = Math.floor((end - start) / DECIM);\n\tif (outLen < maxLag + 2) return 0;\n\tconst win = Math.min(outLen, 1500); // up to ~0.125 s of vowel\n\n\t// Decimate (box-average) + remove DC, and build a prefix sum of squares so\n\t// each lag's window energy is O(1).\n\tconst buf = new Float32Array(win);\n\tlet mean = 0;\n\tfor (let i = 0; i < win; i++) {\n\t\tlet s = 0;\n\t\tconst base = start + i * DECIM;\n\t\tfor (let j = 0; j < DECIM; j++) s += pcm[base + j];\n\t\tbuf[i] = s;\n\t\tmean += s;\n\t}\n\tmean /= win;\n\n\tconst sq = new Float64Array(win + 1);\n\tfor (let i = 0; i < win; i++) {\n\t\tbuf[i] -= mean;\n\t\tsq[i + 1] = sq[i] + buf[i] * buf[i];\n\t}\n\tif (sq[win] < 1) return 0; // effectively silent\n\n\t// Normalized autocorrelation: 2·r(lag) / (E[0..n] + E[lag..lag+n]) ∈ [-1, 1].\n\tconst norm = (lag: number): number => {\n\t\tconst n = win - lag;\n\t\tlet r = 0;\n\t\tfor (let i = 0; i < n; i++) r += buf[i] * buf[i + lag];\n\t\tconst e = sq[n] + (sq[lag + n] - sq[lag]);\n\t\treturn e > 0 ? (2 * r) / e : 0;\n\t};\n\n\tlet bestLag = -1;\n\tlet best = 0;\n\tfor (let lag = minLag; lag <= maxLag; lag++) {\n\t\tconst v = norm(lag);\n\t\tif (v > best) {\n\t\t\tbest = v;\n\t\t\tbestLag = lag;\n\t\t}\n\t}\n\tif (bestLag < 1 || best < 0.4) return 0; // unvoiced / no clear period\n\n\t// Parabolic interpolation around the peak\n\tconst y0 = norm(bestLag - 1);\n\tconst y1 = best;\n\tconst y2 = norm(bestLag + 1);\n\tconst denom = y0 - 2 * y1 + y2;\n\tconst shift = denom !== 0 ? (0.5 * (y0 - y2)) / denom : 0;\n\n\treturn sr / (bestLag + shift);\n}\n","import type { Manifest, PhonemeEntry } from \"../types.js\";\nimport type { OtoEntry } from \"./parse-oto.js\";\nimport { detectF0 } from \"./pitch.js\";\n\nconst TARGET_RATE = 48000;\n\nfunction msToSamples(ms: number): number {\n\treturn Math.round((ms / 1000) * TARGET_RATE);\n}\n\nconst clamp = (v: number, lo: number, hi: number) =>\n\tMath.max(lo, Math.min(hi, v));\n\nexport interface PackInput {\n\toto: OtoEntry;\n\t/** Full normalized PCM of the source WAV (48kHz / 16bit / mono) */\n\tpcm: Int16Array;\n\t/** Known recorded pitch in Hz (e.g. from the .frq file). 0/undefined → auto-detect. */\n\trecordedPitch?: number;\n}\n\nexport interface PackOutput {\n\tmanifest: Manifest;\n\t/** Raw PCM blob — Int16 / 48kHz / mono */\n\tbin: ArrayBuffer;\n}\n\nexport interface TrimmedPhoneme {\n\t/** PCM trimmed to the oto region [offset, cutoff] */\n\tpcm: Int16Array;\n\t/** Manifest params relative to the trimmed start (sample 0 = oto offset) */\n\tentry: Omit<PhonemeEntry, \"offset\">;\n}\n\n/**\n * Cut the full WAV PCM down to its usable oto region and recompute parameters\n * relative to the trimmed start.\n *\n * UTAU oto.ini values are all in ms and measured from `offset` (the left blank),\n * except `cutoff` (right blank):\n * - cutoff >= 0 : measured from the END of the file\n * - cutoff < 0 : region length from offset = |cutoff|\n *\n * After trimming, sample 0 == oto offset, so pre/overlap/consonant carry over\n * unchanged (just converted to samples), and the slice length is the region end.\n */\nexport function trimToOto(\n\tpcm: Int16Array,\n\toto: OtoEntry,\n\trecordedPitch = 0,\n): TrimmedPhoneme {\n\tconst full = pcm.length;\n\n\tconst start = clamp(msToSamples(oto.offset), 0, full);\n\tconst end =\n\t\toto.cutoff < 0\n\t\t\t? clamp(start + msToSamples(-oto.cutoff), start, full)\n\t\t\t: clamp(full - msToSamples(oto.cutoff), start, full);\n\n\tconst slice = pcm.subarray(start, end);\n\tconst length = slice.length;\n\n\tconst pre = clamp(msToSamples(oto.pre), 0, length);\n\tconst overlap = clamp(msToSamples(oto.overlap), 0, length);\n\tconst consonant = clamp(msToSamples(oto.consonant), 0, length);\n\n\t// Prefer the known recorded pitch (from the .frq file or alias suffix); only\n\t// fall back to autocorrelation on the sustained vowel when none is available.\n\tconst pitch =\n\t\trecordedPitch > 0\n\t\t\t? recordedPitch\n\t\t\t: detectF0(\n\t\t\t\t\tslice,\n\t\t\t\t\tMath.min(Math.max(pre, consonant), Math.max(0, length - 1)),\n\t\t\t\t\tlength,\n\t\t\t\t);\n\n\treturn {\n\t\tpcm: slice,\n\t\tentry: { length, pre, overlap, consonant, pitch },\n\t};\n}\n\n/**\n * Pack normalized PCM phonemes into voice.bin + manifest.json.\n * Each phoneme is trimmed to its oto region first.\n * Duplicate aliases are silently overwritten by the later entry.