@onjmin/koe 1.0.0 → 1.0.3

package/README.md

@@ -1,2 +1,258 @@
-# koe
-ブラウザ上でUTAU音源をラグ無しで再生する技術
+# koe
+
+ブラウザ上でUTAU音源をラグ無しで再生するnpmモジュール。
+
+UTAU の oto.ini で定義された音源を `.koe` アーカイブに変換し、WebAssembly + AudioWorklet でリアルタイム再生・高品質再合成を行う。
+
+- [DEMO](https://onjmin.github.io/koe/demo) koeフォーマット作成もこちらで
+- [npm](https://www.npmjs.com/package/@onjmin/koe)
+
+---
+
+## インストール
+
+```bash
+npm install @onjmin/koe
+# or
+pnpm add @onjmin/koe
+```
+
+---
+
+## 設計概要
+
+```
+UTAU音源 (wav + oto.ini + frq)
+        ↓  変換 (pack / koe-convert CLI)
+    .koe アーカイブ
+    ├── 8byte ヘッダ (magic + JSON長)
+    ├── manifest JSON (音素テーブル・ピッチ情報)
+    └── PCM バイナリ (Int16 / 48kHz / mono)
+        ↓
+ VoiceBank  ──── Blob slice / HTTP Range で音素を必要時のみ取得
+        ├── KoeEngine  ── AudioWorklet で連接合成・リアルタイム再生
+        └── Worldline  ── WORLD ボコーダ (WASM) で高品質ノート合成
+```
+
+- **VoiceBank** : `.koe` ファイルの読み取り専用ビュー。AudioContext 不要。音素 PCM をオンデマンドで取得する。
+- **KoeEngine** : メインスレッド API。`VoiceBank` の上に AudioWorklet を組み合わせた連接合成エンジン。
+- **Worldline** : OpenUtau の worldline WASM を使った高品質ノート合成。PCM を入力して Float32 PCM を返す純粋な合成器。
+
+---
+
+## 使い方
+
+### 1. 音源の変換 (oto.ini → .koe)
+
+CLIコマンド `koe-convert` で UTAU 音源を `.koe` アーカイブに変換する。
+
+```bash
+npx koe-convert <音源フォルダ> -o voice.koe
+```
+
+あるいはコードから変換する:
+
+```ts
+import { parseOto, parseWav, toMono, resample, normalizePcm, pack, packKoe, parseFrqAverageF0, frqFileName, pitchFromAliasSuffix } from "@onjmin/koe";
+
+// oto.ini をパース
+const otoText = await fs.readFile("voice/oto.ini", "utf8");
+const entries = parseOto(otoText);
+
+// 各エントリを PackInput に変換
+const inputs = await Promise.all(entries.map(async (oto) => {
+  const wavBuf = await fs.readFile(`voice/${oto.file}`);
+  const wav = parseWav(wavBuf.buffer);
+  const mono = toMono(wav);
+  const resampled = resample(mono, wav.sampleRate, 48000);
+  const pcm = normalizePcm(resampled); // Int16Array
+
+  // .frq ファイルからピッチ取得 (任意)
+  let recordedPitch = pitchFromAliasSuffix(oto.alias); // エイリアス末尾から推定
+  if (!recordedPitch) {
+    try {
+      const frqBuf = await fs.readFile(`voice/${frqFileName(oto.file)}`);
+      recordedPitch = parseFrqAverageF0(frqBuf.buffer);
+    } catch {}
+  }
+
+  return { oto, pcm, recordedPitch };
+}));
+
+// 変換してパック
+const { manifest, bin } = pack(inputs, 220 /* referencePitch Hz */);
+const blob = packKoe(manifest, [bin]);
+
+// blob を voice.koe として保存 (Node.js)
+const buf = Buffer.from(await blob.arrayBuffer());
+await fs.writeFile("voice.koe", buf);
+```
+
+---
+
+### 2. KoeEngine — リアルタイム再生 (ブラウザ)
+
+AudioWorklet を使った連接合成エンジン。`koe-worklet.js` を同じオリジンから配信する必要がある。
+
+GitHub Pages にホストされているファイルをそのまま使える:
+
+```ts
+import { KoeEngine } from "@onjmin/koe";
+
+const engine = new KoeEngine({
+  workletUrl: "https://onjmin.github.io/koe/demo/koe-worklet.js",
+});
+
+// .koe ファイルをロード (Blob でも URL でも可)
+await engine.load("/voice.koe");
+
+// ノートシーケンスを再生
+await engine.play([
+  { phoneme: "a",  pitch: 440,  duration: 48000 }, // 1秒 / A4
+  { phoneme: "i",  pitch: 494,  duration: 24000 }, // 0.5秒 / B4
+  { phoneme: "u",  pitch: 392,  duration: 48000 }, // 1秒 / G4
+]);
+
+// 停止
+engine.stop();
+
+// ブラウザの autoplay ポリシーで停止した場合
+await engine.resume();
+```
+
+**NoteEvent の型:**
+
+```ts
+interface NoteEvent {
+  phoneme: string;  // 音素エイリアス (oto.ini の alias)
+  pitch: number;    // 出力ピッチ (Hz)
+  duration: number; // 出力長 (サンプル数 @ 48kHz)
+}
+```
+
+---
+
+### 3. VoiceBank — 音素 PCM の直接取得
+
+AudioContext 不要。WORLD ボコーダや独自の合成処理に PCM を渡したい場合に使う。
+
+```ts
+import { VoiceBank } from "@onjmin/koe";
+
+// URL からロード (HTTP Range リクエストを使用)
+const bank = await VoiceBank.load("/voice.koe");
+
+// Blob からロード
+const blob = await fetch("/voice.koe").then(r => r.blob());
+const bank2 = await VoiceBank.load(blob);
+
+// マニフェスト参照
+console.log(bank.manifest.phonemes); // 全音素のオフセット・ピッチ情報
+
+// Float64 PCM 取得 ([-1, 1] 正規化済み)
+const pcm = await bank.getPcm("a");
+
+// Int16 PCM バイト列を取得 (AudioWorklet への転送用)
+const buf = await bank.readPcmBytes("a");
+
+// 音素の存在確認
+bank.has("a"); // boolean
+```
+
+---
+
+### 4. Worldline — 高品質ノート合成 (WORLD ボコーダ)
+
+OpenUtau の worldline WASM で F0 分析・再合成を行う。`worldline.js` と `worldline.wasm` を配信する必要がある。
+
+GitHub Pages にホストされているファイルをそのまま使える:
+
+```ts
+import { VoiceBank, Worldline, leadInFromEntry } from "@onjmin/koe";
+
+const bank = await VoiceBank.load("/voice.koe");
+const wl   = await Worldline.load({
+  scriptUrl: "https://onjmin.github.io/koe/demo/world/worldline.js",
+});
+
+const alias = "a";
+const entry = bank.manifest.phonemes[alias];
+const pcm   = await bank.getPcm(alias);
+
+// ノートをレンダリング → Float32 PCM @ 48kHz
+const audio = wl.renderNote({
+  pcm,
+  pitch: 440,        // Hz
+  durationMs: 500,   // 母音部分の長さ (ms)
+  ...leadInFromEntry(entry), // preMs / consonantMs を entry から自動計算
+});
+
+if (audio) {
+  // audio: Float32Array (レイアウト: [子音 ≈ preMs][母音 ≈ durationMs])
+  const ctx = new AudioContext({ sampleRate: 48000 });
+  const buf = ctx.createBuffer(1, audio.length, 48000);
+  buf.copyToChannel(audio, 0);
+  const src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.connect(ctx.destination);
+  src.start();
+}
+```
+
+---
+
+### 5. .koe アーカイブ形式
+
+```
+[4B] magic 'KOE\0' (big-endian)
+[4B] JSON 長 (little-endian)
+[N ] manifest JSON (UTF-8)
+[M ] PCM バイナリ (Int16 / 48kHz / mono)
+```
+
+パース・生成ユーティリティ:
+
+```ts
+import { packKoe, parseKoeHeader, pcmBase } from "@onjmin/koe";
+
+// 生成
+const blob = packKoe(manifest, [pcmArrayBuffer]);
+
+// ヘッダ解析
+const headerBuf = await blob.slice(0, 8).arrayBuffer();
+const { jsonLength } = parseKoeHeader(headerBuf);
+const pcmOffset = pcmBase(jsonLength); // PCM データの開始バイト位置
+```
+
+---
+
+## API リファレンス
+
+| エクスポート | 説明 |
+|---|---|
+| `KoeEngine` | AudioWorklet ベースの連接合成エンジン |
+| `VoiceBank` | .koe から音素 PCM をオンデマンド取得 |
+| `Worldline` | WORLD ボコーダによる高品質ノート合成 |
