carta1 1.1.3 → 1.1.5

package/dist/carta1-worker.min.js.map

@@ -1 +1 @@
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* Carta1 Audio Codec - Core Constants\n */\n\n// Audio format\nexport const SAMPLE_RATE = 44100\nexport const SAMPLES_PER_FRAME = 512\nexport const FRAME_RATE = SAMPLE_RATE / SAMPLES_PER_FRAME\n\n// AEA file format\nexport const AEA_MAGIC = new Uint8Array([0x00, 0x08, 0x00, 0x00])\nexport const AEA_HEADER_SIZE = 2048\nexport const AEA_TITLE_OFFSET = 4\nexport const AEA_TITLE_SIZE = 256\nexport const AEA_FRAME_COUNT_OFFSET = 260\nexport const AEA_CHANNEL_COUNT_OFFSET = 264\n\n// Frame structure\nexport const SOUND_UNIT_SIZE = 212\nexport const FRAME_BITS = SOUND_UNIT_SIZE * 8\nexport const FRAME_OVERHEAD_BITS = 40\nexport const BITRATE_PER_CHANNEL = SOUND_UNIT_SIZE * FRAME_RATE * 8\n\n// BFU (Block Floating Unit) configuration\nexport const NUM_BFUS = 52\nexport const MAX_BFU_SIZE = 20\nexport const BITS_PER_BFU_METADATA = 10\n\nexport const SPECS_PER_BFU = new Int32Array([\n  8, 8, 8, 8, 4, 4, 4, 4, 8, 8, 8, 8, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 7, 7,\n  7, 7, 9, 9, 9, 9, 10, 10, 10, 10, 12, 12, 12, 12, 12, 12, 12, 12, 20, 20, 20,\n  20, 20, 20, 20, 20,\n])\n\nexport const BFU_AMOUNTS_COUNT = 8\nexport const BFU_AMOUNTS = new Int32Array([20, 28, 32, 36, 40, 44, 48, 52])\nexport const BFU_BAND_BOUNDARIES = new Int32Array([20, 36, 52])\n\n// BFU start positions\nexport const BFU_START_LONG = new Int32Array([\n  0, 8, 16, 24, 32, 36, 40, 44, 48, 56, 64, 72, 80, 86, 92, 98, 104, 110, 116,\n  122, 128, 134, 140, 146, 152, 159, 166, 173, 180, 189, 198, 207, 216, 226,\n  236, 246, 256, 268, 280, 292, 304, 316, 328, 340, 352, 372, 392, 412, 432,\n  452, 472, 492,\n])\n\nexport const BFU_START_SHORT = new Int32Array([\n  0, 32, 64, 96, 8, 40, 72, 104, 12, 44, 76, 108, 20, 52, 84, 116, 26, 58, 90,\n  122, 128, 160, 192, 224, 134, 166, 198, 230, 141, 173, 205, 237, 150, 182,\n  214, 246, 256, 288, 320, 352, 384, 416, 448, 480, 268, 300, 332, 364, 396,\n  428, 460, 492,\n])\n\n// Transform sizes\nexport const MDCT_SIZE_SHORT = 64\nexport const MDCT_SIZE_MID = 256\nexport const MDCT_SIZE_LONG = 512\n\n// Window functions\nexport const WINDOW_SHORT = (() => {\n  const table = new Float64Array(32)\n  for (let i = 0; i < 32; i++) {\n    table[i] = Math.sin(((i + 0.5) * Math.PI) / 64)\n  }\n  return table\n})()\n\n// QMF configuration\nexport const QMF_TAPS = 48\nexport const QMF_DELAY = 46\nexport const QMF_HIGH_BAND_DELAY = 39\n\n// QMF prototype filter coefficients\nexport const QMF_COEFFS = new Float32Array([\n  -0.00001461907, -0.00009205479, -0.000056157569, 0.00030117269, 0.0002422519,\n  -0.00085293897, -0.0005205574, 0.0020340169, 0.00078333891, -0.0042153862,\n  -0.00075614988, 0.0078402944, -0.000061169922, -0.01344162, 0.0024626821,\n  0.021736089, -0.007801671, -0.034090221, 0.01880949, 0.054326009,\n  -0.043596379, -0.099384367, 0.13207909, 0.46424159,\n])\n\n// QMF window function\nexport const QMF_WINDOW = (() => {\n  const window = new Float32Array(QMF_TAPS)\n  for (let i = 0; i < 24; i++) {\n    window[i] = QMF_COEFFS[i] * 2.0\n    window[47 - i] = QMF_COEFFS[i] * 2.0\n  }\n  return window\n})()\n\n// QMF even/odd taps for optimization\nexport const QMF_EVEN = (() => {\n  const even = new Float32Array(24)\n  for (let i = 0; i < 24; i++) {\n    even[i] = QMF_WINDOW[i * 2]\n  }\n  return even\n})()\n\nexport const QMF_ODD = (() => {\n  const odd = new Float32Array(24)\n  for (let i = 0; i < 24; i++) {\n    odd[i] = QMF_WINDOW[i * 2 + 1]\n  }\n  return odd\n})()\n\n// FFT sizes for transient detection\nexport const FFT_SIZE_LOW = 128\nexport const FFT_SIZE_MID = 128\nexport const FFT_SIZE_HIGH = 256\n\n// MDCT/IMDCT transform configuration\nexport const MDCT_BAND_CONFIGS = [\n  { size: 128, windowStart: 48 }, // Low band (0-5.5kHz)\n  { size: 128, windowStart: 48 }, // Mid band (5.5-11kHz)\n  { size: 256, windowStart: 112 }, // High band (11-22kHz)\n]\n\nexport const MDCT_SHORT_BLOCK_SIZE = 32\nexport const MDCT_OVERLAP_SIZE = 32\nexport const MDCT_TAIL_WINDOW_SIZE = 16\n\n// WAV format\nexport const WAV_HEADER_SIZE = 44\nexport const WAV_BYTES_PER_SAMPLE = 2\nexport const WAV_BITS_PER_SAMPLE = 16\nexport const WAV_PCM_MAX_POSITIVE = 0x7fff\nexport const WAV_PCM_MAX_NEGATIVE = 0x8000\nexport const WAV_DATA_OFFSET = 36\n\n// Frame serialization\nexport const FRAME_HEADER_BITS = 16\nexport const FRAME_WORD_LENGTH_BITS = 4\nexport const FRAME_SCALE_FACTOR_BITS = 6\n\n// Quantization\nexport const QUANTIZATION_SIGN_BIT_SHIFT = 1\nexport const MAX_WORD_LENGTH_INDEX = 15\nexport const WORD_LENGTH_BITS = new Int32Array([\n  0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,\n])\nexport const SCALE_FACTORS = (() => {\n  const table = new Float64Array(64)\n  for (let i = 0; i < 64; i++) {\n    table[i] = Math.pow(2.0, i / 3.0 - 21)\n  }\n  return table\n})()\n\n// Bit allocation\nexport const INV_POWER_OF_TWO = (() => {\n  const maxBits = WORD_LENGTH_BITS[MAX_WORD_LENGTH_INDEX]\n  const table = new Float64Array(maxBits + 1)\n  for (let b = 0; b <= maxBits; b++) {\n    table[b] = Math.pow(2, -b)\n  }\n  return table\n})()\n","/**\n * Carta1 Audio Codec - Bitstream Operations\n *\n * This module provides low-level bitstream packing and unpacking operations\n * for reading and writing ATRAC1 frame data with bit-level precision.\n */\n\n/**\n * Pack bits into a buffer at a specific bit position\n * @param {Uint8Array} buffer - Destination buffer\n * @param {number} bitPosition - Bit position to start writing at\n * @param {number} value - Value to pack\n * @param {number} bitCount - Number of bits to pack\n */\nexport function packBits(buffer, bitPosition, value, bitCount) {\n  if (bitCount === 0) return\n\n  let byteIndex = Math.floor(bitPosition / 8)\n  let bitOffset = bitPosition % 8\n\n  value &= (1 << bitCount) - 1\n\n  let bitsWritten = 0\n  while (bitsWritten < bitCount && byteIndex < buffer.length) {\n    const bitsAvailable = 8 - bitOffset\n    const bitsToWrite = Math.min(bitCount - bitsWritten, bitsAvailable)\n\n    const shift = bitCount - bitsWritten - bitsToWrite\n    const valueBits = (value >> shift) & ((1 << bitsToWrite) - 1)\n\n    const mask = ((1 << bitsToWrite) - 1) << (bitsAvailable - bitsToWrite)\n    buffer[byteIndex] =\n      (buffer[byteIndex] & ~mask) | (valueBits << (bitsAvailable - bitsToWrite))\n\n    bitsWritten += bitsToWrite\n    byteIndex++\n    bitOffset = 0\n  }\n}\n\n/**\n * Unpack bits from a buffer at a specific bit position\n * @param {Uint8Array} buffer - Source buffer\n * @param {number} bitPosition - Bit position to start reading from\n * @param {number} bitCount - Number of bits to unpack\n * @returns {number} Unpacked unsigned value\n */\nexport function unpackBits(buffer, bitPosition, bitCount) {\n  if (bitCount === 0) return 0\n\n  let byteIndex = Math.floor(bitPosition / 8)\n  let bitOffset = bitPosition % 8\n  let value = 0\n\n  for (let bitsRead = 0; bitsRead < bitCount && byteIndex < buffer.length; ) {\n    const bitsAvailable = 8 - bitOffset\n    const bitsToRead = Math.min(bitCount - bitsRead, bitsAvailable)\n\n    const mask = (1 << bitsToRead) - 1\n    const bits = (buffer[byteIndex] >> (bitsAvailable - bitsToRead)) & mask\n\n    value = (value << bitsToRead) | bits\n    bitsRead += bitsToRead\n    byteIndex++\n    bitOffset = 0\n  }\n\n  return value\n}\n\n/**\n * Unpack signed bits from a buffer at a specific bit position\n * @param {Uint8Array} buffer - Source buffer\n * @param {number} bitPosition - Bit position to start reading from\n * @param {number} bitCount - Number of bits to unpack\n * @returns {number} Unpacked signed value (two's complement)\n */\nexport function unpackSignedBits(buffer, bitPosition, bitCount) {\n  const value = unpackBits(buffer, bitPosition, bitCount)\n  const signBit = 1 << (bitCount - 1)\n  return value >= signBit ? value - (1 << bitCount) : value\n}\n","/**\n * Carta1 Audio Codec - Serialization Module\n *\n * Handles serialization and deserialization of ATRAC1 audio frames and AEA file format.\n * Provides functions for packing encoded frame data into binary format and unpacking\n * it back into structured data for processing.\n *\n * The module manages:\n * - Frame data serialization/deserialization with proper bit packing\n * - AEA file header creation and parsing\n * - Binary format compliance for ATRAC1 specification\n */\n\nimport {\n  SOUND_UNIT_SIZE,\n  BFU_AMOUNTS,\n  SPECS_PER_BFU,\n  AEA_HEADER_SIZE,\n  AEA_MAGIC,\n  AEA_TITLE_OFFSET,\n  AEA_TITLE_SIZE,\n  AEA_FRAME_COUNT_OFFSET,\n  AEA_CHANNEL_COUNT_OFFSET,\n  WORD_LENGTH_BITS,\n  FRAME_HEADER_BITS,\n  FRAME_WORD_LENGTH_BITS,\n  FRAME_SCALE_FACTOR_BITS,\n} from '../core/constants.js'\nimport { packBits, unpackBits, unpackSignedBits } from './bitstream.js'\n\n/**\n * Serializes encoded frame data into binary format\n * @param {Object} frameData - The encoded frame data to serialize\n * @param {number} frameData.nBfu - Number of block floating units\n * @param {number[]} frameData.blockModes - Block modes for each band [0-2]\n * @param {Int32Array} frameData.scaleFactorIndices - Scale factor indices for each BFU\n * @param {Int32Array} frameData.wordLengthIndices - Word length indices for each BFU\n * @param {Int32Array[]} frameData.quantizedCoefficients - Quantized spectral coefficients\n * @returns {Uint8Array} Serialized frame data buffer\n */\nexport function serializeFrame(frameData) {\n  const buffer = new Uint8Array(SOUND_UNIT_SIZE)\n  const view = new DataView(buffer.buffer)\n\n  // Pack header (16 bits)\n  const bfuIndex = BFU_AMOUNTS.indexOf(frameData.nBfu)\n  const header =\n    ((2 - frameData.blockModes[0]) << 14) |\n    ((2 - frameData.blockModes[1]) << 12) |\n    ((3 - frameData.blockModes[2]) << 10) |\n    (bfuIndex << 5)\n\n  view.setUint16(0, header, false)\n  let bitPosition = FRAME_HEADER_BITS\n\n  // Pack word lengths (4 bits each)\n  for (let i = 0; i < frameData.nBfu; i++) {\n    packBits(\n      buffer,\n      bitPosition,\n      frameData.wordLengthIndices[i],\n      FRAME_WORD_LENGTH_BITS\n    )\n    bitPosition += FRAME_WORD_LENGTH_BITS\n  }\n\n  // Pack scale factors (6 bits each)\n  for (let i = 0; i < frameData.nBfu; i++) {\n    packBits(\n      buffer,\n      bitPosition,\n      frameData.scaleFactorIndices[i],\n      FRAME_SCALE_FACTOR_BITS\n    )\n    bitPosition += FRAME_SCALE_FACTOR_BITS\n  }\n\n  // Pack quantized coefficients\n  for (let i = 0; i < frameData.nBfu; i++) {\n    const bitsPerSample = WORD_LENGTH_BITS[frameData.wordLengthIndices[i]]\n    if (bitsPerSample > 0) {\n      const coefficients = frameData.quantizedCoefficients[i]\n      for (const coefficient of coefficients) {\n        const value =\n          coefficient < 0 ? coefficient + (1 << bitsPerSample) : coefficient\n        packBits(buffer, bitPosition, value, bitsPerSample)\n        bitPosition += bitsPerSample\n      }\n    }\n  }\n\n  // Zero padding at end\n  buffer[SOUND_UNIT_SIZE - 3] = 0\n  buffer[SOUND_UNIT_SIZE - 2] = 0\n  buffer[SOUND_UNIT_SIZE - 1] = 0\n\n  return buffer\n}\n\n/**\n * Deserializes binary frame data back into structured format\n * @param {Uint8Array} buffer - Binary frame data buffer\n * @returns {Object} Deserialized frame data\n * @returns {number} returns.nBfu - Number of block floating units\n * @returns {number[]} returns.blockModes - Block modes for each band\n * @returns {Int32Array} returns.scaleFactorIndices - Scale factor indices for each BFU\n * @returns {Int32Array} returns.wordLengthIndices - Word length indices for each BFU\n * @returns {Int32Array[]} returns.quantizedCoefficients - Quantized spectral coefficients\n * @throws {Error} If buffer size is invalid\n */\nexport function deserializeFrame(buffer) {\n  if (buffer.length !