\n */\nexport function pack(inputs: PackInput[], referencePitch = 220): PackOutput {\n\tconst phonemes: Record<string, PhonemeEntry> = {};\n\tconst chunks: Int16Array[] = [];\n\tlet byteOffset = 0;\n\n\tfor (const { oto, pcm, recordedPitch } of inputs) {\n\t\tconst { pcm: slice, entry } = trimToOto(pcm, oto, recordedPitch);\n\t\tif (slice.length === 0) continue;\n\n\t\tphonemes[oto.alias] = { offset: byteOffset, ...entry };\n\t\tbyteOffset += slice.byteLength;\n\t\tchunks.push(slice);\n\t}\n\n\tconst bin = new ArrayBuffer(byteOffset);\n\tconst view = new Uint8Array(bin);\n\tlet pos = 0;\n\tfor (const chunk of chunks) {\n\t\tview.set(\n\t\t\tnew Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength),\n\t\t\tpos,\n\t\t);\n\t\tpos += chunk.byteLength;\n\t}\n\n\tconst manifest: Manifest = {\n\t\tsampleRate: 48000,\n\t\treferencePitch,\n\t\tphonemes,\n\t};\n\n\treturn { manifest, bin };\n}\n","export interface OtoEntry {\n\t/** Source WAV filename */\n\twav: string;\n\t/** Phoneme alias */\n\talias: string;\n\t/** Left blank — offset from WAV start (ms) */\n\toffset: number;\n\t/** Consonant portion end from offset (ms) */\n\tconsonant: number;\n\t/** Right blank — negative = from WAV end, positive = from offset (ms) */\n\tcutoff: number;\n\t/** Preutterance from offset (ms) */\n\tpre: number;\n\t/** Overlap / crossfade region (ms) */\n\toverlap: number;\n}\n\n/**\n * Parse oto.ini content (already decoded to UTF-8 string).\n * Silently skips malformed lines.\n */\nexport function parseOto(content: string): OtoEntry[] {\n\tconst entries: OtoEntry[] = [];\n\n\tfor (const raw of content.split(/\\r?\\n/)) {\n\t\tconst line = raw.trim();\n\t\tif (!line || line.startsWith(\"#\")) continue;\n\n\t\tconst eq = line.indexOf(\"=\");\n\t\tif (eq === -1) continue;\n\n\t\tconst wav = line.slice(0, eq).trim();\n\t\tconst parts = line.slice(eq + 1).split(\",\");\n\t\tif (parts.length < 6) continue;\n\n\t\tconst [alias, offsetStr, consonantStr, cutoffStr, preStr, overlapStr] =\n\t\t\tparts;\n\t\t// Empty alias → use filename without extension (UTAU spec default)\n\t\tconst aliasStr = alias.trim() || wav.replace(/\\.[^.]+$/, \"\");\n\t\tconst entry: OtoEntry = {\n\t\t\twav,\n\t\t\talias: aliasStr,\n\t\t\toffset: parseFloat(offsetStr) || 0,\n\t\t\tconsonant: parseFloat(consonantStr) || 0,\n\t\t\tcutoff: parseFloat(cutoffStr) || 0,\n\t\t\tpre: parseFloat(preStr) || 0,\n\t\t\toverlap: parseFloat(overlapStr) || 0,\n\t\t};\n\n\t\tif (!entry.alias) continue;\n\t\tentries.push(entry);\n\t}\n\n\treturn entries;\n}\n","export interface WavData {\n\tsampleRate: number;\n\tchannels: number;\n\t/** Normalized samples in [-1, 1], interleaved if multi-channel */\n\tsamples: Float32Array;\n}\n\n/** Parse a WAV file from an ArrayBuffer. Supports PCM 8/16/24-bit and IEEE float 32-bit. */\nexport function parseWav(buf: ArrayBuffer): WavData {\n\tconst view = new DataView(buf);\n\n\tconst riff = readFourCC(view, 0);\n\tif (riff !== \"RIFF\") throw new Error(`Not a RIFF file (got \"${riff}\")`);\n\n\tlet sampleRate = 0;\n\tlet channels = 0;\n\tlet bitsPerSample = 0;\n\tlet audioFormat = 1;\n\tlet dataOffset = 0;\n\tlet dataLength = 0;\n\n\tlet pos = 12;\n\twhile (pos < view.byteLength - 8) {\n\t\tconst id = readFourCC(view, pos);\n\t\tconst size = view.getUint32(pos + 4, true);\n\t\tpos += 8;\n\n\t\tif (id === \"fmt \") {\n\t\t\taudioFormat = view.getUint16(pos, true);\n\t\t\tchannels = view.getUint16(pos + 2, true);\n\t\t\tsampleRate = view.getUint32(pos + 4, true);\n\t\t\tbitsPerSample = view.getUint16(pos + 14, true);\n\t\t\t// WAVE_FORMAT_EXTENSIBLE: the real format code is the first 2 bytes of\n\t\t\t// the SubFormat GUID in the extension.\n\t\t\tif (audioFormat === 0xfffe && size >= 40) {\n\t\t\t\taudioFormat = view.getUint16(pos + 24, true);\n\t\t\t}\n\t\t} else if (id === \"data\") {\n\t\t\tdataOffset = pos;\n\t\t\t// Clamp to the actual file size — a corrupt chunk header could otherwise\n\t\t\t// send sample reads past the end of the buffer.\n\t\t\tdataLength = Math.min(size, view.byteLength - pos);\n\t\t\tbreak;\n\t\t}\n\n\t\tpos += size + (size & 1);\n\t}\n\n\tif (!dataOffset) throw new Error(\"WAV has no data chunk\");\n\tif (!channels || !sampleRate) throw new Error(\"WAV fmt chunk missing\");\n\n\t// Reject anything the sample loop below cannot decode — falling through\n\t// would silently produce all-zero (silent) audio.\n\tconst supported =\n\t\t(audioFormat === 3 && bitsPerSample === 32) ||\n\t\t(audioFormat === 1 &&\n\t\t\t(bitsPerSample === 8 || bitsPerSample === 16 || bitsPerSample === 24));\n\tif (!supported) {\n\t\tthrow new Error(\n\t\t\t`Unsupported WAV format ${audioFormat} / ${bitsPerSample}-bit (need PCM 8/16/24-bit or IEEE float 32-bit)`,\n\t\t);\n\t}\n\n\tconst bytesPerSample = bitsPerSample >> 3;\n\tconst totalSamples = Math.floor(dataLength / bytesPerSample);\n\tconst samples = new Float32Array(totalSamples);\n\n\tfor (let i = 0; i < totalSamples; i++) {\n\t\tconst p = dataOffset + i * bytesPerSample;\n\t\tif (audioFormat === 3) {\n\t\t\tsamples[i] = view.getFloat32(p, true);\n\t\t} else if (bitsPerSample === 8) {\n\t\t\tsamples[i] = (view.getUint8(p) - 