+| `parseOto` | oto.ini テキストをパース |
+| `parseWav` | WAV バイナリをパース |
+| `toMono` | ステレオ → モノラル変換 |
+| `resample` | サンプルレート変換 |
+| `normalizePcm` | Float → Int16 正規化 |
+| `pack` | 音素リスト → manifest + PCM バイナリ |
+| `trimToOto` | WAV を oto リージョンにトリミング |
+| `packKoe` | manifest + PCM → .koe Blob |
+| `parseKoeHeader` | .koe ヘッダ解析 |
+| `pcmBase` | JSON長 → PCM 開始バイト位置 |
+| `detectF0` | PCM からピッチ自動検出 |
+| `noteNameToHz` | 音名 → Hz 変換 (例: `"A4"` → `440`) |
+| `pitchFromAliasSuffix` | エイリアス末尾の音名からピッチ推定 |
+| `parseFrqAverageF0` | .frq ファイルから平均 F0 取得 |
+| `frqFileName` | WAV ファイル名 → .frq ファイル名 |
+| `leadInFromEntry` | PhonemeEntry → preMs / consonantMs 変換 |
+| `samplesToMs` | サンプル数 → ミリ秒変換 |
+
+---
+
+## ライセンス
+
+MIT

package/dist/index.d.ts

@@ -155,6 +155,7 @@ type WorldlineFactory = (opts?: {
 }) => Promise<WorldlineWasm>;
 declare global {
     var WorldlineModule: WorldlineFactory | undefined;
+    var importScripts: ((...urls: string[]) => void) | undefined;
 }
 interface WorldlineLoadOptions {
     /**
@@ -217,7 +218,13 @@ declare class Worldline {
     private wasm;
     readonly sampleRate = 48000;
     private constructor();
-    /** Load + instantiate the worldline WASM module (deduped per scriptUrl). */
+    /**
+     * Load + instantiate the worldline WASM module (deduped per scriptUrl).
+     *
+     * Works on the main thread (loads via `<script>`) and inside a classic Web
+     * Worker (loads via `importScripts`), so the heavy synthesis can run
+     * off-thread. The matching `worldline.wasm` is fetched next to scriptUrl.
+     */
     static load(options: WorldlineLoadOptions): Promise<Worldline>;
     /**
      * Render one note to Float32 PCM at 48 kHz.
@@ -290,7 +297,7 @@ interface TrimmedPhoneme {
     /** PCM trimmed to the oto region [offset, cutoff] */
     pcm: Int16Array;
     /** Manifest params relative to the trimmed start (sample 0 = oto offset) */
-    entry: Omit<PhonemeEntry, 'offset'>;
+    entry: Omit<PhonemeEntry, "offset">;
 }
 /**
  * Cut the full WAV PCM down to its usable oto region and recompute parameters

package/dist/index.js

@@ -49,8 +49,11 @@ var RangeVoiceSource = class {
   }
 };
 async function rangeFetch(url, start, length) {
-  const res = await fetch(url, { headers: { Range: `bytes=${start}-${start + length - 1}` } });
-  if (!res.ok && res.status !== 206) throw new Error(`.koe fetch failed: ${res.status}`);
+  const res = await fetch(url, {
+    headers: { Range: `bytes=${start}-${start + length - 1}` }
+  });
+  if (!res.ok && res.status !== 206)
+    throw new Error(`.koe fetch failed: ${res.status}`);
   return res.arrayBuffer();
 }
 var VoiceBank = class _VoiceBank {
@@ -70,13 +73,19 @@ var VoiceBank = class _VoiceBank {
       const { jsonLength: jsonLength2 } = parseKoeHeader(header2);
       const json2 = await rangeFetch(koe, 8, jsonLength2);
       const manifest2 = JSON.parse(new TextDecoder().decode(json2));
-      return new _VoiceBank(manifest2, new RangeVoiceSource(koe, pcmBase(jsonLength2)));
+      return new _VoiceBank(
+        manifest2,
+        new RangeVoiceSource(koe, pcmBase(jsonLength2))
+      );
     }
     const header = await koe.slice(0, 8).arrayBuffer();
     const { jsonLength } = parseKoeHeader(header);
     const json = await koe.slice(8, 8 + jsonLength).arrayBuffer();
     const manifest = JSON.parse(new TextDecoder().decode(json));
-    return new _VoiceBank(manifest, new BlobVoiceSource(koe, pcmBase(jsonLength)));
+    return new _VoiceBank(
+      manifest,
+      new BlobVoiceSource(koe, pcmBase(jsonLength))
+    );
   }
   /** True if the bank contains a phoneme under this alias. */
   has(phoneme) {
@@ -158,7 +167,9 @@ var KoeEngine = class {
     if (!this.bank || !this.node) return Promise.resolve();
     const load = this.bank.readPcmBytes(name).then((buf) => {
       if (!buf) return;
-      this.node.port.postMessage({ type: "phoneme", name, buffer: buf }, [buf]);
+      this.node.port.postMessage({ type: "phoneme", name, buffer: buf }, [
+        buf
+      ]);
       this.delivered.add(name);
       this.pending.delete(name);
     });
@@ -198,12 +209,17 @@ var SYNTH_REQ_SIZE = 120;
 var WL_FRAME_MS = 10;
 var samplesToMs = (samples) => samples / WORLDLINE_SAMPLE_RATE * 1e3;
 function leadInFromEntry(entry) {
-  return { preMs: samplesToMs(entry.pre || 0), consonantMs: samplesToMs(entry.consonant || 0) };
+  return {
+    preMs: samplesToMs(entry.pre || 0),
+    consonantMs: samplesToMs(entry.consonant || 0)
+  };
 }
 var moduleCache = /* @__PURE__ */ new Map();
 function injectScript(src) {
   return new Promise((resolve, reject) => {
-    const existing = document.querySelector(`script[data-koe-worldline="${src}"]`);
+    const existing = document.querySelector(
+      `script[data-koe-worldline="${src}"]`
+    );
     if (existing) {
       resolve();
       return;
@@ -219,17 +235,30 @@ function injectScript(src) {
 function loadWasm(scriptUrl) {
   const cached = moduleCache.get(scriptUrl);
   if (cached) return cached;
-  if (typeof document === "undefined") {
-    return Promise.reject(
-      new Error("Worldline.load requires a DOM (browser) environment to load worldline.js")
-    );
-  }
   const baseUrl = scriptUrl.slice(0, scriptUrl.lastIndexOf("/") + 1);
-  const promise = injectScript(scriptUrl).then(() => {
+  const instantiate = () => {
     const factory = globalThis.WorldlineModule;
-    if (!factory) throw new Error("worldline: WorldlineModule global was not defined by the script");
+    if (!factory)
+      throw new Error(
+        "worldline: WorldlineModule global was not defined by the script"
+      );
     return factory({ locateFile: (f) => baseUrl + f });
-  });
+  };
+  let promise;
+  if (typeof document !== "undefined") {
+    promise = injectScript(scriptUrl).then(instantiate);