== SOUND_UNIT_SIZE) {\n    throw new Error(`Frame must be ${SOUND_UNIT_SIZE} bytes`)\n  }\n\n  const view = new DataView(buffer.buffer, buffer.byteOffset, buffer.byteLength)\n\n  // Unpack header\n  const header = view.getUint16(0, false)\n  const blockModes = [\n    2 - ((header >> 14) & 3),\n    2 - ((header >> 12) & 3),\n    3 - ((header >> 10) & 3),\n  ]\n  const bfuIndex = (header >> 5) & 7\n  const nBfu = BFU_AMOUNTS[bfuIndex]\n\n  let bitPosition = FRAME_HEADER_BITS\n\n  // Unpack word lengths\n  const wordLengthIndices = new Int32Array(nBfu)\n  for (let i = 0; i < nBfu; i++) {\n    wordLengthIndices[i] = unpackBits(\n      buffer,\n      bitPosition,\n      FRAME_WORD_LENGTH_BITS\n    )\n    bitPosition += FRAME_WORD_LENGTH_BITS\n  }\n\n  // Unpack scale factors\n  const scaleFactorIndices = new Int32Array(nBfu)\n  for (let i = 0; i < nBfu; i++) {\n    scaleFactorIndices[i] = unpackBits(\n      buffer,\n      bitPosition,\n      FRAME_SCALE_FACTOR_BITS\n    )\n    bitPosition += FRAME_SCALE_FACTOR_BITS\n  }\n\n  // Unpack coefficients\n  const quantizedCoefficients = []\n  for (let i = 0; i < nBfu; i++) {\n    const bitsPerSample = WORD_LENGTH_BITS[wordLengthIndices[i]]\n    const bfuSize = SPECS_PER_BFU[i]\n    const coefficients = new Int32Array(bfuSize)\n\n    if (bitsPerSample > 0) {\n      for (let j = 0; j < bfuSize; j++) {\n        coefficients[j] = unpackSignedBits(buffer, bitPosition, bitsPerSample)\n        bitPosition += bitsPerSample\n      }\n    }\n\n    quantizedCoefficients.push(coefficients)\n  }\n\n  return {\n    nBfu,\n    scaleFactorIndices,\n    wordLengthIndices,\n    quantizedCoefficients,\n    blockModes,\n  }\n}\n\n/**\n * AEA file format handler for ATRAC1 audio files\n * Manages creation and parsing of AEA file headers\n */\nexport class AeaFile {\n  /**\n   * Creates an AEA file header with the specified metadata\n   * @param {string} [title=''] - Audio file title (max 255 characters)\n   * @param {number} [frameCount=0] - Number of audio frames in the file\n   * @param {number} [channelCount=1] - Number of audio channels (1 or 2)\n   * @returns {Uint8Array} AEA file header buffer\n   */\n  static createHeader(title = '', frameCount = 0, channelCount = 1) {\n    const header = new Uint8Array(AEA_HEADER_SIZE)\n    const view = new DataView(header.buffer)\n\n    // Magic number\n    header.set(AEA_MAGIC, 0)\n\n    // Title (null-terminated string)\n    const titleBytes = new TextEncoder().encode(title)\n    header.set(\n      titleBytes.subarray(0, Math.min(titleBytes.length, AEA_TITLE_SIZE - 1)),\n      AEA_TITLE_OFFSET\n    )\n\n    // Frame count\n    view.setUint32(AEA_FRAME_COUNT_OFFSET, frameCount, true)\n\n    // Channel count\n    header[AEA_CHANNEL_COUNT_OFFSET] = channelCount\n\n    return header\n  }\n\n  /**\n   * Parses an AEA file header to extract metadata\n   * @param {Uint8Array} header - AEA file header buffer\n   * @returns {Object} Parsed header information\n   * @returns {string} returns.title - Audio file title\n   * @returns {number} returns.frameCount - Number of audio frames\n   * @returns {number} returns.channelCount - Number of audio channels\n   * @throws {Error} If header size is invalid or magic number doesn't match\n   */\n  static parseHeader(header) {\n    if (header.length !== AEA_HEADER_SIZE) {\n      throw new Error(`Header must be ${AEA_HEADER_SIZE} bytes`)\n    }\n\n    // Check magic number\n    for (let i = 0; i < AEA_MAGIC.length; i++) {\n      if (header[i] !== AEA_MAGIC[i]) {\n        throw new Error('Invalid AEA file')\n      }\n    }\n\n    const view = new DataView(\n      header.buffer,\n      header.byteOffset,\n      header.byteLength\n    )\n\n    // Extract title\n    const titleEnd = header.indexOf(0, AEA_TITLE_OFFSET)\n    const titleLength =\n      titleEnd === -1 ? AEA_TITLE_SIZE : titleEnd - AEA_TITLE_OFFSET\n    const title = new TextDecoder().decode(\n      header.subarray(AEA_TITLE_OFFSET, AEA_TITLE_OFFSET + titleLength)\n    )\n\n    // Extract metadata\n    const frameCount = view.getUint32(AEA_FRAME_COUNT_OFFSET, true)\n    const channelCount = header[AEA_CHANNEL_COUNT_OFFSET]\n\n    return { title, frameCount, channelCount }\n  }\n}\n","/**\n * Carta1 Audio Codec - Encoder Options\n *\n * This module provides configuration management for the ATRAC1 encoder,\n * including parameter validation and metadata for UI controls.\n */\n\n/**\n * Configuration options for the ATRAC1 encoder\n */\nclass EncoderOptions {\n  /**\n   * Create a new EncoderOptions instance\n   * @param {Object} options - Initial option values\n   */\n  constructor(options = {}) {\n    this.values = {\n      transientThresholdLow: 1.0,\n      transientThresholdMid: 1.5,\n      transientThresholdHigh: 2.0,\n    }\n\n    this.metadata = {\n      transientThresholdLow: {\n        default: this.values.transientThresholdLow,\n        name: 'Low Band Transient Threshold',\n        description:\n          'Controls the transient detection threshold for the low frequency band (0-5.5kHz). Lower values make detection more sensitive, triggering short blocks more easily. Higher values require stronger transients.',\n        range: [0.01, 2],\n        step: 0.01,\n      },\n      transientThresholdMid: {\n        default: this.values.transientThresholdMid,\n        name: 'Mid Band Transient Threshold',\n        description:\n          'Controls the transient detection threshold for the mid frequency band (5.5-11kHz). Lower values make detection more sensitive, triggering short blocks more easily. Higher values require stronger transients.',\n        range: [0.01, 3],\n        step: 0.01,\n      },\n      transientThresholdHigh: {\n        default: this.values.transientThresholdHigh,\n        name: 'High Band Transient Threshold',\n        description:\n          'Controls the transient detection threshold for the high frequency band (11-22kHz). Lower values make detection more sensitive, triggering short blocks more easily. Higher values require stronger transients.',\n        range: [0.01, 4],\n        step: 0.01,\n      },\n    }\n\n    // Apply user-provided options\n    if (options) {\n      this.setOptions(options)\n    }\n  }\n\n  /**\n   * Set multiple options at once\n   * @param {Object} options - Object containing option key-value pairs\n   */\n  setOptions(options) {\n    for (const [key, value] of Object.entries(options)) {\n      if (key in this.values) {\n        this.setValue(key, value)\n      }\n    }\n  }\n\n  /**\n   * Set a single option value with validation\n   * @param {string} key - Option key\n   * @param {*} value - Option value\n   * @throws {Error} If key is unknown or value is out of range\n   */\n  setValue(key, value) {\n    if (!(key in this.metadata)) {\n      throw new Error(`Unknown option: ${key}`)\n    }\n\n    const meta = this.metadata[key]\n    const [min, max] = meta.range\n\n    if (value < min || value > max) {\n      throw new Error(\n        `Value for ${key} must be between ${min} and ${max}, got ${value}`\n      )\n    }\n\n    this.values[key] = value\n  }\n\n  /**\n   * Get the value of an option\n   * @param {string} key - Option key\n   * @returns {*} Option value\n   * @throws {Error} If key is unknown\n   */\n  getValue(key) {\n    if (!(key in this.values)) {\n      throw new Error(`Unknown option: ${key}`)\n    }\n    return this.values[key]\n  }\n\n  get transientThresholdLow() {\n    return this.values.transientThresholdLow\n  }\n\n  get transientThresholdMid() {\n    return this.values.transientThresholdMid\n  }\n\n  get transientThresholdHigh() {\n    return this.values.transientThresholdHigh\n  }\n\n  getMetadata(key) {\n    return this.metadata[key]\n  }\n\n  getAllMetadata() {\n    return { ...this.metadata }\n  }\n\n  reset() {\n    for (const [key, meta] of Object.entries(this.metadata)) {\n      this.values[key] = meta.default\n    }\n  }\n\n  toObject() {\n    return {\n      values: { ...this.values },\n      metadata: { ...this.metadata },\n    }\n  }\n}\n\nexport { EncoderOptions }\n","/**\n * Carta1 Audio Codec - Utilities\n *\n * This module provides common utility functions for the ATRAC1 codec including\n * pipeline composition, error handling, and audio stream processing utilities\n * for delay compensation and frame padding.\n */\n\n/**\n * Throws an error with the given message\n * @param {string} msg - Error message\n * @throws {Error} Always throws an error with the provided message\n */\nexport function throwError(msg) {\n  throw new Error(msg)\n}\n\n/**\n * Creates a pipeline by composing multiple processing stages\n * @param {Object} context - Shared context passed to all stages\n * @param {...Function} stages - Stage functions to compose\n * @returns {Function} Composed pipeline function\n */\nexport function pipe(context, ...stages) {\n  const functions = stages.map((stage) => stage(context))\n\n  return (input) => {\n    return functions.reduce((value, fn) => fn(value), input)\n  }\n}\n\n/**\n * Reverses the spectral order of coefficients for mid/high band processing\n *\n * Used in both MDCT and IMDCT stages to handle spectral reversal required\n * by the ATRAC1 format for mid and high frequency bands.\n *\n * @param {Float32Array} spectrum - Input spectrum coefficients\n * @param {Object} reversalBuffers - Pre-allocated buffers for different spectrum sizes\n * @returns {Float32Array} Spectrum with reversed coefficient order\n */\nexport function reverseSpectrum(spectrum, reversalBuffers) {\n  const reversed = reversalBuffers[spectrum.length]\n  for (let i = 0; i < spectrum.length; i++) {\n    reversed[i] = spectrum[spectrum.length - 1 - i]\n  }\n  return reversed\n}\n\n/**\n * Calculates the starting offset for a frequency band in the coefficient array\n *\n * @param {number} bandIndex - Band index (0=low, 1=mid, 2=high)\n * @returns {number} Starting offset in the 512-sample coefficient array\n */\nexport function calculateBandOffset(bandIndex) {\n  return bandIndex === 0 ? 0 : bandIndex === 1 ? 128 : 256\n}\n\n/**\n * Extracts coefficients for a specific frequency band from the full coefficient array\n *\n * @param {Float32Array} coefficients - Full 512-sample coefficient array\n * @param {number} bandIndex - Band index (0=low, 1=mid, 2=high)\n * @returns {Float32Array} Subarray containing coefficients for the specified band\n */\nexport function extractBandCoefficients(coefficients, bandIndex) {\n  const offsets = [0, 128, 256, 512]\n  return coefficients.subarray(offsets[bandIndex], offsets[bandIndex + 1])\n}\n","/**\n * Carta1 Audio Codec - Quadrature Mirror Filter\n *\n * This module implements QMF analysis and synthesis for subband decomposition\n * in the ATRAC1 codec. The QMF splits the audio signal into multiple frequency\n * bands for independent processing and coding.\n */\n\nimport { QMF_DELAY, QMF_EVEN, QMF_ODD } from '../core/constants.js'\nimport { throwError } from '../utils.js'\n\n/**\n * Perform QMF analysis filtering to split input into low and high frequency bands\n * @param {Float32Array} input - Input audio samples\n * @param {Float32Array} delayLine - QMF delay line state\n * @param {Object} qmfWorkBuffers - Optional work buffers for processing\n * @returns {Object} Object containing lowBand, highBand, and newDelay\n */\nexport function qmfAnalysis(input, delayLine, qmfWorkBuffers = null) {\n  const inputLength = input.length\n  const outputLength = inputLength >> 1\n\n  const workBuffer =\n    qmfWorkBuffers?.delay?.[inputLength] ??