128) / 128;\n\t\t} else if (bitsPerSample === 16) {\n\t\t\tsamples[i] = view.getInt16(p, true) / 32768;\n\t\t} else if (bitsPerSample === 24) {\n\t\t\tconst lo = view.getUint8(p) | (view.getUint8(p + 1) << 8);\n\t\t\tlet hi = view.getUint8(p + 2);\n\t\t\tif (hi & 0x80) hi = hi | 0xffffff00;\n\t\t\tsamples[i] = ((hi << 16) | lo) / 8388608;\n\t\t}\n\t}\n\n\treturn { sampleRate, channels, samples };\n}\n\n/** Mix down to mono by averaging all channels. */\nexport function toMono(wav: WavData): WavData {\n\tif (wav.channels === 1) return wav;\n\tconst len = wav.samples.length / wav.channels;\n\tconst out = new Float32Array(len);\n\tfor (let i = 0; i < len; i++) {\n\t\tlet sum = 0;\n\t\tfor (let c = 0; c < wav.channels; c++)\n\t\t\tsum += wav.samples[i * wav.channels + c];\n\t\tout[i] = sum / wav.channels;\n\t}\n\treturn { sampleRate: wav.sampleRate, channels: 1, samples: out };\n}\n\n/** Linear interpolation resample to targetRate. Expects mono input. */\nexport function resample(wav: WavData, targetRate: number): WavData {\n\tif (wav.sampleRate === targetRate) return wav;\n\tconst ratio = wav.sampleRate / targetRate;\n\tconst outLen = Math.floor(wav.samples.length / ratio);\n\tconst out = new Float32Array(outLen);\n\tconst src = wav.samples;\n\tfor (let i = 0; i < outLen; i++) {\n\t\tconst x = i * ratio;\n\t\tconst xi = Math.floor(x);\n\t\tconst frac = x - xi;\n\t\tout[i] = (src[xi] ?? 0) + ((src[xi + 1] ?? 0) - (src[xi] ?? 0)) * frac;\n\t}\n\treturn { sampleRate: targetRate, channels: 1, samples: out };\n}\n\n/** Convert Float32 [-1,1] samples to Int16 PCM. */\nexport function toInt16(samples: Float32Array): Int16Array {\n\tconst out = new Int16Array(samples.length);\n\tfor (let i = 0; i < samples.length; i++) {\n\t\tout[i] = Math.round(Math.max(-1, Math.min(1, samples[i])) * 32767);\n\t}\n\treturn out;\n}\n\n/** Normalize then convert a WAV to 48kHz/16bit/mono Int16 PCM. */\nexport function normalizePcm(buf: ArrayBuffer): Int16Array {\n\tconst wav = parseWav(buf);\n\tconst mono = toMono(wav);\n\tconst resampled = resample(mono, 48000);\n\treturn toInt16(resampled.samples);\n}\n\nfunction readFourCC(view: DataView, pos: number): string {\n\treturn String.fromCharCode(\n\t\tview.getUint8(pos),\n\t\tview.getUint8(pos + 1),\n\t\tview.getUint8(pos + 2),\n\t\tview.getUint8(pos + 3),\n\t);\n}\n","import type { Manifest } from \"./types.js\";\n\n/** Magic number 'KOE\\0' (stored big-endian as 0x4B 0x4F 0x45 0x00). */\nconst MAGIC = 0x4b4f4500;\n\n/**\n * Koe Archive Format (.koe)\n * [4B] magic 'KOE\\0' (big-endian)\n * [4B] JSON length (little-endian)\n * [N ] manifest JSON (UTF-8)\n * [M ] raw PCM (Int16 / 48kHz / mono); phoneme offsets are relative to here\n */\nexport function packKoe(manifest: Manifest, pcmParts: BlobPart[]): Blob {\n\tconst json = new TextEncoder().encode(JSON.stringify(manifest));\n\tconst header = new ArrayBuffer(8);\n\tconst view = new DataView(header);\n\tview.setUint32(0, MAGIC, false); // magic, big-endian\n\tview.setUint32(4, json.byteLength, true); // json length, little-endian\n\treturn new Blob([header, json, ...pcmParts]);\n}\n\n/** Read the 8-byte header → JSON length. Throws on bad magic. */\nexport function parseKoeHeader(headerBytes: ArrayBuffer): {\n\tjsonLength: number;\n} {\n\tconst view = new DataView(headerBytes);\n\tif (view.byteLength < 8 || view.getUint32(0, false) !== MAGIC) {\n\t\tthrow new Error(\"Not a .koe file (bad magic)\");\n\t}\n\treturn { jsonLength: view.getUint32(4, true) };\n}\n\n/** Byte offset where PCM data begins, given the JSON length. */\nexport const pcmBase = (jsonLength: number): number => 8 + jsonLength;\n","import { parseKoeHeader, pcmBase } from \"../koe.js\";\nimport type { Manifest } from \"../types.js\";\n\n/**\n * Supplies raw bytes from the PCM section of a .koe archive on demand.\n * - BlobVoiceSource: slices an in-memory Blob / File (no full-buffer copy)\n * - RangeVoiceSource: HTTP Range requests against a URL (mobile-friendly)\n * `base` is the byte offset where PCM data starts (8 + jsonLength).\n */\ninterface VoiceSource {\n\treadBytes(offset: number, length: number): Promise<ArrayBuffer>;\n}\n\n/** Hard ceiling on a single phoneme's PCM size (~10.5 MB / ~55s at 48kHz). */\nconst MAX_PHONEME_SAMPLES = 5_242_880;\n/** Hard ceiling on the manifest JSON header itself. */\nconst MAX_JSON_LENGTH = 50 * 1024 * 1024;\n\nclass BlobVoiceSource implements VoiceSource {\n\tconstructor(\n\t\tprivate blob: Blob,\n\t\tprivate base: number,\n\t) {}\n\treadBytes(offset: number, length: number): Promise<ArrayBuffer> {\n\t\tconst start = this.base + offset;\n\t\treturn this.blob.slice(start, start + length).arrayBuffer();\n\t}\n}\n\nclass RangeVoiceSource implements VoiceSource {\n\tconstructor(\n\t\tprivate url: string,\n\t\tprivate base: number,\n\t) {}\n\tasync readBytes(offset: number, length: number): Promise<ArrayBuffer> {\n\t\tconst start = this.base + offset;\n\t\treturn rangeFetch(this.url, start, length);\n\t}\n}\n\nasync function rangeFetch(\n\turl: string,\n\tstart: number,\n\tlength: number,\n): Promise<ArrayBuffer> {\n\tconst res = await fetch(url, {\n\t\theaders: { Range: `bytes=${start}-${start + length - 1}` },\n\t\tcredentials: \"omit\", // never leak cookies / auth to a MML-supplied URL\n\t});\n\t// A server that ignores Range and returns 200 with the full file would\n\t// blow past the requested size and exhaust memory; only accept 206.