+  } else if (typeof globalThis.importScripts === "function") {
+    promise = Promise.resolve().then(() => {
+      globalThis.importScripts(scriptUrl);
+      return instantiate();
+    });
+  } else {
+    return Promise.reject(
+      new Error(
+        "Worldline.load requires a DOM or a classic Web Worker (importScripts) to load worldline.js"
+      )
+    );
+  }
   moduleCache.set(scriptUrl, promise);
   return promise;
 }
@@ -239,7 +268,13 @@ var Worldline = class _Worldline {
   }
   wasm;
   sampleRate = WORLDLINE_SAMPLE_RATE;
-  /** Load + instantiate the worldline WASM module (deduped per scriptUrl). */
+  /**
+   * Load + instantiate the worldline WASM module (deduped per scriptUrl).
+   *
+   * Works on the main thread (loads via `<script>`) and inside a classic Web
+   * Worker (loads via `importScripts`), so the heavy synthesis can run
+   * off-thread. The matching `worldline.wasm` is fetched next to scriptUrl.
+   */
   static async load(options) {
     return new _Worldline(await loadWasm(options.scriptUrl));
   }
@@ -331,7 +366,16 @@ var Worldline = class _Worldline {
     WL.HEAPF64.set(tArr, tPtr >> 3);
     WL.HEAPF64.set(bArr, bPtr >> 3);
     WL.HEAPF64.set(vArr, vPtr >> 3);
-    WL._PhraseSynthSetCurves(ps, f0Ptr, gPtr, tPtr, bPtr, vPtr, nFrames, WL_FRAME_MS);
+    WL._PhraseSynthSetCurves(
+      ps,
+      f0Ptr,
+      gPtr,
+      tPtr,
+      bPtr,
+      vPtr,
+      nFrames,
+      WL_FRAME_MS
+    );
     WL._free(f0Ptr);
     WL._free(gPtr);
     WL._free(tPtr);
@@ -435,7 +479,8 @@ function toMono(wav) {
   const out = new Float32Array(len);
   for (let i = 0; i < len; i++) {
     let sum = 0;
-    for (let c = 0; c < wav.channels; c++) sum += wav.samples[i * wav.channels + c];
+    for (let c = 0; c < wav.channels; c++)
+      sum += wav.samples[i * wav.channels + c];
     out[i] = sum / wav.channels;
   }
   return { sampleRate: wav.sampleRate, channels: 1, samples: out };
@@ -478,7 +523,15 @@ function readFourCC(view, pos) {
 
 // src/converter/pitch.ts
 var SAMPLE_RATE = 48e3;
-var NAME_SEMITONE = { c: 0, d: 2, e: 4, f: 5, g: 7, a: 9, b: 11 };
+var NAME_SEMITONE = {
+  c: 0,
+  d: 2,
+  e: 4,
+  f: 5,
+  g: 7,
+  a: 9,
+  b: 11
+};
 function noteNameToHz(name) {
   const m = /^([A-Ga-g])([#b]?)(-?\d+)$/.exec(name);
   if (!m) return null;
@@ -556,7 +609,11 @@ function trimToOto(pcm, oto, recordedPitch = 0) {
   const pre = clamp(msToSamples(oto.pre), 0, length);
   const overlap = clamp(msToSamples(oto.overlap), 0, length);
   const consonant = clamp(msToSamples(oto.consonant), 0, length);
-  const pitch = recordedPitch > 0 ? recordedPitch : detectF0(slice, Math.min(Math.max(pre, consonant), Math.max(0, length - 1)), length);
+  const pitch = recordedPitch > 0 ? recordedPitch : detectF0(
+    slice,
+    Math.min(Math.max(pre, consonant), Math.max(0, length - 1)),
+    length
+  );
   return {
     pcm: slice,
     entry: { length, pre, overlap, consonant, pitch }
@@ -577,7 +634,10 @@ function pack(inputs, referencePitch = 220) {
   const view = new Uint8Array(bin);
   let pos = 0;
   for (const chunk of chunks) {
-    view.set(new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength), pos);
+    view.set(
+      new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength),
+      pos
+    );
     pos += chunk.byteLength;
   }
   const manifest = {

package/dist/index.js.map

@@ -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  const json = new TextEncoder().encode(JSON.stringify(manifest));\n  const header = new ArrayBuffer(8);\n  const view = new DataView(header);\n  view.setUint32(0, MAGIC, false); // magic, big-endian\n  view.setUint32(4, json.byteLength, true); // json length, little-endian\n  return new Blob([header, json, ...pcmParts]);\n}\n\n/** Read the 8-byte header → JSON length. Throws on bad magic. */\nexport function parseKoeHeader(headerBytes: ArrayBuffer): { jsonLength: number } {\n  const view = new DataView(headerBytes);\n  if (view.byteLength < 8 || view.getUint32(0, false) !== MAGIC) {\n    throw new Error('Not a .koe file (bad magic)');\n  }\n  return { 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  readBytes(offset: number, length: number): Promise<ArrayBuffer>;\n}\n\nclass BlobVoiceSource implements VoiceSource {\n  constructor(\n    private blob: Blob,\n    private base: number,\n  ) {}\n  readBytes(offset: number, length: number): Promise<ArrayBuffer> {\n    const start = this.base + offset;\n    return this.blob.slice(start, start + length).arrayBuffer();\n  }\n}\n\nclass RangeVoiceSource implements VoiceSource {\n  constructor(\n    private url: string,\n    private base: number,\n  ) {}\n  async readBytes(offset: number, length: number): Promise<ArrayBuffer> {\n    const start = this.base + offset;\n    const res = await fetch(this.url, {\n      headers: { Range: `bytes=${start}-${start + length - 1}` },\n    });\n    if (!res.ok && res.status !== 206) {\n      throw new Error(`.koe range request failed: ${res.status}`);\n    }\n    return res.arrayBuffer();\n  }\n}\n\nasync function rangeFetch(url: string, start: number, length: number): Promise<ArrayBuffer> {\n  const res = await fetch(url, { headers: { Range: `bytes=${start}-${start + length - 1}` } });\n  if (!res.ok && res.status !== 206) throw new Error(`.koe fetch failed: ${res.status}`);\n  return 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  private constructor(\n    /** The voice bank manifest (sample rate, reference pitch, phoneme table). */\n    readonly manifest: Manifest,\n    private source: VoiceSource,\n  ) {}\n\n  /**\n   * Parse a .koe archive header + manifest and bind a lazy PCM source.