\n    throwError(`qmfAnalysis: qmfWorkBuffers.delay[${inputLength}] is required`)\n  workBuffer.set(delayLine)\n  workBuffer.set(input, delayLine.length)\n\n  const lowBand = new Float32Array(outputLength)\n  const highBand = new Float32Array(outputLength)\n\n  for (let i = 0; i < outputLength; i++) {\n    let evenSum = 0\n    let oddSum = 0\n    const offset = i * 2\n\n    // Unrolled convolution\n    for (let j = 0; j < 24; j++) {\n      evenSum += workBuffer[offset + 47 - j * 2] * QMF_EVEN[j]\n      oddSum += workBuffer[offset + 46 - j * 2] * QMF_ODD[j]\n    }\n\n    lowBand[i] = evenSum + oddSum\n    highBand[i] = evenSum - oddSum\n  }\n\n  // Update delay line\n  const newDelay = workBuffer.slice(-QMF_DELAY)\n  return { lowBand, highBand, newDelay }\n}\n\n/**\n * Perform QMF synthesis filtering to reconstruct audio from low and high frequency bands\n * @param {Float32Array} lowBand - Low frequency band samples\n * @param {Float32Array} highBand - High frequency band samples\n * @param {Float32Array} delayLine - QMF delay line state\n * @param {Object} qmfWorkBuffers - Optional work buffers for processing\n * @returns {Object} Object containing output and newDelay\n */\nexport function qmfSynthesis(\n  lowBand,\n  highBand,\n  delayLine,\n  qmfWorkBuffers = null\n) {\n  const subbandLength = lowBand.length\n  const outputLength = subbandLength * 2\n\n  const workBuffer =\n    qmfWorkBuffers?.delay?.[outputLength] ??\n    throwError(\n      `qmfSynthesis: qmfWorkBuffers.delay[${outputLength}] is required`\n    )\n  workBuffer.set(delayLine)\n\n  // Upsample and merge\n  for (let i = 0; i < subbandLength; i++) {\n    const low = lowBand[i]\n    const high = highBand[i]\n    const offset = QMF_DELAY + i * 2\n    workBuffer[offset] = 0.5 * (low + high)\n    workBuffer[offset + 1] = 0.5 * (low - high)\n  }\n\n  const output = new Float32Array(outputLength)\n\n  // Convolution\n  for (let i = 0; i < subbandLength; i++) {\n    const offset = i * 2\n    let sample0 = 0\n    let sample1 = 0\n\n    for (let j = 0; j < 24; j++) {\n      const idx = offset + j * 2\n      sample0 += workBuffer[idx] * QMF_EVEN[j]\n      sample1 += workBuffer[idx + 1] * QMF_ODD[j]\n    }\n\n    output[i * 2] = sample1\n    output[i * 2 + 1] = sample0\n  }\n\n  const newDelay = workBuffer.slice(-QMF_DELAY)\n  return { output, newDelay }\n}\n","/**\n * Carta1 Audio Codec - Fast Fourier Transform\n *\n * This module implements a radix-2 Cooley-Tukey FFT algorithm for frequency domain\n * analysis used in psychoacoustic modeling and transient detection.\n */\nexport class FFT {\n  static _cache = new Map()\n\n  /**\n   * Perform in-place FFT on complex data\n   * @param {Float32Array} real - Real part of input/output data\n   * @param {Float32Array} imag - Imaginary part of input/output data\n   */\n  static fft(real, imag) {\n    const n = real.length\n    if (n !== imag.length)\n      throw new Error('Real and imaginary arrays must have the same length.')\n    if ((n & (n - 1)) !== 0) throw new Error('Input size must be a power of 2.')\n    if (n <= 1) return\n\n    let cache = FFT._cache.get(n)\n    if (!cache) {\n      // build plan in Float64 for precision\n      const log2n = Math.log2(n)\n\n      const bitRev = new Uint32Array(n)\n      for (let i = 1; i < n; i++) {\n        bitRev[i] = (bitRev[i >>> 1] >>> 1) | ((i & 1) << (log2n - 1))\n      }\n\n      const cosTbl = new Float64Array(n / 2)\n      const sinTbl = new Float64Array(n / 2)\n      for (let i = 0; i < n / 2; i++) {\n        const angle = (-2 * Math.PI * i) / n\n        // compute in f64, then enforce |W| = 1 to remove tiny drift\n        let c = Math.cos(angle)\n        let s = Math.sin(angle)\n        const invm = 1 / Math.hypot(c, s) // ~1, but fixes cos^2+sin^2 drift\n        cosTbl[i] = c * invm\n        sinTbl[i] = s * invm\n      }\n\n      cache = { bitRev, cos: cosTbl, sin: sinTbl }\n      FFT._cache.set(n, cache)\n    }\n\n    const { bitRev, cos: cosTbl, sin: sinTbl } = cache\n\n    // work in Float64 to avoid per-stage Float32 rounding\n    const needCopyBack =\n      !(real instanceof Float64Array) || !(imag instanceof Float64Array)\n    const r = needCopyBack ? new Float64Array(real) : real\n    const im = needCopyBack ? new Float64Array(imag) : imag\n\n    // bit reversal\n    for (let i = 0; i < n; i++) {\n      const j = bitRev[i]\n      if (j > i) {\n        let t = r[i]\n        r[i] = r[j]\n        r[j] = t\n        t = im[i]\n        im[i] = im[j]\n        im[j] = t\n      }\n    }\n\n    // size-2 butterflies\n    for (let i = 0; i < n; i += 2) {\n      const i1 = i + 1\n      const er = r[i],\n        ei = im[i]\n      const tr = r[i1],\n        ti = im[i1]\n      r[i] = er + tr\n      im[i] = ei + ti\n      r[i1] = er - tr\n      im[i1] = ei - ti\n    }\n\n    // size-4 butterflies (uses exact W = -i at k=1)\n    if (n >= 4) {\n      for (let i = 0; i < n; i += 4) {\n        const e0 = i,\n          o0 = i + 2\n        const er0 = r[e0],\n          ei0 = im[e0]\n        const tr0 = r[o0],\n          ti0 = im[o0]\n        r[e0] = er0 + tr0\n        im[e0] = ei0 + ti0\n        r[o0] = er0 - tr0\n        im[o0] = ei0 - ti0\n\n        const e1 = i + 1,\n          o1 = i + 3\n        const tr1 = im[o1] // multiply by -i\n        const ti1 = -r[o1]\n        const er1 = r[e1],\n          ei1 = im[e1]\n        r[e1] = er1 + tr1\n        im[e1] = ei1 + ti1\n        r[o1] = er1 - tr1\n        im[o1] = ei1 - ti1\n      }\n    }\n\n    // size >= 8\n    for (let stride = 8; stride <= n; stride <<= 1) {\n      const half = stride >>> 1\n      const step = (n / stride) | 0\n      for (let i = 0; i < n; i += stride) {\n        let k = 0\n        for (let j = 0; j < half; j++, k += step) {\n          const even = i + j\n          const odd = even + half\n\n          const cr = cosTbl[k],\n            sr = sinTbl[k]\n          const orr = r[odd],\n            oii = im[odd]\n\n          const tr = orr * cr - oii * sr\n          const ti = orr * sr + oii * cr\n\n          const er = r[even],\n            ei = im[even]\n          r[even] = er + tr\n          im[even] = ei + ti\n          r[odd] = er - tr\n          im[odd] = ei - ti\n        }\n      }\n    }\n\n    if (needCopyBack) {\n      for (let i = 0; i < n; i++) {\n        real[i] = r[i]\n        imag[i] = im[i]\n      }\n    }\n  }\n}\n","/**\n * Carta1 Audio Codec - Modified Discrete Cosine Transform\n *\n * This module implements forward and inverse MDCT transforms for the ATRAC1 codec.\n * The MDCT provides time-frequency analysis with perfect reconstruction and\n * 50% overlap between adjacent blocks to avoid blocking artifacts.\n */\n\nimport { FFT } from './fft.js'\nimport {\n  MDCT_SIZE_SHORT,\n  MDCT_SIZE_MID,\n  MDCT_SIZE_LONG,\n} from '../core/constants.js'\nimport { throwError } from '../utils.js'\n\n/**\n * Base class for MDCT and IMDCT transforms\n */\nclass MDCTBase {\n  constructor(size, scale) {\n    this.size = size\n    this.halfSize = size >> 1\n    this.quarterSize = size >> 2\n    this.fftSize = this.halfSize >> 1\n\n    const alpha = (2.0 * Math.PI) / (8.0 * size)\n    const omega = (2.0 * Math.PI) / size\n    const scaleRoot = Math.sqrt(scale / size)\n\n    this.sinCosTable = new Float64Array(this.halfSize)\n    for (let i = 0; i < this.quarterSize; i++) {\n      const angle = omega * i + alpha\n      this.sinCosTable[i * 2] = scaleRoot * Math.cos(angle)\n      this.sinCosTable[i * 2 + 1] = scaleRoot * Math.sin(angle)\n    }\n  }\n}\n\n/**\n * Forward MDCT transform\n */\nexport class MDCT extends MDCTBase {\n  constructor(size, scale) {\n    super(size, scale)\n  }\n\n  /**\n   * Perform forward MDCT transform\n   * @param {Float32Array} input - Time-domain input samples\n   * @param {Object} mdctBuffers - Optional work buffers for FFT\n   * @returns {Float32Array} Frequency-domain MDCT coefficients\n   */\n  transform(input, mdctBuffers = null) {\n    const buffers =\n      mdctBuffers?.[this.fftSize] ??\n      throwError(`MDCT.transform: mdctBuffers[${this.fftSize}] is required`)\n    const real =\n      buffers?.real ??\n      throwError(\n        `MDCT.transform: mdctBuffers[${this.fftSize}].real is required`\n      )\n    const imag =\n      buffers?.imag ??\n      throwError(\n        `MDCT.transform: mdctBuffers[${this.fftSize}].imag is required`\n      )\n\n    const n4 = this.quarterSize\n    const n34 = 3 * n4\n\n    real.fill(0)\n    imag.fill(0)\n\n    // Pre-FFT butterfly\n    for (let i = 0; i < n4; i += 2) {\n      const i1 = n34 - 1 - i\n      const i2 = n34 + i\n      const i3 = n4 + i\n      const i4 = n4 - 1 - i\n\n      const r = input[i1] + input[i2]\n      const im = input[i3] - input[i4]\n      const c = this.sinCosTable[i]\n      const s = this.sinCosTable[i + 1]\n\n      real[i >> 1] = r * c + im * s\n      imag[i >> 1] = im * c - r * s\n    }\n\n    for (let i = n4; i < this.halfSize; i += 2) {\n      const idx = i >> 1\n      const i1 = n34 - 1 - i\n      const i2 = i - n4\n      const i3 = n4 + i\n      const i4 = 5 * n4 - 1 - i\n\n      const r = input[i1] - input[i2]\n      const im = input[i3] + input[i4]\n      const c = this.sinCosTable[i]\n      const s = this.sinCosTable[i + 1]\n\n      real[idx] = r * c + im * s\n      imag[idx] = im * c - r * s\n    }\n\n    FFT.fft(real, imag)\n\n    // Post-FFT processing\n    const output = new Float32Array(this.halfSize)\n    for (let i = 0; i < this.fftSize; i++) {\n      const c = this.sinCosTable[i * 2]\n      const s = this.sinCosTable[i * 2 + 1]\n      const re = real[i]\n      const im = imag[i]\n\n      output[i * 2] = -re * c - im * s\n      output[this.halfSize - 1 - i * 2] = -re * s + im * c\n    }\n\n    return output\n  }\n}\n\n/**\n * Inverse MDCT transform\n */\nexport class IMDCT extends MDCTBase {\n  constructor(size, scale = null) {\n    super(size, scale || size)\n  }\n\n  /**\n   * Perform inverse MDCT transform\n   * @param {Float32Array} input - Frequency-domain MDCT coefficients\n   * @param {Object} mdctBuffers - Optional work buffers for FFT\n   * @returns {Float32Array} Time-domain output samples\n   */\n  transform(input, mdctBuffers = null) {\n    const buffers =\n      mdctBuffers?.[this.fftSize] ??\n      throwError(`IMDCT.transform: mdctBuffers[${this.fftSize}] is required`)\n    const real =\n      buffers?.real ??\n      throwError(\n        `IMDCT.transform: mdctBuffers[${this.fftSize}].real is required`\n      )\n    const imag =\n      buffers?.imag ??\n      throwError(\n        `IMDCT.transform: mdctBuffers[${this.fftSize}].imag is required`\n      )\n\n    const n4 = this.quarterSize\n    const n34 = 3 * n4\n\n    real.fill(0)\n    imag.fill(0)\n\n    // Pre-FFT processing\n    for (let i = 0; i < this.fftSize; i++) {\n      const i2 = i * 2\n      const r = -input[i2]\n      const im = -input[this.halfSize - 1 - i2]\n      const c = this.sinCosTable[i2]\n      const s = this.sinCosTable[i2 + 1]\n\n      real[i] = im * s + r * c\n      imag[i] = im * c - r * s\n    }\n\n    FFT.fft(real, imag)\n\n    // Post-FFT butterfly\n    const output = new Float32Array(this.size)\n\n    for (let i = 0; i < this.fftSize / 2; i++) {\n      const i2 = i * 2\n      const c = this.sinCosTable[i2]\n      const s = this.sinCosTable[i2 + 1]\n      const re = real[i]\n      const im = imag[i]\n\n      const r1 = re * c + im * s\n      const i1 = re * s - im * c\n\n      output[n34 - 1 - i2] = r1\n      output[n34 + i2] = r1\n      output[n4 + i2] = i1\n      output[n4 - 1 - i2] = -i1\n    }\n\n    for (let i = this.fftSize / 2; i < this.fftSize; i++) {\n      const idx = (i - this.fftSize / 2) * 2 + n4\n      const i2 = i * 2\n      const c = this.sinCosTable[i2]\n      const s = this.sinCosTable[i2 + 1]\n      const re = real[i]\n      const im = imag[i]\n\n      const r1 = re * c + im * s\n      const i1 = re * s - im * c\n\n      output[n34 - 1 - idx] = r1\n      output[idx - n4] = -r1\n      output[n4 + idx] = i1\n      output[5 * n4 - 1 - idx] = i1\n    }\n\n    return output\n  }\n}\n\n// Pre-instantiated transforms\nexport const mdct64 = new MDCT(MDCT_SIZE_SHORT, 0.5)\nexport const mdct256 = new MDCT(MDCT_SIZE_MID, 0.5)\nexport const mdct512 = new MDCT(MDCT_SIZE_LONG, 1.0)\n\nexport const imdct64 = new IMDCT(MDCT_SIZE_SHORT, MDCT_SIZE_SHORT * 8)\nexport const imdct256 = new IMDCT(MDCT_SIZE_MID, MDCT_SIZE_MID * 8)\nexport const imdct512 = new IMDCT(MDCT_SIZE_LONG, MDCT_SIZE_LONG * 4)\n\n/**\n * Perform overlap-add operation for MDCT reconstruction\n * @param {Float32Array} prev - Previous block samples\n * @param {Float32Array} curr - Current block samples\n * @param {Float32Array} window - Window function coefficients\n * @returns {Float32Array} Overlap-added output samples\n */\nexport function overlapAdd(prev, curr, window) {\n  const size = prev.length\n  const output = new Float32Array(size * 2)\n\n  for (let i = 0; i < size; i++) {\n    const w1 = window[i]\n    const w2 = window[2 * size - 1 - i]\n    const p = prev[i]\n    const c = curr[size - 1 - i]\n\n    output[i] = p * w2 - c * w1\n    output[2 * size - 1 - i] = p * w1 + c * w2\n  }\n\n  return output\n}\n","/**\n * Carta1 Audio Codec - Quantization Module\n *\n * This module handles the quantization and dequantization of MDCT frequency coefficients\n * for the Carta1 audio codec. It provides functions to convert floating-point frequency\n * domain data into integer representations suitable for bitstream encoding.