\n\tif (res.status !== 206) {\n\t\tthrow new Error(\n\t\t\t`.koe fetch failed: expected 206 Partial Content, got ${res.status}`,\n\t\t);\n\t}\n\treturn readCapped(res, length);\n}\n\n/**\n * Read a response body without trusting its declared size: a server can answer\n * 206 yet still stream an oversized body. Abort as soon as the read exceeds\n * the requested byte count, before buffering the excess.\n */\nasync function readCapped(res: Response, length: number): Promise<ArrayBuffer> {\n\tconst reader = res.body?.getReader();\n\tif (!reader) {\n\t\tconst buf = await res.arrayBuffer();\n\t\tif (buf.byteLength > length) {\n\t\t\tthrow new Error(\n\t\t\t\t`.koe fetch failed: response exceeds requested ${length} bytes`,\n\t\t\t);\n\t\t}\n\t\treturn buf;\n\t}\n\tconst out = new Uint8Array(length);\n\tlet received = 0;\n\tfor (;;) {\n\t\tconst { done, value } = await reader.read();\n\t\tif (done) break;\n\t\tif (received + value.byteLength > length) {\n\t\t\tawait reader.cancel();\n\t\t\tthrow new Error(\n\t\t\t\t`.koe fetch failed: response exceeds requested ${length} bytes`,\n\t\t\t);\n\t\t}\n\t\tout.set(value, received);\n\t\treceived += value.byteLength;\n\t}\n\treturn received === length ? out.buffer : out.buffer.slice(0, received);\n}\n\nfunction validateJsonLength(jsonLength: number): void {\n\tif (\n\t\t!Number.isInteger(jsonLength) ||\n\t\tjsonLength < 0 ||\n\t\tjsonLength > MAX_JSON_LENGTH\n\t) {\n\t\tthrow new Error(`manifest JSON length out of bounds: ${jsonLength}`);\n\t}\n}\n\nfunction parseManifest(json: ArrayBuffer): Manifest {\n\tconst manifest = JSON.parse(new TextDecoder().decode(json)) as Manifest;\n\tif (\n\t\t!manifest ||\n\t\ttypeof manifest !== \"object\" ||\n\t\ttypeof manifest.phonemes !== \"object\" ||\n\t\tmanifest.phonemes === null\n\t) {\n\t\tthrow new Error(\"invalid manifest: missing phonemes table\");\n\t}\n\treturn manifest;\n}\n\n/**\n * Read-only access to a .koe voice bank: its manifest plus per-phoneme PCM,\n * fetched on demand (Blob slice or HTTP Range). The full bank is never held in\n * memory.\n *\n * Pure data — no AudioContext, no AudioWorklet, no DOM. Use this when you only\n * need the source samples (e.g. to feed the {@link Worldline} renderer or any\n * other vocoder). {@link KoeEngine} builds its concatenative playback on top of\n * this same class.\n *\n * const bank = await VoiceBank.load(koeBlobOrUrl);\n * const pcm = await bank.getPcm('a'); // Float64 [-1, 1]\n */\nexport class VoiceBank {\n\tprivate constructor(\n\t\t/** The voice bank manifest (sample rate, reference pitch, phoneme table). */\n\t\treadonly manifest: Manifest,\n\t\tprivate source: VoiceSource,\n\t) {}\n\n\t/**\n\t * Parse a .koe archive header + manifest and bind a lazy PCM source.\n\t * @param koe a Blob/File of the .koe archive, or a URL (served with Range support)\n\t */\n\tstatic async load(koe: Blob | string): Promise<VoiceBank> {\n\t\ttry {\n\t\t\tif (typeof koe === \"string\") {\n\t\t\t\t// Range requests on blob: URLs are not portable (Firefox/Safari answer\n\t\t\t\t// 200); the data already lives in memory, so read it back as a Blob.\n\t\t\t\tif (/^blob:/i.test(koe)) {\n\t\t\t\t\tconst res = await fetch(koe);\n\t\t\t\t\tif (!res.ok) {\n\t\t\t\t\t\tthrow new Error(`blob: URL fetch failed: ${res.status}`);\n\t\t\t\t\t}\n\t\t\t\t\treturn await VoiceBank.fromBlob(await res.blob());\n\t\t\t\t}\n\t\t\t\tif (!/^https?:/i.test(koe)) {\n\t\t\t\t\tthrow new Error(`unsupported URL protocol: ${koe}`);\n\t\t\t\t}\n\t\t\t\tconst header = await rangeFetch(koe, 0, 8);\n\t\t\t\tconst { jsonLength } = parseKoeHeader(header);\n\t\t\t\tvalidateJsonLength(jsonLength);\n\t\t\t\tconst json = await rangeFetch(koe, 8, jsonLength);\n\t\t\t\tconst manifest = parseManifest(json);\n\t\t\t\treturn new VoiceBank(\n\t\t\t\t\tmanifest,\n\t\t\t\t\tnew RangeVoiceSource(koe, pcmBase(jsonLength)),\n\t\t\t\t);\n\t\t\t}\n\t\t\treturn await VoiceBank.fromBlob(koe);\n\t\t} catch (error) {\n\t\t\tthrow new Error(\n\t\t\t\t`Failed to load .koe voice bank: ${error instanceof Error ? error.message : String(error)}`,\n\t\t\t);\n\t\t}\n\t}\n\n\tprivate static async fromBlob(koe: Blob): Promise<VoiceBank> {\n\t\tconst header = await koe.slice(0, 8).arrayBuffer();\n\t\tconst { jsonLength } = parseKoeHeader(header);\n\t\tvalidateJsonLength(jsonLength);\n\t\tconst json = await koe.slice(8, 8 + jsonLength).arrayBuffer();\n\t\tconst manifest = parseManifest(json);\n\t\treturn new VoiceBank(\n\t\t\tmanifest,\n\t\t\tnew BlobVoiceSource(koe, pcmBase(jsonLength)),\n\t\t);\n\t}\n\n\t/** True if the bank contains a phoneme under this alias. */\n\thas(phoneme: string): boolean {\n\t\t// Own-property check: a plain [phoneme] access would also match inherited\n\t\t// Object.prototype keys like \"toString\" / \"constructor\".