\n   * @param koe a Blob/File of the .koe archive, or a URL (served with Range support)\n   */\n  static async load(koe: Blob | string): Promise<VoiceBank> {\n    if (typeof koe === 'string') {\n      const header = await rangeFetch(koe, 0, 8);\n      const { jsonLength } = parseKoeHeader(header);\n      const json = await rangeFetch(koe, 8, jsonLength);\n      const manifest = JSON.parse(new TextDecoder().decode(json)) as Manifest;\n      return new VoiceBank(manifest, new RangeVoiceSource(koe, pcmBase(jsonLength)));\n    }\n    const header = await koe.slice(0, 8).arrayBuffer();\n    const { jsonLength } = parseKoeHeader(header);\n    const json = await koe.slice(8, 8 + jsonLength).arrayBuffer();\n    const manifest = JSON.parse(new TextDecoder().decode(json)) as Manifest;\n    return new VoiceBank(manifest, new BlobVoiceSource(koe, pcmBase(jsonLength)));\n  }\n\n  /** True if the bank contains a phoneme under this alias. */\n  has(phoneme: string): boolean {\n    return this.manifest.phonemes[phoneme] !== undefined;\n  }\n\n  /**\n   * Raw Int16 PCM bytes (48 kHz / mono) for a phoneme, or null if unknown.\n   * The returned ArrayBuffer is freshly allocated and safe to transfer to a\n   * worker / AudioWorklet.\n   */\n  async readPcmBytes(phoneme: string): Promise<ArrayBuffer | null> {\n    const entry = this.manifest.phonemes[phoneme];\n    if (!entry) return null;\n    return this.source.readBytes(entry.offset, entry.length * 2); // Int16 = 2 bytes/sample\n  }\n\n  /**\n   * A phoneme's PCM as a Float64Array normalised to [-1, 1], or null if unknown.\n   * Intended for external analysis / resynthesis such as the WORLD vocoder.\n   */\n  async getPcm(phoneme: string): Promise<Float64Array | null> {\n    const buf = await this.readPcmBytes(phoneme);\n    if (!buf) return null;\n    const int16 = new Int16Array(buf);\n    const f64 = new Float64Array(int16.length);\n    for (let i = 0; i < int16.length; i++) f64[i] = int16[i] / 32768;\n    return f64;\n  }\n}\n","import type { Manifest, NoteEvent } from '../types.js';\nimport { VoiceBank } from './voice-bank.js';\n\nexport type { NoteEvent };\n\nexport interface KoeEngineOptions {\n  /** URL to koe-worklet.js. Defaults to './koe-worklet.js'. */\n  workletUrl?: 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  private ctx: AudioContext;\n  private workletUrl: string;\n  private node: AudioWorkletNode | null = null;\n  private bank: VoiceBank | null = null;\n  private delivered = new Set<string>();\n  private pending = new Map<string, Promise<void>>();\n\n  constructor(options: KoeEngineOptions = {}) {\n    this.ctx = new AudioContext({ sampleRate: 48000 });\n    this.workletUrl = options.workletUrl ?? './koe-worklet.js';\n  }\n\n  get audioContext(): AudioContext {\n    return this.ctx;\n  }\n\n  get manifest(): Manifest | null {\n    return this.bank?.manifest ?? null;\n  }\n\n  /** The underlying voice bank (manifest + on-demand PCM), or null before load(). */\n  get voiceBank(): VoiceBank | null {\n    return this.bank;\n  }\n\n  /**\n   * Register the worklet and bind a .koe voice bank.\n   * @param koe a Blob/File of the .koe archive, or a URL (served with Range support)\n   */\n  async load(koe: Blob | string): Promise<void> {\n    await this.ctx.audioWorklet.addModule(this.workletUrl);\n\n    this.bank = await VoiceBank.load(koe);\n    this.delivered.clear();\n    this.pending.clear();\n\n    this.node = new AudioWorkletNode(this.ctx, 'koe-processor', {\n      numberOfInputs: 0,\n      numberOfOutputs: 1,\n      outputChannelCount: [1],\n    });\n    this.node.port.postMessage({ type: 'init', manifest: this.bank.manifest });\n    this.node.connect(this.ctx.destination);\n\n    console.log(\n      '[koe] ready —',\n      Object.keys(this.bank.manifest.phonemes).length,\n      'phonemes (on-demand)',\n    );\n  }\n\n  /** Fetch one phoneme's PCM and deliver it to the worklet (deduped, cached). */\n  private ensurePhoneme(name: string): Promise<void> {\n    if (this.delivered.has(name)) return Promise.resolve();\n    const existing = this.pending.get(name);\n    if (existing) return existing;\n    if (!this.bank || !this.node) return Promise.resolve();\n\n    const load = this.bank.readPcmBytes(name).then(buf => {\n      if (!buf) return;\n      this.node!.port.postMessage({ type: 'phoneme', name, buffer: buf }, [buf]);\n      this.delivered.add(name);\n      this.pending.delete(name);\n    });\n\n    this.pending.set(name, load);\n    return load;\n  }\n\n  /** Stop current playback, preload the phonemes for `notes`, then queue them. */\n  async play(notes: NoteEvent[]): Promise<void> {\n    if (!this.node) throw new Error('KoeEngine: call load() before play()');\n    this.node.port.postMessage({ type: 'stop' });\n    const names = [...new Set(notes.map(n => n.phoneme))].filter(Boolean);\n    await Promise.all(names.map(n => this.ensurePhoneme(n)));\n    this.node.port.postMessage({ type: 'play', notes });\n  }\n\n  /** Stop playback and clear the queue. */\n  stop(): void {\n    this.node?.port.postMessage({ type: 'stop' });\n  }\n\n  /** Resume the AudioContext if suspended (e.g. after autoplay block). */\n  async resume(): Promise<void> {\n    if (this.ctx.state === 'suspended') await this.ctx.resume();\n  }\n\n  /**\n   * Read a phoneme's raw PCM and return it as a Float64Array normalised to\n   * [-1, 1]. Convenience that forwards to the underlying {@link VoiceBank}.\n   * Intended for external analysis such as the WORLD vocoder.