\n *\n * Key functions:\n * - Quantization of coefficients using scale factors and bit depths\n * - Dequantization for reconstruction during decoding\n * - BFU (Band Frequency Unit) organization for psychoacoustic processing\n *\n * The quantization process uses configurable scale factors to maintain audio quality\n * while achieving the target compression ratio. The module supports both uniform\n * and non-uniform quantization schemes based on perceptual importance.\n */\n\nimport {\n  SPECS_PER_BFU,\n  BFU_START_LONG,\n  BFU_START_SHORT,\n  BFU_BAND_BOUNDARIES,\n  SCALE_FACTORS,\n  QUANTIZATION_SIGN_BIT_SHIFT,\n} from '../core/constants.js'\n\n/**\n * Quantize coefficients.\n * @param {Float32Array} coefficients\n * @param {number} scaleFactorIndex\n * @param {number} bitsPerSample\n * @returns {Int32Array} out\n */\nexport function quantize(coefficients, scaleFactorIndex, bitsPerSample) {\n  const length = coefficients.length\n  const out = new Int32Array(length)\n  if (bitsPerSample === 0 || scaleFactorIndex === 0) {\n    out.fill(0, 0, length)\n    return out\n  }\n\n  const scaleFactor = SCALE_FACTORS[scaleFactorIndex]\n  const quantRange = (1 << (bitsPerSample - QUANTIZATION_SIGN_BIT_SHIFT)) - 1\n  const normFactor = quantRange / scaleFactor\n\n  const hi = quantRange\n  const lo = -quantRange\n\n  for (let i = 0; i < length; i++) {\n    const x = coefficients[i] * normFactor\n    const y = (x + (x >= 0 ? 0.5 : -0.5)) | 0\n    out[i] = y > hi ? hi : y < lo ? lo : y\n  }\n\n  return out\n}\n\n/**\n * Dequantize coefficients.\n * @param {Int32Array} quantized\n * @param {number} scaleFactorIndex\n * @param {number} bitsPerSample\n * @returns {Float32Array}\n */\nexport function dequantize(quantized, scaleFactorIndex, bitsPerSample) {\n  if (bitsPerSample === 0 || scaleFactorIndex === 0) {\n    return new Float32Array(quantized.length)\n  }\n\n  const scaleFactor = SCALE_FACTORS[scaleFactorIndex]\n  const quantRange = (1 << (bitsPerSample - QUANTIZATION_SIGN_BIT_SHIFT)) - 1\n\n  const deq = new Float32Array(quantized.length)\n  for (let i = 0; i < quantized.length; i++) {\n    deq[i] = (quantized[i] * scaleFactor) / quantRange\n  }\n  return deq\n}\n\n/**\n * Groups MDCT coefficients into Band Frequency Units (BFUs) for bit allocation.\n *\n * This function organizes frequency-domain coefficients into perceptually meaningful\n * frequency bands (BFUs) based on the block modes for each of the three frequency bands.\n * It handles both long and short blocks, with different frequency resolutions and\n * BFU start positions for each block type.\n *\n * The function implements copy-avoidance optimization: when a BFU slice is fully\n * contained within a frequency band, it directly copies the data without buffer\n * filling. For BFUs that span band boundaries or extend beyond the available\n * coefficients, it handles partial copying and zero-padding appropriately.\n *\n * Frequency Band Structure:\n * - Band 0: 0-128 coefficients (low frequencies)\n * - Band 1: 128-256 coefficients (mid frequencies)\n * - Band 2: 256-512 coefficients (high frequencies)\n *\n * Block Types:\n * - Long blocks (mode 0): Higher frequency resolution, fewer temporal segments\n * - Short blocks (mode 1): Lower frequency resolution, more temporal segments\n *\n * @param {Float32Array} coefficients\n * @param {Array<number>} blockModes\n * @returns {{bfuData: Array<Float32Array>, bfuSizes: Array<number>, bfuCount: number}}\n */\nexport function groupIntoBFUs(coefficients, blockModes) {\n  const bfuData = []\n  const bfuSizes = []\n\n  let coeffIndex = 0\n  let bfuIndex = 0\n\n  for (let band = 0; band < 3; band++) {\n    const bandStart = coeffIndex\n    const bandSize = band === 2 ? 256 : 128\n    const bandEnd = band < 2 ? BFU_BAND_BOUNDARIES[band] : SPECS_PER_BFU.length\n    const isLongBlock = blockModes[band] === 0\n    const startPositions = isLongBlock ? BFU_START_LONG : BFU_START_SHORT\n\n    while (bfuIndex < bandEnd) {\n      const size = SPECS_PER_BFU[bfuIndex]\n      const startPos = startPositions[bfuIndex] - bandStart\n      const endPos = startPos + size\n      const bfu = new Float32Array(size)\n\n      if (startPos >= 0 && endPos <= bandSize) {\n        bfu.set(coefficients.subarray(bandStart + startPos, bandStart + endPos))\n      } else {\n        bfu.fill(0)\n        if (startPos < bandSize && endPos > 0) {\n          const srcStart = Math.max(0, startPos)\n          const srcEnd = Math.min(bandSize, endPos)\n          bfu.set(\n            coefficients.subarray(bandStart + srcStart, bandStart + srcEnd),\n            Math.max(0, -startPos)\n          )\n        }\n      }\n\n      bfuData.push(bfu)\n      bfuSizes.push(size)\n      bfuIndex++\n    }\n\n    coeffIndex += bandSize\n  }\n\n  return { bfuData, bfuSizes, bfuCount: bfuIndex }\n}\n","/**\n * Carta1 Audio Codec - Bit Allocation Module\n *\n * This module implements the bit allocation system for the Carta1 audio codec,\n * responsible for distributing available bits across frequency bands to minimize\n * perceptual distortion while maintaining the target bitrate.\n *\n * The module uses Rate-Distortion Optimization (RDO) with a greedy algorithm\n * that allocates bits to Band Frequency Units (BFUs) based on the perceptual\n * benefit per bit spent. It employs psychoacoustic modeling to determine\n * scale factors and uses a priority queue to efficiently find the optimal\n * bit distribution.\n *\n * Key components:\n * - RDO-based bit allocation using exponential distortion modeling\n * - Scale factor computation for each frequency band\n * - Max-heap priority queue for greedy optimization\n * - Support for variable word lengths per BFU\n *\n * The algorithm ensures that bits are allocated where they provide the\n * greatest reduction in audible distortion, resulting in high-quality\n * audio compression at the target bitrate.\n */\n\nimport {\n  NUM_BFUS,\n  FRAME_BITS,\n  FRAME_OVERHEAD_BITS,\n  BITS_PER_BFU_METADATA,\n  MAX_WORD_LENGTH_INDEX,\n  WORD_LENGTH_BITS,\n  INV_POWER_OF_TWO,\n  SCALE_FACTORS,\n} from '../core/constants.js'\n\n/**\n * Allocate bits to BFUs using optimized greedy Lp-RDO\n * @param {Array<Float32Array>} bfuData\n * @param {Int32Array} bfuSizes\n * @param {number} maxBfuCount\n * @returns {{bfuCount:number, allocation:Int32Array, scaleFactorIndices:Int32Array}}\n */\nexport function allocateBits(bfuData, bfuSizes, maxBfuCount) {\n  const usedBFU = maxBfuCount\n  const avail =\n    FRAME_BITS - FRAME_OVERHEAD_BITS - usedBFU * BITS_PER_BFU_METADATA\n\n  const rdoResult = distributeBitsRDO(usedBFU, bfuData, bfuSizes, avail)\n\n  return {\n    bfuCount: usedBFU,\n    allocation: rdoResult.wordLengths,\n    scaleFactorIndices: rdoResult.scaleFactorIndices,\n  }\n}\n\n/**\n * Distributes available bits across Band Frequency Units (BFUs) using Rate-Distortion Optimization.\n *\n * This function implements an optimized greedy RDO allocator that minimizes perceptual distortion\n * while staying within the bit budget. It uses a priority queue (max-heap) to greedily allocate\n * bits to the BFUs that provide the greatest distortion reduction per bit spent.\n *\n * @param {number} activeBfuCount\n * @param {Array<Float32Array>} bfuData\n * @param {Int32Array} bfuSizes\n * @param {number} remainingBits\n * @returns {{wordLengths: Int32Array, scaleFactorIndices: Int32Array}}\n */\nfunction distributeBitsRDO(activeBfuCount, bfuData, bfuSizes, remainingBits) {\n  const scaleFactorTable = new Int32Array(NUM_BFUS)\n  const wordLengths = new Int32Array(activeBfuCount)\n\n  const heapIndices = []\n  const heapPriorities = []\n\n  const deltaDistPerCoeff = (bfuIndex, currentWl, nextWl) => {\n    const sfi = scaleFactorTable[bfuIndex]\n    const scaleFactor = SCALE_FACTORS[sfi]\n    if (scaleFactor === 0) return 0\n\n    const bits1 = WORD_LENGTH_BITS[currentWl] | 0\n    const bits2 = WORD_LENGTH_BITS[nextWl] | 0\n\n    const f1 = bits1 === 0 ? 2.0 : INV_POWER_OF_TWO[bits1]\n    const f2 = INV_POWER_OF_TWO[bits2]\n\n    return scaleFactor * (f1 - f2)\n  }\n\n  for (let bfuIndex = 0; bfuIndex < activeBfuCount; bfuIndex++) {\n    const sz = bfuSizes[bfuIndex] | 0\n    if (sz === 0) continue\n\n    const sfi = findScaleFactor(bfuData[bfuIndex].subarray(0, sz))\n    scaleFactorTable[bfuIndex] = sfi\n\n    if (sfi === 0) continue\n\n    const currentWl = 0\n    const nextWl = 1\n    const deltaBits =\n      (WORD_LENGTH_BITS[nextWl] - WORD_LENGTH_BITS[currentWl]) | 0\n    if (deltaBits <= 0) continue\n\n    const ddPerCoeff = deltaDistPerCoeff(bfuIndex, currentWl, nextWl)\n    heapIndices.push(bfuIndex)\n    heapPriorities.push(ddPerCoeff / deltaBits)\n  }\n\n  // Heapify\n  for (let i = (heapIndices.length >> 1) - 1; i >= 0; i--)\n    siftDown(heapIndices, heapPriorities, i)\n\n  // Greedy spending\n  while (remainingBits > 0 && heapIndices.length > 0) {\n    const bfu = heapIndices[0]\n    const cur = wordLengths[bfu] | 0\n    const nxt = cur + 1\n\n    const sz = bfuSizes[bfu] | 0\n    const deltaBits = (WORD_LENGTH_BITS[nxt] - WORD_LENGTH_BITS[cur]) | 0\n    const cost = deltaBits * sz\n\n    if (cost > remainingBits || cost <= 0) {\n      popRoot(heapIndices, heapPriorities)\n      continue\n    }\n\n    remainingBits -= cost\n    wordLengths[bfu] = nxt\n\n    if (nxt < MAX_WORD_LENGTH_INDEX) {\n      const nxt2 = nxt + 1\n      const deltaBits2 = (WORD_LENGTH_BITS[nxt2] - WORD_LENGTH_BITS[nxt]) | 0\n      if (deltaBits2 > 0) {\n        heapPriorities[0] = deltaDistPerCoeff(bfu, nxt, nxt2) / deltaBits2\n        siftDown(heapIndices, heapPriorities, 0)\n      } else {\n        popRoot(heapIndices, heapPriorities)\n      }\n    } else {\n      popRoot(heapIndices, heapPriorities)\n    }\n  }\n\n  return { wordLengths, scaleFactorIndices: scaleFactorTable }\n}\n\n/**\n * Find the optimal scale factor index for a set of coefficients.\n * Same semantics as the original.\n * @param {Float32Array} coefficients\n * @returns {number}\n */\nexport function findScaleFactor(coefficients) {\n  let maxAmplitude = 0.0\n  for (let i = 0; i < coefficients.length; i++) {\n    const a = Math.abs(coefficients[i])\n    if (a > maxAmplitude) maxAmplitude = a\n  }\n  if (maxAmplitude === 0) return 0\n  const index = Math.ceil(3 * (Math.log2(maxAmplitude) + 21))\n  return Math.max(0, Math.min(63, index))\n}\n\n/**\n * Removes the root element from a max-heap and maintains heap property.