\n\t\treturn Object.hasOwn(this.manifest.phonemes, phoneme);\n\t}\n\n\t/**\n\t * Raw Int16 PCM bytes (48 kHz / mono) for a phoneme, or null if unknown.\n\t * The returned ArrayBuffer is freshly allocated and safe to transfer to a\n\t * worker / AudioWorklet.\n\t */\n\tasync readPcmBytes(phoneme: string): Promise<ArrayBuffer | null> {\n\t\tif (!Object.hasOwn(this.manifest.phonemes, phoneme)) return null;\n\t\tconst entry = this.manifest.phonemes[phoneme];\n\t\tif (\n\t\t\t!Number.isInteger(entry.offset) ||\n\t\t\t!Number.isInteger(entry.length) ||\n\t\t\tentry.offset < 0 ||\n\t\t\tentry.length < 0 ||\n\t\t\tentry.length > MAX_PHONEME_SAMPLES\n\t\t) {\n\t\t\tthrow new Error(`manifest entry out of bounds for phoneme: ${phoneme}`);\n\t\t}\n\t\treturn this.source.readBytes(entry.offset, entry.length * 2); // Int16 = 2 bytes/sample\n\t}\n\n\t/**\n\t * A phoneme's PCM as a Float64Array normalised to [-1, 1], or null if unknown.\n\t * Intended for external analysis / resynthesis such as the WORLD vocoder.\n\t */\n\tasync getPcm(phoneme: string): Promise<Float64Array | null> {\n\t\tconst buf = await this.readPcmBytes(phoneme);\n\t\tif (!buf) return null;\n\t\t// A truncated read (Blob clamped at EOF) can yield an odd byte count,\n\t\t// which the Int16Array(buffer) constructor rejects — floor to whole samples.\n\t\tconst int16 = new Int16Array(buf, 0, Math.floor(buf.byteLength / 2));\n\t\tconst f64 = new Float64Array(int16.length);\n\t\tfor (let i = 0; i < int16.length; i++) f64[i] = int16[i] / 32768;\n\t\treturn f64;\n\t}\n}\n","import type { Manifest, NoteEvent } from \"../types.js\";\nimport { VoiceBank } from \"./voice-bank.js\";\n\nexport type { NoteEvent };\n\nexport interface KoeEngineOptions {\n\t/** URL to koe-worklet.js. Defaults to './koe-worklet.js'. */\n\tworkletUrl?: string;\n}\n\n/**\n * Main-thread API for the koe concatenative synthesis engine.\n *\n * Loads a single .koe archive. The full voice bank is never held in memory:\n * phonemes are fetched on demand (via {@link VoiceBank}) and cached in the\n * audio thread.\n *\n * const engine = new KoeEngine();\n * await engine.load(koeBlobOrUrl);\n * await engine.play([{ phoneme: 'a', pitch: 440, duration: 48000 }]);\n *\n * For analysis / resynthesis (e.g. {@link Worldline}) you usually only need the\n * raw samples — use {@link VoiceBank} directly instead, which needs no\n * AudioContext or worklet.\n */\nexport class KoeEngine {\n\tprivate ctx: AudioContext;\n\tprivate workletUrl: string;\n\tprivate node: AudioWorkletNode | null = null;\n\tprivate bank: VoiceBank | null = null;\n\tprivate delivered = new Set<string>();\n\tprivate pending = new Map<string, Promise<void>>();\n\n\tconstructor(options: KoeEngineOptions = {}) {\n\t\tthis.ctx = new AudioContext({ sampleRate: 48000 });\n\t\tthis.workletUrl = options.workletUrl ?? \"./koe-worklet.js\";\n\t}\n\n\tget audioContext(): AudioContext {\n\t\treturn this.ctx;\n\t}\n\n\tget manifest(): Manifest | null {\n\t\treturn this.bank?.manifest ?? null;\n\t}\n\n\t/** The underlying voice bank (manifest + on-demand PCM), or null before load(). */\n\tget voiceBank(): VoiceBank | null {\n\t\treturn this.bank;\n\t}\n\n\t/**\n\t * Register the worklet and bind a .koe voice bank.\n\t * @param koe a Blob/File of the .koe archive, or a URL (served with Range support)\n\t */\n\tasync load(koe: Blob | string): Promise<void> {\n\t\tawait this.ctx.audioWorklet.addModule(this.workletUrl);\n\n\t\t// Re-loading: tear down the previous node, or both would keep playing.