\n   */\n  async getPcm(phoneme: string): Promise<Float64Array | null> {\n    return this.bank?.getPcm(phoneme) ?? null;\n  }\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  _PhraseSynthNew(): number;\n  _PhraseSynthDelete(ps: number): void;\n  _PhraseSynthAddRequest(\n    ps: number,\n    req: number,\n    posMs: number,\n    skipMs: number,\n    lengthMs: number,\n    fadeInMs: number,\n    fadeOutMs: number,\n    flag: number,\n  ): void;\n  _PhraseSynthSetCurves(\n    ps: number,\n    f0: number,\n    gender: number,\n    tension: number,\n    breathiness: number,\n    voicing: number,\n    length: number,\n    frameMs: number,\n  ): void;\n  _PhraseSynthSynth(ps: number, yPtrPtr: number, flag: number): number;\n  _malloc(size: number): number;\n  _free(ptr: number): void;\n  setValue(ptr: number, value: number, type: string): void;\n  getValue(ptr: number, type: string): number;\n  HEAPF32: Float32Array;\n  HEAPF64: Float64Array;\n}\n\ntype WorldlineFactory = (opts?: {\n  locateFile?: (path: string) => string;\n}) => Promise<WorldlineWasm>;\n\ndeclare global {\n  // Injected by worldline.js when loaded as a classic <script>.\n  // eslint-disable-next-line no-var\n  var WorldlineModule: WorldlineFactory | undefined;\n}\n\nexport interface WorldlineLoadOptions {\n  /**\n   * URL of `worldline.js` (the Emscripten loader). The matching\n   * `worldline.wasm` must sit next to it — it is resolved relative to this URL.\n   *\n   * When consuming koe from npm + a CDN this is typically e.g.\n   * `https://cdn.jsdelivr.net/npm/@onjmin/koe/dist/world/worldline.js`, or your\n   * own hosted copy of `dist/world/`.\n   */\n  scriptUrl: string;\n}\n\nexport interface RenderNoteParams {\n  /**\n   * Source phoneme PCM normalised to [-1, 1] (e.g. from\n   * `VoiceBank.getPcm()` / `KoeEngine.getPcm()`).\n   */\n  pcm: Float64Array;\n  /** Target output pitch in Hz. */\n  pitch: number;\n  /** Sustain / vowel duration in ms (the lead-in below is rendered on top). */\n  durationMs: number;\n  /** Preutterance / lead-in in ms — convert from {@link PhonemeEntry.pre}. */\n  preMs: number;\n  /** Consonant length in ms — convert from {@link PhonemeEntry.consonant}. */\n  consonantMs: number;\n  /** Reference tempo in BPM for worldline's internal timing. Default 120. */\n  tempo?: number;\n}\n\n/** Convert a sample count at 48 kHz to milliseconds. */\nexport const samplesToMs = (samples: number): number => (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): { preMs: number; consonantMs: number } {\n  return { preMs: samplesToMs(entry.pre || 0), consonantMs: samplesToMs(entry.consonant || 0) };\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  return new Promise((resolve, reject) => {\n    const existing = document.querySelector<HTMLScriptElement>(`script[data-koe-worldline=\"${src}\"]`);\n    if (existing) {\n      resolve();\n      return;\n    }\n    const s = document.createElement('script');\n    s.src = src;\n    s.dataset.koeWorldline = src;\n    s.onload = () => resolve();\n    s.onerror = () => reject(new Error(`worldline: failed to load ${src}`));\n    document.head.appendChild(s);\n  });\n}\n\nfunction loadWasm(scriptUrl: string): Promise<WorldlineWasm> {\n  const cached = moduleCache.get(scriptUrl);\n  if (cached) return cached;\n\n  if (typeof document === 'undefined') {\n    return Promise.reject(\n      new Error('Worldline.load requires a DOM (browser) environment to load worldline.js'),\n    );\n  }\n\n  const baseUrl = scriptUrl.slice(0, scriptUrl.lastIndexOf('/') + 1);\n  const promise = injectScript(scriptUrl).then(() => {\n    const factory = globalThis.WorldlineModule;\n    if (!factory) throw new Error('worldline: WorldlineModule global was not defined by the script');\n    return factory({ locateFile: (f: string) => baseUrl + f });\n  });\n  moduleCache.set(scriptUrl, promise);\n  return 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  readonly sampleRate = WORLDLINE_SAMPLE_RATE;\n\n  private constructor(private wasm: WorldlineWasm) {}\n\n  /** Load + instantiate the worldline WASM module (deduped per scriptUrl). */\n  static async load(options: WorldlineLoadOptions): Promise<Worldline> {\n    return new Worldline(await loadWasm(options.scriptUrl));\n  }\n\n  /**\n   * Render one note to Float32 PCM at 48 kHz.\n   *\n   * The output buffer is laid out as [lead-in/consonant ≈ preMs][vowel ≈\n   * durationMs], rendered from sample offset 0 (no leading silence). The vowel\n   * onset (the \"beat\") sits at ≈ preMs into the buffer, so a sequencer should\n   * place the buffer at `beatTime − preMs` and may trim/crossfade the lead-in.\n   *\n   * No internal crossfade is applied — apply fades externally.\n   *\n   * @returns Float32 PCM, or null when `pcm` is shorter than\n   *          {@link MIN_WORLDLINE_SAMPLES} (too short for stable F0 analysis).\n   */\n  renderNote(params: RenderNoteParams): Float32Array | null {\n    const { pcm, pitch, durationMs, preMs, consonantMs, tempo = 120 } = params;\n    if (!pcm || pcm.length < MIN_WORLDLINE_SAMPLES) return null;\n\n    const WL = this.wasm;\n    const FS = WORLDLINE_SAMPLE_RATE;\n    const midiNote = Math.round(69 + 12 * Math.log2(pitch / 440));\n    const posMs = 0; // no leading silence\n    const reqLen = preMs + durationMs; // render lead-in/consonant + vowel\n\n    // cut_off must trim at least ~2 frames off the tail. PhraseSynth::AddRequest\n    // computes the trim frame count as ceil(in_length_ms)/frame_ms, but\n    // PyinEstimator yields floor(N/nhop)−1 frames — one fewer. With cut_off=0 the\n    // requested frame count exceeds f0_.size() and Model::Trim's f0_.erase() runs\n    // past end(), causing \"memory access out of bounds\". Trimming 2 frames keeps\n    // it within bounds.