\n *\n * This function efficiently removes the maximum priority element from the heap\n * by replacing it with the last element and then restoring the heap property\n * through sifting down.\n *\n * @param {Array<number>} heapIndices\n * @param {Array<number>} heapPriorities\n */\nfunction popRoot(heapIndices, heapPriorities) {\n  const last = heapIndices.length - 1\n  heapIndices[0] = heapIndices[last]\n  heapPriorities[0] = heapPriorities[last]\n  heapIndices.pop()\n  heapPriorities.pop()\n  if (heapIndices.length) siftDown(heapIndices, heapPriorities, 0)\n}\n\n/**\n * Restores max-heap property by sifting an element down from a given position.\n *\n * This function maintains the heap invariant where parent nodes have higher\n * priority values than their children. It compares the element at startIndex\n * with its children and swaps with the larger child until the heap property\n * is satisfied, or it reaches a leaf position.\n *\n * @param {Array<number>} heapIndices\n * @param {Array<number>} heapPriorities\n * @param {number} startIndex\n */\nfunction siftDown(heapIndices, heapPriorities, startIndex) {\n  const heapSize = heapIndices.length\n  let i = startIndex\n  const idxVal = heapIndices[i]\n  const prVal = heapPriorities[i]\n\n  while (true) {\n    const l = (i << 1) + 1\n    const r = l + 1\n    let maxI = i\n    let maxP = prVal\n\n    if (l < heapSize && heapPriorities[l] > maxP) {\n      maxI = l\n      maxP = heapPriorities[l]\n    }\n    if (r < heapSize && heapPriorities[r] > maxP) {\n      maxI = r\n    }\n    if (maxI === i) break\n\n    heapIndices[i] = heapIndices[maxI]\n    heapPriorities[i] = heapPriorities[maxI]\n    i = maxI\n  }\n\n  heapIndices[i] = idxVal\n  heapPriorities[i] = prVal\n}\n","/**\n * Carta1 Audio Codec - Buffer Management\n */\n\nimport { QMF_DELAY, QMF_HIGH_BAND_DELAY } from './constants.js'\n\nexport class BufferPool {\n  constructor() {\n    // Transform buffers\n    this.transformBuffers = {\n      64: new Float32Array(64),\n      128: new Float32Array(128),\n      256: new Float32Array(256),\n      512: new Float32Array(512),\n    }\n\n    // QMF work buffers\n    this.qmfWorkBuffers = {\n      delay: {\n        128: new Float32Array(QMF_DELAY + 128),\n        256: new Float32Array(QMF_DELAY + 256),\n        512: new Float32Array(QMF_DELAY + 512),\n      },\n      highBandDelay: {\n        128: new Float32Array(QMF_HIGH_BAND_DELAY + 128),\n        256: new Float32Array(QMF_HIGH_BAND_DELAY + 256),\n      },\n    }\n\n    // QMF delay lines\n    this.qmfDelays = {\n      lowBand: new Float32Array(QMF_DELAY),\n      midBand: new Float32Array(QMF_DELAY),\n      highBand: new Float32Array(QMF_HIGH_BAND_DELAY),\n    }\n\n    // Transient detector\n    this.transientDetection = [\n      new Float32Array(64),\n      new Float32Array(64),\n      new Float32Array(128),\n    ]\n\n    // MDCT/IMDCT work buffers\n    this.mdctBuffers = {\n      16: {\n        real: new Float32Array(16),\n        imag: new Float32Array(16),\n      },\n      64: {\n        real: new Float32Array(64),\n        imag: new Float32Array(64),\n      },\n      128: {\n        real: new Float32Array(128),\n        imag: new Float32Array(128),\n      },\n    }\n\n    // MDCT overlap buffers (encoder)\n    this.mdctOverlap = [\n      new Float32Array(32),\n      new Float32Array(32),\n      new Float32Array(32),\n    ]\n\n    // IMDCT overlap buffers (decoder)\n    this.imdctOverlap = [\n      new Float32Array(256),\n      new Float32Array(256),\n      new Float32Array(512),\n    ]\n\n    // Spectrum reversal buffers\n    this.reversalBuffers = {\n      32: new Float32Array(32),\n      128: new Float32Array(128),\n      256: new Float32Array(256),\n    }\n  }\n}\n","/**\n * Carta1 Audio Codec - Transient Detection Module\n *\n * Detects transients using spectral flux analysis to prevent pre-echo artifacts.\n * Combines multiple spectral features to identify sudden changes in signal\n * characteristics that require special handling during encoding.\n */\n\nimport { FFT } from '../transforms/fft.js'\n\n/**\n * Perform FFT on time-domain samples and return magnitude spectrum\n * @param {Float32Array} samples - Time-domain samples\n * @param {number} fftSize - FFT size (must be power of 2)\n * @returns {Float32Array} Magnitude spectrum (positive frequencies only)\n */\nexport function performFFT(samples, fftSize) {\n  const real = new Float32Array(fftSize)\n  const imag = new Float32Array(fftSize)\n\n  // Copy samples with zero-padding if necessary\n  const copyLen = Math.min(samples.length, fftSize)\n  real.set(samples.subarray(0, copyLen))\n\n  // Perform FFT\n  FFT.fft(real, imag)\n\n  // Calculate magnitude spectrum for positive frequencies\n  const magnitude = new Float32Array(fftSize / 2)\n  for (let i = 0; i < fftSize / 2; i++) {\n    magnitude[i] = Math.sqrt(real[i] * real[i] + imag[i] * imag[i])\n  }\n\n  return magnitude\n}\n\n/**\n * Detect transients by analyzing spectral changes between frames\n * @param {Float32Array} currentCoeffs - Current frame coefficients\n * @param {Float32Array} prevCoeffs - Previous frame coefficients\n * @param {number} threshold - Detection threshold\n * @returns {boolean} True if transient detected\n */\nexport function detectTransient(currentCoeffs, prevCoeffs, threshold) {\n  // Can't detect transients without previous frame\n  if (!prevCoeffs) return false\n\n  // Calculate all spectral features\n  const features = calculateSpectralFeatures(currentCoeffs, prevCoeffs)\n\n  // Combine features using perceptually-motivated weighting\n  const transientScore = calculateTransientScore(features)\n\n  return transientScore > threshold\n}\n\n/**\n * Calculate all spectral features for transient detection\n * @param {Float32Array} currentCoeffs - Current frame coefficients\n * @param {Float32Array} prevCoeffs - Previous frame coefficients\n * @returns {Object} Computed spectral features\n */\nfunction calculateSpectralFeatures(currentCoeffs, prevCoeffs) {\n  // Calculate spectral flux (positive differences only)\n  const spectralFlux = calculateSpectralFlux(currentCoeffs, prevCoeffs)\n\n  // Calculate spectral flatness for both frames\n  const currentFlatness = calculateSpectralFlatness(currentCoeffs)\n  const prevFlatness = calculateSpectralFlatness(prevCoeffs)\n  const flatnessChange = Math.abs(currentFlatness - prevFlatness)\n\n  // Calculate high-frequency content ratio\n  const currentHfRatio = calculateHighFrequencyRatio(currentCoeffs)\n  const prevHfRatio = calculateHighFrequencyRatio(prevCoeffs)\n  const hfChange = Math.abs(currentHfRatio - prevHfRatio)\n\n  // Calculate energy change in dB\n  const energyChange = calculateEnergyChange(currentCoeffs, prevCoeffs)\n\n  return {\n    spectralFlux,\n    flatnessChange,\n    hfChange,\n    energyChange,\n  }\n}\n\n/**\n * Calculate normalized spectral flux (positive spectral differences)\n * High values indicate sudden onset of new spectral components\n */\nfunction calculateSpectralFlux(currentCoeffs, prevCoeffs) {\n  let flux = 0\n  let currentEnergy = 0\n\n  for (let i = 0; i < currentCoeffs.length; i++) {\n    const currentMag = Math.abs(currentCoeffs[i])\n    const prevMag = Math.abs(prevCoeffs[i])\n\n    // Only count positive differences (new energy)\n    const diff = currentMag - prevMag\n    if (diff > 0) {\n      flux += diff\n    }\n\n    currentEnergy += currentMag * currentMag\n  }\n\n  // Normalize by current frame energy to make threshold independent of level\n  const normalization = Math.sqrt(currentEnergy) || 1e-6\n  return flux / normalization\n}\n\n/**\n * Calculate spectral flatness (geometric mean / arithmetic mean)\n * Measures how noise-like vs tonal the spectrum is\n * @param {Float32Array} coeffs - Spectral coefficients\n * @returns {number} Flatness measure (0 = tonal, 1 = noise-like)\n */\nfunction calculateSpectralFlatness(coeffs) {\n  const EPSILON = 1e-10\n  let sumLog = 0\n  let sumLinear = 0\n  let validBins = 0\n\n  for (let i = 0; i < coeffs.length; i++) {\n    const magnitude = Math.abs(coeffs[i])\n    if (magnitude > EPSILON) {\n      sumLog += Math.log(magnitude)\n      sumLinear += magnitude\n      validBins++\n    }\n  }\n\n  if (validBins === 0) return 0\n\n  const geometricMean = Math.exp(sumLog / validBins)\n  const arithmeticMean = sumLinear / validBins\n\n  return arithmeticMean > EPSILON ? geometricMean / arithmeticMean : 0\n}\n\n/**\n * Calculate ratio of high-frequency to total energy\n * High values indicate bright/harsh sounds that are perceptually important\n * @param {Float32Array} coeffs - Spectral coefficients\n * @returns {number} High-frequency energy ratio (0-1)\n */\nfunction calculateHighFrequencyRatio(coeffs) {\n  const midPoint = Math.floor(coeffs.length / 2)\n  let lowEnergy = 0\n  let highEnergy = 0\n\n  for (let i = 0; i < midPoint; i++) {\n    lowEnergy += coeffs[i] * coeffs[i]\n  }\n\n  for (let i = midPoint; i < coeffs.length; i++) {\n    highEnergy += coeffs[i] * coeffs[i]\n  }\n\n  const totalEnergy = lowEnergy + highEnergy\n  return totalEnergy > 0 ? highEnergy / totalEnergy : 0\n}\n\n/**\n * Calculate energy change between frames in dB\n * @param {Float32Array} currentCoeffs - Current frame coefficients\n * @param {Float32Array} prevCoeffs - Previous frame coefficients\n * @returns {number} Energy change in dB (positive values only)\n */\nfunction calculateEnergyChange(currentCoeffs, prevCoeffs) {\n  let currentEnergy = 0\n  let prevEnergy = 0\n\n  for (let i = 0; i < currentCoeffs.length; i++) {\n    currentEnergy += currentCoeffs[i] * currentCoeffs[i]\n    prevEnergy += prevCoeffs[i] * prevCoeffs[i]\n  }\n\n  // Avoid log of zero and ensure minimum energy\n  currentEnergy = Math.max(currentEnergy, 1e-10)\n  prevEnergy = Math.max(prevEnergy, 1e-10)\n\n  const changeDb = 10 * Math.log10(currentEnergy / prevEnergy)\n\n  // Only interested in energy increases (potential transients)\n  return Math.max(0, changeDb)\n}\n\n/**\n * Calculate unified transient score from spectral features\n * Uses perceptually-motivated combination instead of magic numbers\n * @param {Object} features - Spectral features\n * @returns {number} Transient score\n */\nfunction calculateTransientScore(features) {\n  const { spectralFlux, flatnessChange, hfChange, energyChange } = features\n\n  // Spectral flux is the primary indicator - it directly measures\n  // sudden onset of new frequency components\n  const fluxContribution = spectralFlux\n\n  // Flatness change indicates transition between tonal and noise-like\n  // signals (e.g., pitched to unpitched percussion)\n  // Scale by sqrt to reduce dominance\n  const flatnessContribution = Math.sqrt(flatnessChange)\n\n  // High-frequency changes are perceptually important but shouldn't\n  // dominate the decision. Use log scaling for perceptual relevance\n  const hfContribution = Math.log1p(hfChange * 10) / Math.log1p(10)\n\n  // Energy changes in dB are already perceptually scaled\n  // Normalize to 0-1 range assuming 30dB is a strong transient\n  const energyContribution = Math.min(energyChange / 30, 1)\n\n  // Combine features with equal perceptual importance\n  // Each contribution is normalized to roughly 0-1 range\n  return (\n    (fluxContribution +\n      flatnessContribution +\n      hfContribution +\n      energyContribution) /\n    4\n  )\n}\n","/**\n * Carta1 Audio Codec - Encoding Pipeline\n *\n * The encoding pipeline transforms PCM audio samples into ATRAC1 compressed format\n * through a series of processing stages:\n *\n * 1. QMF Analysis: Splits audio into frequency bands using quadrature mirror filters\n * 2. Block Selection: Detects transients to choose appropriate transform block sizes\n * 3. MDCT Transform: Applies Modified Discrete Cosine Transform to frequency bands\n * 4. Quantization: Performs psychoacoustic analysis, bit allocation, and quantization\n *\n * Each stage is implemented as a functional pipeline component that can be composed\n * using the pipe utility. The pipeline maintains state through a shared context\n * containing buffer pools and encoding options.