\n\t\tthis.node?.disconnect();\n\t\tthis.node = null;\n\n\t\tthis.bank = await VoiceBank.load(koe);\n\t\tthis.delivered.clear();\n\t\tthis.pending.clear();\n\n\t\tthis.node = new AudioWorkletNode(this.ctx, \"koe-processor\", {\n\t\t\tnumberOfInputs: 0,\n\t\t\tnumberOfOutputs: 1,\n\t\t\toutputChannelCount: [1],\n\t\t});\n\t\tthis.node.port.postMessage({ type: \"init\", manifest: this.bank.manifest });\n\t\tthis.node.connect(this.ctx.destination);\n\n\t\tconsole.log(\n\t\t\t\"[koe] ready —\",\n\t\t\tObject.keys(this.bank.manifest.phonemes).length,\n\t\t\t\"phonemes (on-demand)\",\n\t\t);\n\t}\n\n\t/** Fetch one phoneme's PCM and deliver it to the worklet (deduped, cached). */\n\tprivate ensurePhoneme(name: string): Promise<void> {\n\t\tif (this.delivered.has(name)) return Promise.resolve();\n\t\tconst existing = this.pending.get(name);\n\t\tif (existing) return existing;\n\t\tif (!this.bank || !this.node) return Promise.resolve();\n\n\t\tconst load = this.bank\n\t\t\t.readPcmBytes(name)\n\t\t\t.then((buf) => {\n\t\t\t\tif (!buf || !this.node) return;\n\t\t\t\tthis.node.port.postMessage({ type: \"phoneme\", name, buffer: buf }, [\n\t\t\t\t\tbuf,\n\t\t\t\t]);\n\t\t\t\tthis.delivered.add(name);\n\t\t\t})\n\t\t\t.finally(() => {\n\t\t\t\t// Always clear pending — a failed fetch must stay retryable, or the\n\t\t\t\t// rejected promise would poison every later play() of this phoneme.\n\t\t\t\tthis.pending.delete(name);\n\t\t\t});\n\n\t\tthis.pending.set(name, load);\n\t\treturn load;\n\t}\n\n\t/** Stop current playback, preload the phonemes for `notes`, then queue them. */\n\tasync play(notes: NoteEvent[]): Promise<void> {\n\t\tif (!this.node) throw new Error(\"KoeEngine: call load() before play()\");\n\t\tthis.node.port.postMessage({ type: \"stop\" });\n\t\tconst names = [...new Set(notes.map((n) => n.phoneme))].filter(Boolean);\n\t\tawait Promise.all(names.map((n) => this.ensurePhoneme(n)));\n\t\tthis.node.port.postMessage({ type: \"play\", notes });\n\t}\n\n\t/** Stop playback and clear the queue. */\n\tstop(): void {\n\t\tthis.node?.port.postMessage({ type: \"stop\" });\n\t}\n\n\t/** Resume the AudioContext if suspended (e.g. after autoplay block). */\n\tasync resume(): Promise<void> {\n\t\tif (this.ctx.state === \"suspended\") await this.ctx.resume();\n\t}\n\n\t/**\n\t * Tear down the worklet node and close the AudioContext, releasing the audio\n\t * hardware. The engine cannot be reused afterwards — create a new one.\n\t */\n\tasync dispose(): Promise<void> {\n\t\tthis.node?.disconnect();\n\t\tthis.node = null;\n\t\tthis.bank = null;\n\t\tthis.delivered.clear();\n\t\tthis.pending.clear();\n\t\tif (this.ctx.state !== \"closed\") await this.ctx.close();\n\t}\n\n\t/**\n\t * Read a phoneme's raw PCM and return it as a Float64Array normalised to\n\t * [-1, 1]. Convenience that forwards to the underlying {@link VoiceBank}.\n\t * Intended for external analysis such as the WORLD vocoder.\n\t */\n\tasync getPcm(phoneme: string): Promise<Float64Array | null> {\n\t\treturn this.bank?.getPcm(phoneme) ?? null;\n\t}\n}\n","import type { PhonemeEntry } from \"../types.js\";\n\n/** Output sample rate of the worldline synthesizer. */\nexport const WORLDLINE_SAMPLE_RATE = 48000;\n\n/**\n * WORLD needs roughly 85 ms of audio for stable F0 analysis. Phonemes with\n * fewer samples than this are rejected (renderNote returns null).\n */\nexport const MIN_WORLDLINE_SAMPLES = 4096;\n\n/** SynthRequest struct layout — 120 bytes, WASM32. */\nconst SYNTH_REQ_SIZE = 120;\n/** Frame period hardcoded in worldline's phrase_synth.cpp. */\nconst WL_FRAME_MS = 10;\n\n/**\n * The subset of the Emscripten module surface that we call. worldline.js is an\n * MODULARIZE=1 / EXPORT_NAME=WorldlineModule build of OpenUtau's worldline.\n */\ninterface WorldlineWasm {\n\t_PhraseSynthNew(): number;\n\t_PhraseSynthDelete(ps: number): void;\n\t_PhraseSynthAddRequest(\n\t\tps: number,\n\t\treq: number,\n\t\tposMs: number,\n\t\tskipMs: number,\n\t\tlengthMs: number,\n\t\tfadeInMs: number,\n\t\tfadeOutMs: number,\n\t\tflag: number,\n\t): void;\n\t_PhraseSynthSetCurves(\n\t\tps: number,\n\t\tf0: number,\n\t\tgender: number,\n\t\ttension: number,\n\t\tbreathiness: number,\n\t\tvoicing: number,\n\t\tlength: number,\n\t\tframeMs: number,\n\t): void;\n\t_PhraseSynthSynth(ps: number, yPtrPtr: number, flag: number): number;\n\t_malloc(size: number): number;\n\t_free(ptr: number): void;\n\tsetValue(ptr: number, value: number, type: string): void;\n\tgetValue(ptr: number, type: string): number;\n\tHEAPF32: Float32Array;\n\tHEAPF64: Float64Array;\n}\n\ntype WorldlineFactory = (opts?: {\n\tlocateFile?: (path: string) => string;\n}) => Promise<WorldlineWasm>;\n\ndeclare global {\n\t// Injected by worldline.js when loaded as a classic <script> or via importScripts.\n\t// eslint-disable-next-line no-var\n\tvar WorldlineModule: WorldlineFactory | undefined;\n\t// Present in classic Web Workers (used to load worldline.js off the main thread).\n\t// eslint-disable-next-line no-var\n\tvar importScripts: ((...urls: string[]) => void) | undefined;\n}\n\nexport interface WorldlineLoadOptions {\n\t/**\n\t * URL of `worldline.js` (the Emscripten loader). The matching\n\t * `worldline.wasm` must sit next to it — it is resolved relative to this URL.\n\t *\n\t * When consuming koe from npm + a CDN this is typically e.g.