\n    const cutMs = WL_FRAME_MS * 2;\n\n    const ps = WL._PhraseSynthNew();\n    if (!ps) return null;\n    const reqPtr = WL._malloc(SYNTH_REQ_SIZE);\n    if (!reqPtr) {\n      WL._PhraseSynthDelete(ps);\n      return null;\n    }\n    const samplePtr = WL._malloc(pcm.length * 8);\n    if (!samplePtr) {\n      WL._free(reqPtr);\n      WL._PhraseSynthDelete(ps);\n      return null;\n    }\n    WL.HEAPF64.set(pcm, samplePtr >> 3);\n\n    const sv = (off: number, val: number, type: string) => WL.setValue(reqPtr + off, val, type);\n    sv(0, FS, 'i32');\n    sv(4, pcm.length, 'i32');\n    sv(8, samplePtr, '*');\n    sv(12, 0, 'i32');\n    sv(16, 0, '*');\n    sv(20, midiNote, 'i32');\n    sv(24, 100.0, 'double');\n    sv(32, 0.0, 'double');\n    sv(40, reqLen, 'double');\n    sv(48, consonantMs, 'double');\n    sv(56, cutMs, 'double');\n    sv(64, 100.0, 'double');\n    sv(72, 0.0, 'double');\n    sv(80, tempo, 'double');\n    sv(88, 0, 'i32');\n    sv(92, 0, '*');\n    sv(96, 0, 'i32');\n    sv(100, 0, 'i32');\n    sv(104, 100, 'i32');\n    sv(108, 0, 'i32');\n    sv(112, 0, 'i32');\n    sv(116, 100, 'i32');\n\n    // skip/fadeIn/fadeOut all 0 — crossfade is handled externally.\n    WL._PhraseSynthAddRequest(ps, reqPtr, posMs, 0.0, reqLen, 0.0, 0.0, 0);\n    WL._free(samplePtr);\n    WL._free(reqPtr);\n\n    const totalMs = posMs + reqLen + WL_FRAME_MS * 2;\n    const nFrames = Math.ceil(totalMs / WL_FRAME_MS) + 4;\n    const f0Arr = new Float64Array(nFrames).fill(pitch);\n    const gArr = new Float64Array(nFrames).fill(0.5);\n    const tArr = new Float64Array(nFrames).fill(0.5);\n    const bArr = new Float64Array(nFrames).fill(0.5);\n    const vArr = new Float64Array(nFrames).fill(1.0);\n\n    const f0Ptr = WL._malloc(nFrames * 8);\n    const gPtr = WL._malloc(nFrames * 8);\n    const tPtr = WL._malloc(nFrames * 8);\n    const bPtr = WL._malloc(nFrames * 8);\n    const vPtr = WL._malloc(nFrames * 8);\n    if (!f0Ptr || !gPtr || !tPtr || !bPtr || !vPtr) {\n      if (f0Ptr) WL._free(f0Ptr);\n      if (gPtr) WL._free(gPtr);\n      if (tPtr) WL._free(tPtr);\n      if (bPtr) WL._free(bPtr);\n      if (vPtr) WL._free(vPtr);\n      WL._PhraseSynthDelete(ps);\n      return null;\n    }\n    WL.HEAPF64.set(f0Arr, f0Ptr >> 3);\n    WL.HEAPF64.set(gArr, gPtr >> 3);\n    WL.HEAPF64.set(tArr, tPtr >> 3);\n    WL.HEAPF64.set(bArr, bPtr >> 3);\n    WL.HEAPF64.set(vArr, vPtr >> 3);\n    WL._PhraseSynthSetCurves(ps, f0Ptr, gPtr, tPtr, bPtr, vPtr, nFrames, WL_FRAME_MS);\n    WL._free(f0Ptr);\n    WL._free(gPtr);\n    WL._free(tPtr);\n    WL._free(bPtr);\n    WL._free(vPtr);\n\n    const yPtrPtr = WL._malloc(4);\n    if (!yPtrPtr) {\n      WL._PhraseSynthDelete(ps);\n      return null;\n    }\n    const outLen = WL._PhraseSynthSynth(ps, yPtrPtr, 0);\n    const yPtr = WL.getValue(yPtrPtr, '*');\n    const audio =\n      outLen > 0 ? new Float32Array(WL.HEAPF32.buffer, yPtr, outLen).slice() : null;\n    WL._free(yPtrPtr);\n    WL._PhraseSynthDelete(ps);\n    return audio;\n  }\n}\n","export interface OtoEntry {\n  /** Source WAV filename */\n  wav: string;\n  /** Phoneme alias */\n  alias: string;\n  /** Left blank — offset from WAV start (ms) */\n  offset: number;\n  /** Consonant portion end from offset (ms) */\n  consonant: number;\n  /** Right blank — negative = from WAV end, positive = from offset (ms) */\n  cutoff: number;\n  /** Preutterance from offset (ms) */\n  pre: number;\n  /** Overlap / crossfade region (ms) */\n  overlap: 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  const entries: OtoEntry[] = [];\n\n  for (const raw of content.split(/\\r?\\n/)) {\n    const line = raw.trim();\n    if (!line || line.startsWith('#')) continue;\n\n    const eq = line.indexOf('=');\n    if (eq === -1) continue;\n\n    const wav = line.slice(0, eq).trim();\n    const parts = line.slice(eq + 1).split(',');\n    if (parts.length < 6) continue;\n\n    const [alias, offsetStr, consonantStr, cutoffStr, preStr, overlapStr] = parts;\n    // Empty alias → use filename without extension (UTAU spec default)\n    const aliasStr = alias.trim() || wav.replace(/\\.[^.]+$/, '');\n    const entry: OtoEntry = {\n      wav,\n      alias: aliasStr,\n      offset: parseFloat(offsetStr) || 0,\n      consonant: parseFloat(consonantStr) || 0,\n      cutoff: parseFloat(cutoffStr) || 0,\n      pre: parseFloat(preStr) || 0,\n      overlap: parseFloat(overlapStr) || 0,\n    };\n\n    if (!entry.alias) continue;\n    entries.push(entry);\n  }\n\n  return entries;\n}\n","export interface WavData {\n  sampleRate: number;\n  channels: number;\n  /** Normalized samples in [-1, 1], interleaved if multi-channel */\n  samples: 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  const view = new DataView(buf);\n\n  const riff = readFourCC(view, 0);\n  if (riff !== 'RIFF') throw new Error(`Not a RIFF file (got \"${riff}\")`);\n\n  let sampleRate = 0;\n  let channels = 0;\n  let bitsPerSample = 0;\n  let audioFormat = 1;\n  let dataOffset = 0;\n  let dataLength = 0;\n\n  let pos = 12;\n  while (pos < view.byteLength - 8) {\n    const id = readFourCC(view, pos);\n    const size = view.getUint32(pos + 4, true);\n    pos += 8;\n\n    if (id === 'fmt ') {\n      audioFormat = view.getUint16(pos, true);\n      channels = view.getUint16(pos + 2, true);\n      sampleRate = view.getUint32(pos + 4, true);\n      bitsPerSample = view.getUint16(pos + 14, true);\n    } else if (id === 'data') {\n      dataOffset = pos;\n      dataLength = size;\n      break;\n    }\n\n    pos += size + (size & 1);\n  }\n\n  if (!dataOffset) throw new Error('WAV has no data chunk');\n  if (!channels || !sampleRate) throw new Error('WAV fmt chunk missing');\n\n  const bytesPerSample = bitsPerSample >> 3;\n  const totalSamples = Math.floor(dataLength / bytesPerSample);\n  const samples = new Float32Array(totalSamples);\n\n  for (let i = 0; i < totalSamples; i++) {\n    const p = dataOffset + i * bytesPerSample;\n    if (audioFormat === 3) {\n      samples[i] = view.getFloat32(p, true);\n    } else if (bitsPerSample === 8) {\n      samples[i] = (view.getUint8(p) - 128) / 128;\n    } else if (bitsPerSample === 16) {\n      samples[i] = view.getInt16(p, true) / 32768;\n    } else if (bitsPerSample === 24) {\n      