\n */\n\nimport {\n  pipe,\n  throwError,\n  reverseSpectrum,\n  calculateBandOffset,\n} from '../utils.js'\nimport { qmfAnalysis } from '../transforms/qmf.js'\nimport { mdct64, mdct256, mdct512 } from '../transforms/mdct.js'\nimport { groupIntoBFUs, quantize } from '../coding/quantization.js'\nimport { allocateBits } from '../coding/bitallocation.js'\nimport { BufferPool } from '../core/buffers.js'\nimport { EncoderOptions } from '../core/options.js'\nimport {\n  WORD_LENGTH_BITS,\n  WINDOW_SHORT,\n  FFT_SIZE_LOW,\n  FFT_SIZE_MID,\n  FFT_SIZE_HIGH,\n  MDCT_SIZE_SHORT,\n  MDCT_SIZE_MID,\n  MDCT_SIZE_LONG,\n  MDCT_BAND_CONFIGS,\n  MDCT_SHORT_BLOCK_SIZE,\n  MDCT_OVERLAP_SIZE,\n} from '../core/constants.js'\nimport { performFFT, detectTransient } from '../analysis/transient.js'\n\n/**\n * QMF Analysis Stage - Splits audio into frequency bands\n *\n * Performs two-stage QMF analysis to decompose the input signal into three bands:\n * - Low band (0-5.5kHz): processed at full resolution\n * - Mid band (5.5-11kHz): processed at half resolution\n * - High band (11-22kHz): processed at quarter resolution with delay compensation\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes PCM samples\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function qmfAnalysisStage(context) {\n  const bufferPool =\n    context?.bufferPool ??\n    throwError('qmfAnalysisStage: bufferPool is required')\n  const delays = bufferPool.qmfDelays\n\n  /**\n   * Process PCM samples through QMF analysis\n   * @param {Float32Array} pcmSamples - Input PCM samples\n   * @returns {Object} Analysis result containing frequency bands\n   * @returns {Array<Float32Array>} returns.bands - Three frequency bands [low, mid, high]\n   */\n  return (pcmSamples) => {\n    const stage1 = qmfAnalysis(\n      pcmSamples,\n      delays.lowBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.lowBand = stage1.newDelay\n\n    const stage2 = qmfAnalysis(\n      stage1.lowBand,\n      delays.midBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.midBand = stage2.newDelay\n\n    const delayedHigh =\n      bufferPool.qmfWorkBuffers.highBandDelay[stage1.highBand.length]\n    delayedHigh.set(delays.highBand)\n    delayedHigh.set(stage1.highBand, delays.highBand.length)\n\n    const highBand = delayedHigh.slice(0, stage1.highBand.length)\n    delays.highBand = delayedHigh.slice(stage1.highBand.length)\n\n    return {\n      bands: [stage2.lowBand, stage2.highBand, highBand],\n    }\n  }\n}\n\n/**\n * Block Selection Stage - Detects transients to choose transform block sizes\n *\n * Analyzes frequency bands using FFT and transient detection to determine\n * appropriate MDCT block sizes. Short blocks are used for transient signals\n * to prevent pre-echo artifacts, while long blocks provide better frequency\n * resolution for steady-state signals.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes QMF analysis results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function blockSelectorStage(context) {\n  const bufferPool =\n    context?.bufferPool ??\n    throwError('blockSelectorStage: bufferPool is required')\n  const options =\n    context?.options ?? throwError('blockSelectorStage: options is required')\n\n  /**\n   * Analyze frequency bands and determine block modes\n   * @param {Object} input - QMF analysis results\n   * @param {Array<Float32Array>} input.bands - Three frequency bands [low, mid, high]\n   * @returns {Object} Block selection results\n   * @returns {Array<Float32Array>} returns.bands - Original frequency bands\n   * @returns {Array<number>} returns.blockModes - Block modes for each band\n   */\n  return (input) => {\n    const { bands } = input\n    const fftSizes = [FFT_SIZE_LOW, FFT_SIZE_MID, FFT_SIZE_HIGH]\n    const blockModes = bands.map((bandSamples, bandIndex) => {\n      const coeffs = performFFT(bandSamples, fftSizes[bandIndex])\n      const transient = detectTransient(\n        coeffs,\n        bufferPool.transientDetection[bandIndex],\n        options.transientThresholdLow\n      )\n      bufferPool.transientDetection[bandIndex] = coeffs\n      return transient * Math.max(bandIndex + 1, 2)\n    })\n\n    return {\n      bands,\n      blockModes,\n    }\n  }\n}\n\n/**\n * MDCT Transform Stage - Applies Modified Discrete Cosine Transform\n *\n * Transforms time-domain samples to frequency-domain coefficients using MDCT.\n * The transform size depends on the block mode determined by transient detection:\n * - Long blocks: 512/256 samples for better frequency resolution\n * - Short blocks: 64 samples for better time resolution\n *\n * Applies windowing and overlap-add processing to prevent blocking artifacts.\n * Mid and high bands are spectrally reversed to match ATRAC1 format.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes block selection results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function mdctStage(context) {\n  const bufferPool =\n    context?.bufferPool ?? throwError('mdctStage: bufferPool is required')\n  const overlapBuffers = bufferPool.mdctOverlap\n\n  // Transform function mapping for each band\n  const TRANSFORM_FUNCS = [mdct256, mdct256, mdct512]\n\n  /**\n   * Transform a single frequency band using MDCT.\n   * @param {Float32Array} samples\n   * @param {number} bandIndex\n   * @param {number} blockMode\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @returns {Float32Array}\n   */\n  function transformBand(\n    samples,\n    bandIndex,\n    blockMode,\n    overlapBuffer,\n    bufferPool\n  ) {\n    const config = MDCT_BAND_CONFIGS[bandIndex]\n    const transformFunc = TRANSFORM_FUNCS[bandIndex]\n    const isLongBlock = blockMode === 0\n\n    if (isLongBlock) {\n      return transformLongBlock(\n        samples,\n        bandIndex,\n        config,\n        transformFunc,\n        overlapBuffer,\n        bufferPool\n      )\n    } else {\n      return transformShortBlocks(\n        samples,\n        bandIndex,\n        config,\n        overlapBuffer,\n        bufferPool\n      )\n    }\n  }\n\n  /**\n   * Transform using long block (better frequency resolution).\n   * @param {Float32Array} samples\n   * @param {number} bandIndex\n   * @param {Object} config\n   * @param {Object} transformFunc\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @returns {Float32Array}\n   */\n  function transformLongBlock(\n    samples,\n    bandIndex,\n    config,\n    transformFunc,\n    overlapBuffer,\n    bufferPool\n  ) {\n    const mdctSize = bandIndex === 2 ? MDCT_SIZE_LONG : MDCT_SIZE_MID\n    const mdctInput = bufferPool.transformBuffers[mdctSize]\n    mdctInput.fill(0)\n\n    // Apply overlap from previous frame\n    mdctInput.set(overlapBuffer, config.windowStart)\n\n    // Window and save tail for next frame\n    applyTailWindowing(samples, overlapBuffer, config.size)\n\n    // Add current frame samples\n    mdctInput.set(samples, config.windowStart + MDCT_OVERLAP_SIZE)\n\n    // Transform\n    let spectrum = transformFunc.transform(mdctInput, bufferPool.mdctBuffers)\n\n    // Apply spectral reversal for mid/high bands\n    if (bandIndex > 0) {\n      spectrum = reverseSpectrum(spectrum, bufferPool.reversalBuffers)\n    }\n\n    return spectrum\n  }\n\n  /**\n   * Transform using short blocks (better time resolution for transients).\n   * @param {Float32Array} samples\n   * @param {number} bandIndex\n   * @param {Object} config\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @returns {Float32Array}\n   */\n  function transformShortBlocks(\n    samples,\n    bandIndex,\n    config,\n    overlapBuffer,\n    bufferPool\n  ) {\n    const numBlocks = 1 << (config.size === 256 ? 3 : 2) // 8 for band 2, 4 for bands 0-1\n    const output = new Float32Array(config.size)\n\n    for (let block = 0; block < numBlocks; block++) {\n      const blockStart = block * MDCT_SHORT_BLOCK_SIZE\n      const blockSamples = samples.subarray(\n        blockStart,\n        blockStart + MDCT_SHORT_BLOCK_SIZE\n      )\n\n      // Prepare MDCT input\n      const mdctInput = bufferPool.transformBuffers[MDCT_SIZE_SHORT]\n      mdctInput.fill(0)\n      mdctInput.set(overlapBuffer, 0)\n\n      // Window and save tail\n      applyTailWindowing(blockSamples, overlapBuffer, MDCT_SHORT_BLOCK_SIZE)\n      mdctInput.set(blockSamples, MDCT_OVERLAP_SIZE)\n\n      // Transform\n      let spectrum = mdct64.transform(mdctInput, bufferPool.mdctBuffers)\n\n      // Apply spectral reversal for mid/high bands\n      if (bandIndex > 0) {\n        spectrum = reverseSpectrum(spectrum, bufferPool.reversalBuffers)\n      }\n\n      output.set(spectrum, blockStart)\n    }\n\n    return output\n  }\n\n  function applyTailWindowing(samples, overlapBuffer, blockSize) {\n    const tailStart = blockSize - MDCT_OVERLAP_SIZE\n    for (let i = 0; i < MDCT_OVERLAP_SIZE; i++) {\n      const tailValue = samples[tailStart + i]\n      overlapBuffer[i] = WINDOW_SHORT[i] * tailValue\n      samples[tailStart + i] = tailValue * WINDOW_SHORT[31 - i]\n    }\n  }\n\n  /**\n   * Transform frequency bands using MDCT\n   * @param {Object} input - Block selection results\n   * @param {Array<Float32Array>} input.bands - Three frequency bands\n   * @param {Array<number>} input.blockModes - Block modes for each band\n   * @param {any} input.originalFrame - Original frame data (passed through)\n   * @returns {Object} MDCT transform results\n   * @returns {Array<Float32Array>} returns.bands - Original frequency bands\n   * @returns {Float32Array} returns.coefficients - MDCT coefficients (512 samples)\n   * @returns {Array<number>} returns.blockModes - Block modes\n   * @returns {any} returns.originalFrame - Original frame data\n   */\n  return (input) => {\n    const { bands, blockModes, originalFrame } = input\n    const coefficients = new Float32Array(512)\n\n    bands.forEach((bandSamples, bandIndex) => {\n      const transformed = transformBand(\n        bandSamples,\n        bandIndex,\n        blockModes[bandIndex],\n        overlapBuffers[bandIndex],\n        bufferPool\n      )\n\n      const offset = calculateBandOffset(bandIndex)\n      coefficients.set(transformed, offset)\n    })\n\n    return { bands, coefficients, blockModes, originalFrame }\n  }\n}\n\n/**\n * Quantization Stage - Performs RDO bit allocation and quantization\n *\n * The final encoding stage that:\n * 1. Groups coefficients into Block Floating Units (BFUs)\n * 2. Allocates bits using Rate-Distortion Optimization (RDO)\n * 3. Quantizes coefficients using scale factors and word lengths\n *\n * Uses RDO to optimize the allocation of bits for maximum quality\n * within the available bit budget.