\n\t * `https://cdn.jsdelivr.net/npm/@onjmin/koe/dist/world/worldline.js`, or your\n\t * own hosted copy of `dist/world/`.\n\t */\n\tscriptUrl: string;\n}\n\nexport interface RenderNoteParams {\n\t/**\n\t * Source phoneme PCM normalised to [-1, 1] (e.g. from\n\t * `VoiceBank.getPcm()` / `KoeEngine.getPcm()`).\n\t */\n\tpcm: Float64Array;\n\t/** Target output pitch in Hz. */\n\tpitch: number;\n\t/** Sustain / vowel duration in ms (the lead-in below is rendered on top). */\n\tdurationMs: number;\n\t/** Preutterance / lead-in in ms — convert from {@link PhonemeEntry.pre}. */\n\tpreMs: number;\n\t/** Consonant length in ms — convert from {@link PhonemeEntry.consonant}. */\n\tconsonantMs: number;\n\t/** Reference tempo in BPM for worldline's internal timing. Default 120. */\n\ttempo?: number;\n}\n\n/** Convert a sample count at 48 kHz to milliseconds. */\nexport const samplesToMs = (samples: number): number =>\n\t(samples / WORLDLINE_SAMPLE_RATE) * 1000;\n\n/**\n * Derive {@link RenderNoteParams} lead-in / consonant fields from a manifest\n * entry, so callers don't repeat the sample→ms conversion.\n *\n * const params = { pcm, pitch, durationMs, ...leadInFromEntry(entry) };\n */\nexport function leadInFromEntry(entry: PhonemeEntry): {\n\tpreMs: number;\n\tconsonantMs: number;\n} {\n\treturn {\n\t\tpreMs: samplesToMs(entry.pre || 0),\n\t\tconsonantMs: samplesToMs(entry.consonant || 0),\n\t};\n}\n\n// Cache module factories by script URL so repeated loads share one WASM instance.\nconst moduleCache = new Map<string, Promise<WorldlineWasm>>();\n\nfunction injectScript(src: string): Promise<void> {\n\treturn new Promise((resolve, reject) => {\n\t\tconst existing = document.querySelector<HTMLScriptElement>(\n\t\t\t`script[data-koe-worldline=\"${src}\"]`,\n\t\t);\n\t\tif (existing) {\n\t\t\tresolve();\n\t\t\treturn;\n\t\t}\n\t\tconst s = document.createElement(\"script\");\n\t\ts.src = src;\n\t\ts.dataset.koeWorldline = src;\n\t\ts.onload = () => resolve();\n\t\ts.onerror = () => reject(new Error(`worldline: failed to load ${src}`));\n\t\tdocument.head.appendChild(s);\n\t});\n}\n\nfunction loadWasm(scriptUrl: string): Promise<WorldlineWasm> {\n\tconst cached = moduleCache.get(scriptUrl);\n\tif (cached) return cached;\n\n\tconst baseUrl = scriptUrl.slice(0, scriptUrl.lastIndexOf(\"/\") + 1);\n\tconst instantiate = (): Promise<WorldlineWasm> => {\n\t\tconst factory = globalThis.WorldlineModule;\n\t\tif (!factory)\n\t\t\tthrow new Error(\n\t\t\t\t\"worldline: WorldlineModule global was not defined by the script\",\n\t\t\t);\n\t\t// locateFile resolves worldline.wasm next to the script; fetch works in both\n\t\t// the DOM and Web Workers, so the WASM instantiates the same way off-thread.\n\t\treturn factory({ locateFile: (f: string) => baseUrl + f });\n\t};\n\n\tlet promise: Promise<WorldlineWasm>;\n\tif (typeof document !== \"undefined\") {\n\t\t// Main thread / DOM: load worldline.js via a <script> tag.\n\t\tpromise = injectScript(scriptUrl).then(instantiate);\n\t} else if (typeof globalThis.importScripts === \"function\") {\n\t\t// Classic Web Worker: importScripts runs worldline.js synchronously in the\n\t\t// worker's global scope, defining globalThis.WorldlineModule. This lets the\n\t\t// heavy WORLD synthesis run off the main thread.\n\t\tpromise = Promise.resolve().then(() => {\n\t\t\t(globalThis.importScripts as (...urls: string[]) => void)(scriptUrl);\n\t\t\treturn instantiate();\n\t\t});\n\t} else {\n\t\treturn Promise.reject(\n\t\t\tnew Error(\n\t\t\t\t\"Worldline.load requires a DOM or a classic Web Worker (importScripts) to load worldline.js\",\n\t\t\t),\n\t\t);\n\t}\n\tmoduleCache.set(scriptUrl, promise);\n\treturn promise;\n}\n\n/**\n * High-quality WORLD-vocoder note renderer (OpenUtau's worldline via WASM).\n *\n * Pure synthesis: PCM in → PCM out. No AudioContext, no scheduling — the caller\n * owns playback (schedule the returned buffer on its own timeline). Pair it with\n * {@link VoiceBank} for the source samples:\n *\n * const bank = await VoiceBank.load(koeUrl);\n * const wl = await Worldline.load({ scriptUrl: '.../world/worldline.js' });\n *\n * const entry = bank.manifest.phonemes[alias];\n * const pcm = await bank.getPcm(alias);\n * const audio = wl.renderNote({\n * pcm, pitch: 440, durationMs: 500, ...leadInFromEntry(entry),\n * });\n * // audio: Float32 @ 48 kHz, layout [lead-in/consonant ≈ preMs][vowel ≈ durationMs]\n */\nexport class Worldline {\n\treadonly sampleRate = WORLDLINE_SAMPLE_RATE;\n\n\tprivate constructor(private wasm: WorldlineWasm) {}\n\n\t/**\n\t * Load + instantiate the worldline WASM module (deduped per scriptUrl).