const lo = view.getUint8(p) | (view.getUint8(p + 1) << 8);\n      let hi = view.getUint8(p + 2);\n      if (hi & 0x80) hi = hi | 0xffffff00;\n      samples[i] = ((hi << 16) | lo) / 8388608;\n    }\n  }\n\n  return { sampleRate, channels, samples };\n}\n\n/** Mix down to mono by averaging all channels. */\nexport function toMono(wav: WavData): WavData {\n  if (wav.channels === 1) return wav;\n  const len = wav.samples.length / wav.channels;\n  const out = new Float32Array(len);\n  for (let i = 0; i < len; i++) {\n    let sum = 0;\n    for (let c = 0; c < wav.channels; c++) sum += wav.samples[i * wav.channels + c];\n    out[i] = sum / wav.channels;\n  }\n  return { 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  if (wav.sampleRate === targetRate) return wav;\n  const ratio = wav.sampleRate / targetRate;\n  const outLen = Math.floor(wav.samples.length / ratio);\n  const out = new Float32Array(outLen);\n  const src = wav.samples;\n  for (let i = 0; i < outLen; i++) {\n    const x = i * ratio;\n    const xi = Math.floor(x);\n    const frac = x - xi;\n    out[i] = (src[xi] ?? 0) + ((src[xi + 1] ?? 0) - (src[xi] ?? 0)) * frac;\n  }\n  return { sampleRate: targetRate, channels: 1, samples: out };\n}\n\n/** Convert Float32 [-1,1] samples to Int16 PCM. */\nexport function toInt16(samples: Float32Array): Int16Array {\n  const out = new Int16Array(samples.length);\n  for (let i = 0; i < samples.length; i++) {\n    out[i] = Math.round(Math.max(-1, Math.min(1, samples[i])) * 32767);\n  }\n  return out;\n}\n\n/** Normalize then convert a WAV to 48kHz/16bit/mono Int16 PCM. */\nexport function normalizePcm(buf: ArrayBuffer): Int16Array {\n  const wav = parseWav(buf);\n  const mono = toMono(wav);\n  const resampled = resample(mono, 48000);\n  return toInt16(resampled.samples);\n}\n\nfunction readFourCC(view: DataView, pos: number): string {\n  return String.fromCharCode(\n    view.getUint8(pos),\n    view.getUint8(pos + 1),\n    view.getUint8(pos + 2),\n    view.getUint8(pos + 3),\n  );\n}\n","const SAMPLE_RATE = 48000;\n\nconst NAME_SEMITONE: Record<string, number> = { c: 0, d: 2, e: 4, f: 5, g: 7, a: 9, b: 11 };\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  const m = /^([A-Ga-g])([#b]?)(-?\\d+)$/.exec(name);\n  if (!m) return null;\n  let semi = NAME_SEMITONE[m[1].toLowerCase()];\n  if (m[2] === '#') semi++;\n  else if (m[2] === 'b') semi--;\n  const midi = (parseInt(m[3], 10) + 1) * 12 + semi;\n  return 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  const m = /_([A-Ga-g][#b]?-?\\d+)$/.exec(alias);\n  return 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  const DECIM = 4;\n  const sr = SAMPLE_RATE / DECIM; // 12 kHz analysis rate\n  const minLag = Math.floor(sr / 700); // highest f0 we look for\n  const maxLag = Math.floor(sr / 70); // lowest f0 we look for\n\n  const outLen = Math.floor((end - start) / DECIM);\n  if (outLen < maxLag + 2) return 0;\n  const win = Math.min(outLen, 1500); // up to ~0.125 s of vowel\n\n  // Decimate (box-average) + remove DC, and build a prefix sum of squares so\n  // each lag's window energy is O(1).\n  const buf = new Float32Array(win);\n  let mean = 0;\n  for (let i = 0; i < win; i++) {\n    let s = 0;\n    const base = start + i * DECIM;\n    for (let j = 0; j < DECIM; j++) s += pcm[base + j];\n    buf[i] = s;\n    mean += s;\n  }\n  mean /= win;\n\n  const sq = new Float64Array(win + 1);\n  for (let i = 0; i < win; i++) {\n    buf[i] -= mean;\n    sq[i + 1] = sq[i] + buf[i] * buf[i];\n  }\n  if (sq[win] < 1) return 0; // effectively silent\n\n  // Normalized autocorrelation: 2·r(lag) / (E[0..n] + E[lag..lag+n]) ∈ [-1, 1].\n  const norm = (lag: number): number => {\n    const n = win - lag;\n    let r = 0;\n    for (let i = 0; i < n; i++) r += buf[i] * buf[i + lag];\n    const e = sq[n] + (sq[lag + n] - sq[lag]);\n    return e > 0 ? (2 * r) / e : 0;\n  };\n\n  let bestLag = -1;\n  let best = 0;\n  for (let lag = minLag; lag <= maxLag; lag++) {\n    const v = norm(lag);\n    if (v > best) {\n      best = v;\n      bestLag = lag;\n    }\n  }\n  if (bestLag < 1 || best < 0.4) return 0; // unvoiced / no clear period\n\n  // Parabolic interpolation around the peak\n  const y0 = norm(bestLag - 1);\n  const y1 = best;\n  const y2 = norm(bestLag + 1);\n  const denom = y0 - 2 * y1 + y2;\n  const shift = denom !== 0 ? (0.5 * (y0 - y2)) / denom : 0;\n\n  return 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  return Math.round((ms / 1000) * TARGET_RATE);\n}\n\nconst clamp = (v: number, lo: number, hi: number) => Math.max(lo, Math.min(hi, v));\n\nexport interface PackInput {\n  oto: OtoEntry;\n  /** Full normalized PCM of the source WAV (48kHz / 16bit / mono) */\n  pcm: Int16Array;\n  /** Known recorded pitch in Hz (e.g. from the .frq file). 0/undefined → auto-detect. */\n  recordedPitch?: number;\n}\n\nexport interface PackOutput {\n  manifest: Manifest;\n  /** Raw PCM blob — Int16 / 48kHz / mono */\n  bin: ArrayBuffer;\n}\n\nexport interface TrimmedPhoneme {\n  /** PCM trimmed to the oto region [offset, cutoff] */\n  pcm: Int16Array;\n  /** Manifest params relative to the trimmed start (sample 0 = oto offset) */\n  entry: 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(pcm: Int16Array, oto: OtoEntry, recordedPitch = 0): TrimmedPhoneme {\n  const full = pcm.length;\n\n  const start = clamp(msToSamples(oto.offset), 0, full);\n  const end = oto.cutoff < 0\n    ? clamp(start + msToSamples(-oto.cutoff), start, full)\n    : clamp(full - msToSamples(oto.cutoff), start, full);\n\n  const slice = pcm.subarray(start, end);\n  const length = slice.length;\n\n  const pre = clamp(msToSamples(oto.pre), 0, length);\n  const overlap = clamp(msToSamples(oto.overlap), 0, length);\n  const consonant = clamp(msToSamples(oto.consonant), 0, length);\n\n  // Prefer the known recorded pitch (from the .frq file or alias suffix); only\n  // fall back to autocorrelation on the sustained vowel when none is available.