\n *\n * @returns {Function} Stage function that processes MDCT results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function quantizationStage() {\n  /**\n   * Perform RDO bit allocation and quantization\n   * @param {Object} input - MDCT transform results\n   * @param {Float32Array} input.coefficients - MDCT coefficients\n   * @param {Array<number>} input.blockModes - Block modes\n   * @returns {Object} Quantization results ready for bitstream encoding\n   * @returns {number} returns.nBfu - Number of active BFUs\n   * @returns {Int32Array} returns.scaleFactorIndices - Scale factor indices for each BFU\n   * @returns {Int32Array} returns.wordLengthIndices - Word length indices for each BFU\n   * @returns {Array<Int32Array>} returns.quantizedCoefficients - Quantized coefficient data\n   * @returns {Array<number>} returns.blockModes - Block modes for each band\n   */\n  return (input) => {\n    const { coefficients, blockModes } = input\n\n    const { bfuData, bfuSizes, bfuCount } = groupIntoBFUs(\n      coefficients,\n      blockModes\n    )\n\n    const {\n      bfuCount: selectedBfuCount,\n      allocation,\n      scaleFactorIndices,\n    } = allocateBits(bfuData, bfuSizes, bfuCount)\n\n    const quantizedCoefficients = []\n\n    for (let bfu = 0; bfu < selectedBfuCount; bfu++) {\n      const data = bfuData[bfu].subarray(0, bfuSizes[bfu])\n      const wordLength = allocation[bfu]\n      const bitsPerSample = WORD_LENGTH_BITS[wordLength]\n      const quantized = quantize(data, scaleFactorIndices[bfu], bitsPerSample)\n      quantizedCoefficients.push(quantized)\n    }\n\n    return {\n      nBfu: selectedBfuCount,\n      scaleFactorIndices,\n      wordLengthIndices: allocation.slice(0, selectedBfuCount),\n      quantizedCoefficients,\n      blockModes,\n    }\n  }\n}\n\n/**\n * Create ATRAC1 encoding pipeline\n *\n * Constructs a complete encoding pipeline that transforms PCM audio samples\n * into ATRAC1 compressed format. The pipeline processes audio through QMF\n * analysis, transient detection, MDCT transform, and psychoacoustic quantization.\n *\n * The returned function can be called repeatedly to encode audio frames,\n * maintaining state through the shared buffer pool for efficient processing.\n *\n * @param {EncoderOptions} [options=new EncoderOptions()] - Encoding configuration\n * @param {BufferPool} [bufferPool=new BufferPool()] - Shared buffer pool for state\n * @returns {Function} Encoding pipeline function that processes PCM samples\n *\n * @example\n * const encoder = encode(new EncoderOptions(), new BufferPool())\n * const result = encoder(pcmSamples) // Returns quantized ATRAC1 frame data\n */\nexport function encode(\n  options = new EncoderOptions(),\n  bufferPool = new BufferPool()\n) {\n  const context = { options, bufferPool }\n  return pipe(\n    context,\n    qmfAnalysisStage,\n    blockSelectorStage,\n    mdctStage,\n    quantizationStage\n  )\n}\n","/**\n * Carta1 Audio Codec - Decoding Pipeline\n *\n * The decoding pipeline transforms ATRAC1 compressed format back into PCM audio samples\n * through a series of processing stages:\n *\n * 1. Dequantization: Reconstructs MDCT coefficients from quantized data using scale factors\n * 2. Inverse MDCT: Applies Inverse Modified Discrete Cosine Transform to frequency bands\n * 3. QMF Synthesis: Reconstructs full-spectrum audio using quadrature mirror filters\n *\n * Each stage is implemented as a functional pipeline component that can be composed\n * using the pipe utility. The pipeline maintains state through a shared context\n * containing buffer pools and decoding options.\n */\n\nimport {\n  pipe,\n  throwError,\n  reverseSpectrum,\n  extractBandCoefficients,\n} from '../utils.js'\nimport { qmfSynthesis } from '../transforms/qmf.js'\nimport { imdct64, imdct256, imdct512, overlapAdd } from '../transforms/mdct.js'\nimport { dequantize } from '../coding/quantization.js'\nimport { BufferPool } from '../core/buffers.js'\nimport {\n  BFU_BAND_BOUNDARIES,\n  BFU_START_LONG,\n  BFU_START_SHORT,\n  WORD_LENGTH_BITS,\n  WINDOW_SHORT,\n  MDCT_SIZE_LONG,\n  MDCT_BAND_CONFIGS,\n  MDCT_SHORT_BLOCK_SIZE,\n  MDCT_TAIL_WINDOW_SIZE,\n} from '../core/constants.js'\n\n/**\n * Dequantization Stage - Reconstructs MDCT coefficients from quantized data\n *\n * Reverses the quantization process by:\n * - Reconstructing coefficient values using scale factors and word lengths\n * - Organizing coefficients back into Block Floating Units (BFUs)\n * - Preparing coefficient array for inverse MDCT transform\n *\n * Uses the same BFU organization as the encoder but reconstructs the original\n * coefficient values from the quantized representation.\n *\n * @returns {Function} Stage function that processes quantized frame data\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function dequantizationStage() {\n  /**\n   * Reconstruct MDCT coefficients from quantized data\n   * @param {Object} frameData - Quantized frame data\n   * @param {number} frameData.nBfu - Number of active BFUs\n   * @param {Int32Array} frameData.scaleFactorIndices - Scale factor indices for each BFU\n   * @param {Int32Array} frameData.wordLengthIndices - Word length indices for each BFU\n   * @param {Array<Int32Array>} frameData.quantizedCoefficients - Quantized coefficient data\n   * @param {Array<number>} frameData.blockModes - Block modes for each band\n   * @returns {Object} Dequantization results\n   * @returns {Float32Array} returns.coefficients - Reconstructed MDCT coefficients (512 samples)\n   * @returns {Array<number>} returns.blockModes - Block modes for each band\n   */\n  return (frameData) => {\n    const coefficients = new Float32Array(512)\n    const {\n      nBfu,\n      scaleFactorIndices,\n      wordLengthIndices,\n      quantizedCoefficients,\n      blockModes,\n    } = frameData\n\n    for (let bfu = 0; bfu < nBfu; bfu++) {\n      const bitsPerSample = WORD_LENGTH_BITS[wordLengthIndices[bfu]]\n\n      let band = 0\n      if (bfu >= BFU_BAND_BOUNDARIES[0]) band = 1\n      if (bfu >= BFU_BAND_BOUNDARIES[1]) band = 2\n\n      const isLongBlock = blockModes[band] === 0\n      const position = isLongBlock ? BFU_START_LONG[bfu] : BFU_START_SHORT[bfu]\n\n      if (bitsPerSample > 0) {\n        const quantized = quantizedCoefficients[bfu]\n        const dequantized = dequantize(\n          quantized,\n          scaleFactorIndices[bfu],\n          bitsPerSample\n        )\n        coefficients.set(dequantized, position)\n      }\n    }\n\n    return { coefficients, blockModes }\n  }\n}\n\n/**\n * Inverse MDCT Stage - Transforms frequency coefficients back to time domain\n *\n * Applies Inverse Modified Discrete Cosine Transform to reconstruct time-domain\n * samples from frequency coefficients. The transform size depends on block modes:\n * - Long blocks: 512/256 samples for steady-state signals\n * - Short blocks: 64 samples for transient signals\n *\n * Applies windowing and overlap-add processing to prevent blocking artifacts.\n * Mid and high bands are spectrally unreversed from ATRAC1 format.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes dequantization results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function imdctStage(context) {\n  const bufferPool =\n    context?.bufferPool ?? throwError('imdctStage: bufferPool is required')\n  const overlapBuffers = bufferPool.imdctOverlap\n\n  // Transform function mapping for each band\n  const TRANSFORM_FUNCS = [imdct256, imdct256, imdct512]\n\n  /**\n   * Inverse transform a single frequency band.\n   * @param {Float32Array} coeffs\n   * @param {number} bandIndex\n   * @param {number} blockMode\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @param {Object} config\n   * @returns {Float32Array}\n   */\n  function transformBand(\n    coeffs,\n    bandIndex,\n    blockMode,\n    overlapBuffer,\n    bufferPool,\n    config\n  ) {\n    const isLongBlock = blockMode === 0\n    const transformFunc = TRANSFORM_FUNCS[bandIndex]\n\n    if (isLongBlock) {\n      return inverseLongBlock(\n        coeffs,\n        bandIndex,\n        overlapBuffer,\n        bufferPool,\n        config,\n        transformFunc\n      )\n    } else {\n      return inverseShortBlocks(\n        coeffs,\n        bandIndex,\n        overlapBuffer,\n        bufferPool,\n        config\n      )\n    }\n  }\n\n  /**\n   * Inverse transform using long block.\n   * @param {Float32Array} coeffs\n   * @param {number} bandIndex\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @param {Object} config\n   * @param {Object} transformFunc\n   * @returns {Float32Array}\n   */\n  function inverseLongBlock(\n    coeffs,\n    bandIndex,\n    overlapBuffer,\n    bufferPool,\n    config,\n    transformFunc\n  ) {\n    // Unreverse spectral reversal for mid/high bands\n    let blockSpecs = coeffs\n    if (bandIndex > 0) {\n      blockSpecs = reverseSpectrum(coeffs, bufferPool.reversalBuffers)\n    }\n\n    // Apply inverse transform\n    const inv = transformFunc.transform(blockSpecs, bufferPool.mdctBuffers)\n    const invStart = inv.length / 4\n\n    // Prepare output buffer\n    const invBuf = bufferPool.transformBuffers[MDCT_SIZE_LONG]\n    invBuf.fill(0)\n\n    // Copy transformed samples\n    for (let i = 0; i < config.size; i++) {\n      invBuf[i] = inv[invStart + i]\n    }\n\n    // Handle overlap-add with previous frame\n    const prevBuf = overlapBuffer.slice(\n      config.size * 2 - MDCT_TAIL_WINDOW_SIZE,\n      config.size * 2\n    )\n    const dstPart = overlapAdd(\n      prevBuf.slice(0, MDCT_TAIL_WINDOW_SIZE),\n      invBuf.slice(0, MDCT_TAIL_WINDOW_SIZE),\n      WINDOW_SHORT\n    )\n\n    overlapBuffer.set(dstPart, 0)\n\n    // Save tail for next frame\n    for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n      prevBuf[i] = invBuf[MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    // Copy main samples\n    const copyLen = bandIndex === 2 ? 240 : 112\n    for (let i = 0; i < copyLen; i++) {\n      overlapBuffer[32 + i] = invBuf[MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    // Copy final samples\n    for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n      overlapBuffer[config.size * 2 - MDCT_TAIL_WINDOW_SIZE + i] =\n        invBuf[config.size - MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    return overlapBuffer.slice(0, config.size)\n  }\n\n  /**\n   * Inverse transform using short blocks.\n   * @param {Float32Array} coeffs\n   * @param {number} bandIndex\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @param {Object} config\n   * @returns {Float32Array}\n   */\n  function inverseShortBlocks(\n    coeffs,\n    bandIndex,\n    overlapBuffer,\n    bufferPool,\n    config\n  ) {\n    const numBlocks = 1 << (config.size === 256 ? 3 : 2)\n    const invBuf = bufferPool.transformBuffers[MDCT_SIZE_LONG]\n    invBuf.fill(0)\n\n    const prevBuf = overlapBuffer.slice(\n      config.size * 2 - MDCT_TAIL_WINDOW_SIZE,\n      config.size * 2\n    )\n\n    let start = 0\n    let pos = 0\n\n    for (let block = 0; block < numBlocks; block++) {\n      // Extract block coefficients\n      let blockSpecs = coeffs.slice(pos, pos + MDCT_SHORT_BLOCK_SIZE)\n\n      // Unreverse spectral reversal for mid/high bands\n      if (bandIndex > 0) {\n        blockSpecs = reverseSpectrum(blockSpecs, bufferPool.reversalBuffers)\n      }\n\n      // Apply inverse transform\n      const inv = imdct64.transform(blockSpecs, bufferPool.mdctBuffers)\n      const invStart = inv.length / 4\n\n      // Copy to intermediate buffer\n      for (let i = 0; i < MDCT_SHORT_BLOCK_SIZE; i++) {\n        invBuf[start + i] = inv[invStart + i]\n      }\n\n      // Overlap-add with previous block\n      const dstPart = overlapAdd(\n        prevBuf.slice(0, MDCT_TAIL_WINDOW_SIZE),\n        invBuf.slice(start, start + MDCT_TAIL_WINDOW_SIZE),\n        WINDOW_SHORT\n      )\n\n      overlapBuffer.set(dstPart, start)\n\n      // Save tail for next block\n      for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n        prevBuf[i] = invBuf[start + MDCT_TAIL_WINDOW_SIZE + i]\n      }\n\n      start += MDCT_SHORT_BLOCK_SIZE\n      pos += MDCT_SHORT_BLOCK_SIZE\n    }\n\n    // Copy final tail samples\n    for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n      overlapBuffer[config.size * 2 - MDCT_TAIL_WINDOW_SIZE + i] =\n        invBuf[config.size - MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    return overlapBuffer.slice(0, config.size)\n  }\n\n  /**\n   * Transform coefficients back to time domain using inverse MDCT\n   * @param {Object} input - Dequantization results\n   * @param {Float32Array} input.coefficients - MDCT coefficients (512 samples)\n   * @param {Array<number>} input.blockModes - Block modes for each band\n   * @returns {Array<Float32Array>} Three reconstructed frequency bands [low, mid, high]\n   */\n  return (input) => {\n    const { coefficients, blockModes } = input\n\n    return MDCT_BAND_CONFIGS.map((config, bandIndex) => {\n      const bandCoeffs = extractBandCoefficients(coefficients, bandIndex)\n      return transformBand(\n        bandCoeffs,\n        bandIndex,\n        blockModes[bandIndex],\n        overlapBuffers[bandIndex],\n        bufferPool,\n        config\n      )\n    })\n  }\n}\n\n/**\n * QMF Synthesis Stage - Reconstructs full-spectrum audio from frequency bands\n *\n * Performs two-stage QMF synthesis to reconstruct the full-spectrum signal\n * from three frequency bands:\n * - Low band (0-5.5kHz): processed at full resolution\n * - Mid band (5.5-11kHz): processed at half resolution\n * - High band (11-22kHz): processed at quarter resolution with delay compensation\n *\n * This is the inverse operation of the QMF analysis performed during encoding,\n * combining the separate frequency bands back into a single wideband signal.