\n\t *\n\t * Works on the main thread (loads via `<script>`) and inside a classic Web\n\t * Worker (loads via `importScripts`), so the heavy synthesis can run\n\t * off-thread. The matching `worldline.wasm` is fetched next to scriptUrl.\n\t */\n\tstatic async load(options: WorldlineLoadOptions): Promise<Worldline> {\n\t\treturn new Worldline(await loadWasm(options.scriptUrl));\n\t}\n\n\t/**\n\t * Render one note to Float32 PCM at 48 kHz.\n\t *\n\t * The output buffer is laid out as [lead-in/consonant ≈ preMs][vowel ≈\n\t * durationMs], rendered from sample offset 0 (no leading silence). The vowel\n\t * onset (the \"beat\") sits at ≈ preMs into the buffer, so a sequencer should\n\t * place the buffer at `beatTime − preMs` and may trim/crossfade the lead-in.\n\t *\n\t * No internal crossfade is applied — apply fades externally.\n\t *\n\t * @returns Float32 PCM, or null when `pcm` is shorter than\n\t * {@link MIN_WORLDLINE_SAMPLES} (too short for stable F0 analysis).\n\t */\n\trenderNote(params: RenderNoteParams): Float32Array | null {\n\t\tconst { pcm, pitch, durationMs, preMs, consonantMs, tempo = 120 } = params;\n\t\tif (!pcm || pcm.length < MIN_WORLDLINE_SAMPLES) return null;\n\n\t\tconst WL = this.wasm;\n\t\tconst FS = WORLDLINE_SAMPLE_RATE;\n\t\tconst midiNote = Math.round(69 + 12 * Math.log2(pitch / 440));\n\t\tconst posMs = 0; // no leading silence\n\t\tconst reqLen = preMs + durationMs; // render lead-in/consonant + vowel\n\n\t\t// cut_off must trim at least ~2 frames off the tail. PhraseSynth::AddRequest\n\t\t// computes the trim frame count as ceil(in_length_ms)/frame_ms, but\n\t\t// PyinEstimator yields floor(N/nhop)−1 frames — one fewer. With cut_off=0 the\n\t\t// requested frame count exceeds f0_.size() and Model::Trim's f0_.erase() runs\n\t\t// past end(), causing \"memory access out of bounds\". Trimming 2 frames keeps\n\t\t// it within bounds.\n\t\tconst cutMs = WL_FRAME_MS * 2;\n\n\t\tconst ps = WL._PhraseSynthNew();\n\t\tif (!ps) return null;\n\t\tconst reqPtr = WL._malloc(SYNTH_REQ_SIZE);\n\t\tif (!reqPtr) {\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tconst samplePtr = WL._malloc(pcm.length * 8);\n\t\tif (!samplePtr) {\n\t\t\tWL._free(reqPtr);\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tWL.HEAPF64.set(pcm, samplePtr >> 3);\n\n\t\tconst sv = (off: number, val: number, type: string) =>\n\t\t\tWL.setValue(reqPtr + off, val, type);\n\t\tsv(0, FS, \"i32\");\n\t\tsv(4, pcm.length, \"i32\");\n\t\tsv(8, samplePtr, \"*\");\n\t\tsv(12, 0, \"i32\");\n\t\tsv(16, 0, \"*\");\n\t\tsv(20, midiNote, \"i32\");\n\t\tsv(24, 100.0, \"double\");\n\t\tsv(32, 0.0, \"double\");\n\t\tsv(40, reqLen, \"double\");\n\t\tsv(48, consonantMs, \"double\");\n\t\tsv(56, cutMs, \"double\");\n\t\tsv(64, 100.0, \"double\");\n\t\tsv(72, 0.0, \"double\");\n\t\tsv(80, tempo, \"double\");\n\t\tsv(88, 0, \"i32\");\n\t\tsv(92, 0, \"*\");\n\t\tsv(96, 0, \"i32\");\n\t\tsv(100, 0, \"i32\");\n\t\tsv(104, 100, \"i32\");\n\t\tsv(108, 0, \"i32\");\n\t\tsv(112, 0, \"i32\");\n\t\tsv(116, 100, \"i32\");\n\n\t\t// skip/fadeIn/fadeOut all 0 — crossfade is handled externally.\n\t\tWL._PhraseSynthAddRequest(ps, reqPtr, posMs, 0.0, reqLen, 0.0, 0.0, 0);\n\t\tWL._free(samplePtr);\n\t\tWL._free(reqPtr);\n\n\t\tconst totalMs = posMs + reqLen + WL_FRAME_MS * 2;\n\t\tconst nFrames = Math.ceil(totalMs / WL_FRAME_MS) + 4;\n\t\tconst f0Arr = new Float64Array(nFrames).fill(pitch);\n\t\tconst gArr = new Float64Array(nFrames).fill(0.5);\n\t\tconst tArr = new Float64Array(nFrames).fill(0.5);\n\t\tconst bArr = new Float64Array(nFrames).fill(0.5);\n\t\tconst vArr = new Float64Array(nFrames).fill(1.0);\n\n\t\tconst f0Ptr = WL._malloc(nFrames * 8);\n\t\tconst gPtr = WL._malloc(nFrames * 8);\n\t\tconst tPtr = WL._malloc(nFrames * 8);\n\t\tconst bPtr = WL._malloc(nFrames * 8);\n\t\tconst vPtr = WL._malloc(nFrames * 8);\n\t\tif (!f0Ptr || !gPtr || !tPtr || !bPtr || !vPtr) {\n\t\t\tif (f0Ptr) WL._free(f0Ptr);\n\t\t\tif (gPtr) WL._free(gPtr);\n\t\t\tif (tPtr) WL._free(tPtr);\n\t\t\tif (bPtr) WL._free(bPtr);\n\t\t\tif (vPtr) WL._free(vPtr);\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tWL.HEAPF64.set(f0Arr, f0Ptr >> 3);\n\t\tWL.HEAPF64.set(gArr, gPtr >> 3);\n\t\tWL.HEAPF64.set(tArr, tPtr >> 3);\n\t\tWL.HEAPF64.set(bArr, bPtr >> 3);\n\t\tWL.HEAPF64.set(vArr, vPtr >> 3);\n\t\tWL._PhraseSynthSetCurves(\n\t\t\tps,\n\t\t\tf0Ptr,\n\t\t\tgPtr,\n\t\t\ttPtr,\n\t\t\tbPtr,\n\t\t\tvPtr,\n\t\t\tnFrames,\n\t\t\tWL_FRAME_MS,\n\t\t);\n\t\tWL._free(f0Ptr);\n\t\tWL._free(gPtr);\n\t\tWL._free(tPtr);\n\t\tWL._free(bPtr);\n\t\tWL._free(vPtr);\n\n\t\tconst yPtrPtr = WL._malloc(4);\n\t\tif (!yPtrPtr) {\n\t\t\tWL._PhraseSynthDelete(ps);\n\t\t\treturn null;\n\t\t}\n\t\tconst outLen = WL._PhraseSynthSynth(ps, yPtrPtr, 0);\n\t\tconst yPtr = WL.getValue(yPtrPtr, 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