\n  const pitch = recordedPitch > 0\n    ? recordedPitch\n    : detectF0(slice, Math.min(Math.max(pre, consonant), Math.max(0, length - 1)), length);\n\n  return {\n    pcm: slice,\n    entry: { length, pre, overlap, consonant, pitch },\n  };\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  const phonemes: Record<string, PhonemeEntry> = {};\n  const chunks: Int16Array[] = [];\n  let byteOffset = 0;\n\n  for (const { oto, pcm, recordedPitch } of inputs) {\n    const { pcm: slice, entry } = trimToOto(pcm, oto, recordedPitch);\n    if (slice.length === 0) continue;\n\n    phonemes[oto.alias] = { offset: byteOffset, ...entry };\n    byteOffset += slice.byteLength;\n    chunks.push(slice);\n  }\n\n  const bin = new ArrayBuffer(byteOffset);\n  const view = new Uint8Array(bin);\n  let pos = 0;\n  for (const chunk of chunks) {\n    view.set(new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength), pos);\n    pos += chunk.byteLength;\n  }\n\n  const manifest: Manifest = {\n    sampleRate: 48000,\n    referencePitch,\n    phonemes,\n  };\n\n  return { 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  if (buffer.byteLength < 20) return null;\n  const view = new DataView(buffer);\n\n  let header = '';\n  for (let i = 0; i < 8; i++) header += String.fromCharCode(view.getUint8(i));\n  if (header !== 'FREQ0003') return null;\n\n  const avg = view.getFloat64(12, true); // little-endian\n  return 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  const dot = wavName.lastIndexOf('.');\n  const base = dot >= 0 ? wavName.slice(0, dot) : wavName;\n  const ext = dot >= 0 ? wavName.slice(dot + 1) : 'wav';\n  return 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+{"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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package/dist/koe-convert.js

@@ -89,7 +89,8 @@ function toMono(wav) {
   const out = new Float32Array(len);
   for (let i = 0; i < len; i++) {
     let sum = 0;
-    for (let c = 0; c < wav.channels; c++) sum += wav.samples[i * wav.channels + c];
+    for (let c = 0; c < wav.channels; c++)
+      sum += wav.samples[i * wav.channels + c];
     out[i] = sum / wav.channels;
   }
   return { sampleRate: wav.sampleRate, channels: 1, samples: out };
@@ -132,7 +133,15 @@ function readFourCC(view, pos) {
 
 // src/converter/pitch.ts
 var SAMPLE_RATE = 48e3;
-var NAME_SEMITONE = { c: 0, d: 2, e: 4, f: 5, g: 7, a: 9, b: 11 };
+var NAME_SEMITONE = {
+  c: 0,
+  d: 2,
+  e: 4,
+  f: 5,
+  g: 7,
+  a: 9,
+  b: 11
+};
 function noteNameToHz(name) {
   const m = /^([A-Ga-g])([#b]?)(-?\d+)$/.exec(name);
   if (!m) return null;
@@ -210,7 +219,11 @@ function trimToOto(pcm, oto, recordedPitch = 0) {
   const pre = clamp(msToSamples(oto.pre), 0, length);
   const overlap = clamp(msToSamples(oto.overlap), 0, length);
   const consonant = clamp(msToSamples(oto.consonant), 0, length);
-  const pitch = recordedPitch > 0 ? recordedPitch : detectF0(slice, Math.min(Math.max(pre, consonant), Math.max(0, length - 1)), length);
+  const pitch = recordedPitch > 0 ? recordedPitch : detectF0(
+    slice,
+    Math.min(Math.max(pre, consonant), Math.max(0, length - 1)),
+    length
+  );
   return {
     pcm: slice,
     entry: { length, pre, overlap, consonant, pitch }
@@ -231,7 +244,10 @@ function pack(inputs, referencePitch = 220) {
   const view = new Uint8Array(bin);
   let pos = 0;
   for (const chunk of chunks) {
-    view.set(new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength), pos);
+    view.set(
+      new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength),
+      pos
+    );
     pos += chunk.byteLength;
   }
   const manifest = {
@@ -315,8 +331,10 @@ async function main() {
         inputs.push({ oto, pcm, recordedPitch });
       } catch (err) {
         const msg = err instanceof Error ? err.message : String(err);
-        process.stderr.write(`[skip] ${oto.alias} (${basename(wavPath)}): ${msg}
-`);
+        process.stderr.write(
+          `[skip] ${oto.alias} (${basename(wavPath)}): ${msg}
+`
+        );
         skipped++;
       }
     }

package/package.json

@@ -2,7 +2,7 @@
   "author": "onjmin",
   "license": "MIT",
   "name": "@onjmin/koe",
-  "version": "1.0.0",
+  "version": "1.0.3",
   "description": "UTAU concatenative synthesis engine optimized for browser / mobile via WebAssembly + AudioWorklet",
   "type": "module",
   "main": "./dist/index.js",
@@ -20,17 +20,12 @@
   "files": [
     "dist"
   ],
-  "scripts": {
-    "build": "tsup",
-    "dev": "tsx src/dev-server.ts",
-    "watch": "tsup --watch",
-    "convert": "tsx src/converter/cli.ts",
-    "typecheck": "tsc --noEmit",
-    "lint": "biome check src",
-    "clean": "rimraf dist"
-  },
-  "keywords": ["utau", "singing-synthesis", "audioworklet", "wasm"],
-  "packageManager": "pnpm@10.17.1",
+  "keywords": [
+    "utau",
+    "singing-synthesis",
+    "audioworklet",
+    "wasm"
+  ],
   "devDependencies": {
     "@biomejs/biome": "2.5.0",
     "@hono/node-server": "2.0.5",
@@ -39,10 +34,23 @@
     "rimraf": "6.1.3",
     "tsup": "8.5.1",
     "tsx": "4.22.4",
+    "typedoc": "0.28.19",
     "typescript": "6.0.3"
   },
   "volta": {
     "node": "24.16.0",
     "pnpm": "11.7.0"
+  },
+  "scripts": {
+    "build": "tsup",
+    "dev": "tsx src/dev-server.ts",
+    "watch": "tsup --watch",
+    "convert": "tsx src/converter/cli.ts",
+    "typecheck": "tsc --noEmit",
+    "lint": "biome check src",
+    "lint2": "biome format src --write",
+    "clean": "rimraf dist",
+    "patch": "pnpm version patch",
+    "publish-dangerous-dont-use-this-command": "pnpm publish --access public"
   }
-}
+}