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes inverse MDCT results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function qmfSynthesisStage(context) {\n  const bufferPool =\n    context?.bufferPool ??\n    throwError('qmfSynthesisStage: bufferPool is required')\n  const delays = bufferPool.qmfDelays\n\n  /**\n   * Reconstruct full-spectrum PCM samples from frequency bands\n   * @param {Array<Float32Array>} bands - Three frequency bands [low, mid, high]\n   * @returns {Float32Array} Reconstructed PCM samples\n   */\n  return (bands) => {\n    // Apply high band delay compensation\n    const delayedHigh = bufferPool.qmfWorkBuffers.highBandDelay[bands[2].length]\n    delayedHigh.set(delays.highBand)\n    delayedHigh.set(bands[2], delays.highBand.length)\n\n    const highBand = delayedHigh.slice(0, bands[0].length * 2)\n    delays.highBand = delayedHigh.slice(bands[0].length * 2)\n\n    // Second stage synthesis: combine low and mid bands\n    const stage2 = qmfSynthesis(\n      bands[0],\n      bands[1],\n      delays.midBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.midBand = stage2.newDelay\n\n    // First stage synthesis: combine result with high band\n    const stage1 = qmfSynthesis(\n      stage2.output,\n      highBand,\n      delays.lowBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.lowBand = stage1.newDelay\n\n    return stage1.output\n  }\n}\n\n/**\n * Create ATRAC1 decoding pipeline\n *\n * Constructs a complete decoding pipeline that transforms ATRAC1 compressed format\n * back into PCM audio samples. The pipeline processes quantized data through\n * dequantization, inverse MDCT transform, and QMF synthesis.\n *\n * The returned function can be called repeatedly to decode audio frames,\n * maintaining state through the shared buffer pool for efficient processing.\n *\n * @param {BufferPool} [bufferPool=new BufferPool()] - Shared buffer pool for state\n * @returns {Function} Decoding pipeline function that processes quantized frame data\n *\n * @example\n * const decoder = decode(new BufferPool())\n * const result = decoder(quantizedFrame) // Returns reconstructed PCM samples\n */\nexport function decode(bufferPool = new BufferPool()) {\n  const context = { bufferPool }\n  return pipe(context, dequantizationStage, imdctStage, qmfSynthesisStage)\n}\n","/**\n * Carta1 Audio Codec - Audio Processing Module\n *\n * High-level audio processing interface that provides stream-based encoding and decoding\n * for ATRAC1 audio format. Handles both mono and stereo audio processing with proper\n * delay compensation and frame management.\n *\n * Key features:\n * - Stream-based processing for large audio files\n * - Automatic delay compensation for codec processing\n * - Support for mono and stereo audio\n * - Progress tracking callbacks\n * - AEA file format creation and parsing\n * - WAV file generation from decoded audio\n */\n\nimport { encode } from '../pipeline/encoder.js'\nimport { decode } from '../pipeline/decoder.js'\nimport { EncoderOptions } from '../core/options.js'\nimport {\n  SAMPLES_PER_FRAME,\n  SAMPLE_RATE,\n  AEA_HEADER_SIZE,\n  SOUND_UNIT_SIZE,\n  WAV_HEADER_SIZE,\n  WAV_BYTES_PER_SAMPLE,\n  WAV_BITS_PER_SAMPLE,\n  WAV_PCM_MAX_POSITIVE,\n  WAV_PCM_MAX_NEGATIVE,\n  WAV_DATA_OFFSET,\n} from '../core/constants.js'\nimport { serializeFrame, deserializeFrame, AeaFile } from './serialization.js'\n\n/**\n * Audio processing utility class providing high-level encoding and decoding operations\n */\nexport class AudioProcessor {\n  /**\n   * Encodes a stream of audio frames to ATRAC1 format\n   * @param {AsyncIterable<Float32Array>|AsyncIterable<[Float32Array, Float32Array]>} audioFrames - Audio frame stream\n   * @param {Object} [options={}] - Encoding options\n   * @param {number} [options.channelCount=1] - Number of audio channels (1 or 2)\n   * @param {Function} [options.onProgress] - Progress callback function\n   * @param {EncoderOptions} [options.encoderOptions] - Encoder configuration options\n   * @returns {AsyncGenerator<Object>} Stream of encoded frame data\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static async *encodeStream(audioFrames, options = {}) {\n    const { channelCount = 1, onProgress, encoderOptions } = options\n\n    if (channelCount === 1) {\n      yield* AudioProcessor._encodeMonoStream(\n        audioFrames,\n        onProgress,\n        encoderOptions\n      )\n    } else if (channelCount === 2) {\n      yield* AudioProcessor._encodeStereoStream(\n        audioFrames,\n        onProgress,\n        encoderOptions\n      )\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Internal method for encoding mono audio streams\n   * @param {AsyncIterable<Float32Array>} audioFrames - Mono audio frame stream\n   * @param {Function} onProgress - Progress callback function\n   * @param {EncoderOptions} encoderOptions - Encoder configuration\n   * @returns {AsyncGenerator<Object>} Stream of encoded frame data\n   * @private\n   */\n  static async *_encodeMonoStream(audioFrames, onProgress, encoderOptions) {\n    const options = encoderOptions || new EncoderOptions()\n    const encoder = encode(options)\n    let frameIndex = 0\n    for await (const frame of audioFrames) {\n      const result = encoder(frame)\n      yield result\n\n      if (onProgress) {\n        onProgress(frameIndex++)\n      }\n    }\n  }\n\n  /**\n   * Internal method for encoding stereo audio streams\n   * @param {AsyncIterable<[Float32Array, Float32Array]>} audioFrames - Stereo audio frame stream\n   * @param {Function} onProgress - Progress callback function\n   * @param {EncoderOptions} encoderOptions - Encoder configuration\n   * @returns {AsyncGenerator<Object>} Stream of encoded frame data (left then right)\n   * @private\n   */\n  static async *_encodeStereoStream(audioFrames, onProgress, encoderOptions) {\n    const options = encoderOptions || new EncoderOptions()\n    const leftEncoder = encode(options)\n    const rightEncoder = encode(options)\n    let frameIndex = 0\n\n    for await (const [leftFrame, rightFrame] of audioFrames) {\n      const leftResult = leftEncoder(leftFrame)\n      const rightResult = rightEncoder(rightFrame)\n\n      yield leftResult\n      yield rightResult\n\n      if (onProgress) {\n        onProgress(frameIndex++)\n      }\n    }\n  }\n\n  /**\n   * Decodes a stream of ATRAC1 encoded frames back to PCM audio\n   * @param {AsyncIterable<Object>} encodedFrames - Stream of encoded frame data\n   * @param {Object} [options={}] - Decoding options\n   * @param {number} [options.channelCount=1] - Number of audio channels (1 or 2)\n   * @param {Function} [options.onProgress] - Progress callback function\n   * @returns {AsyncGenerator<Float32Array>|AsyncGenerator<[Float32Array, Float32Array]>} Stream of decoded PCM frames\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static async *decodeStream(encodedFrames, options = {}) {\n    const { channelCount = 1, onProgress } = options\n\n    if (channelCount === 1) {\n      yield* AudioProcessor._decodeMonoStream(encodedFrames, onProgress)\n    } else if (channelCount === 2) {\n      yield* AudioProcessor._decodeStereoStream(encodedFrames, onProgress)\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Internal method for decoding mono audio streams\n   * @param {AsyncIterable<Object>} encodedFrames - Mono encoded frame stream\n   * @param {Function} onProgress - Progress callback function\n   * @returns {AsyncGenerator<Float32Array>} Stream of decoded mono PCM frames\n   * @private\n   */\n  static async *_decodeMonoStream(encodedFrames, onProgress) {\n    const decoder = decode()\n    let frameIndex = 0\n\n    // Create raw PCM frame stream\n    const rawPcmFrames = async function* () {\n      for await (const frame of encodedFrames) {\n        const result = decoder(frame)\n        yield result\n\n        if (onProgress) {\n          onProgress(frameIndex++)\n        }\n      }\n    }\n\n    // Always apply delay compensation\n    yield* rawPcmFrames()\n  }\n\n  /**\n   * Internal method for decoding stereo audio streams\n   * @param {AsyncIterable<Object>} encodedFrames - Stereo encoded frame stream (interleaved left/right)\n   * @param {Function} onProgress - Progress callback function\n   * @returns {AsyncGenerator<[Float32Array, Float32Array]>} Stream of decoded stereo PCM frame pairs\n   * @private\n   */\n  static async *_decodeStereoStream(encodedFrames, onProgress) {\n    const leftDecoder = decode()\n    const rightDecoder = decode()\n    let frameIndex = 0\n\n    const rawPcmFrames = async function* () {\n      let leftFrame = null\n      let isLeftChannel = true\n\n      for await (const frame of encodedFrames) {\n        if (isLeftChannel) {\n          leftFrame = frame\n        } else {\n          const rightFrame = frame\n\n          // Decode and yield the pair\n          const leftResult = leftDecoder(leftFrame)\n          const rightResult = rightDecoder(rightFrame)\n\n          yield [leftResult, rightResult]\n\n          if (onProgress) {\n            onProgress(frameIndex++)\n          }\n        }\n\n        isLeftChannel = !isLeftChannel\n      }\n\n      if (!isLeftChannel && leftFrame) {\n        const rightFrame = AudioProcessor._createDummyFrame()\n        const leftResult = leftDecoder(leftFrame)\n        const rightResult = rightDecoder(rightFrame)\n\n        yield [leftResult, rightResult]\n\n        if (onProgress) {\n          onProgress(frameIndex++)\n        }\n      }\n    }\n\n    // Always apply stereo delay compensation\n    yield* rawPcmFrames()\n  }\n\n  /**\n   * Converts channel buffer arrays to frame-based stream format\n   * @param {Float32Array[]} buffers - Array of channel buffers (1 for mono, 2 for stereo)\n   * @param {number} [frameSize=SAMPLES_PER_FRAME] - Frame size in samples\n   * @returns {Generator<Float32Array>|Generator<[Float32Array, Float32Array]>} Stream of audio frames\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static *frameBufferToFrames(buffers, frameSize = SAMPLES_PER_FRAME) {\n    const channelCount = buffers.length\n\n    if (channelCount === 1) {\n      const [left] = buffers\n      for (let i = 0; i < left.length; i += frameSize) {\n        const frame = new Float32Array(frameSize)\n        const end = Math.min(i + frameSize, left.length)\n        for (let j = 0; j < end - i; j++) {\n          frame[j] = left[i + j]\n        }\n        yield frame\n      }\n    } else if (channelCount === 2) {\n      const [left, right] = buffers\n      const maxLength = Math.max(left.length, right.length)\n\n      for (let i = 0; i < maxLength; i += frameSize) {\n        const leftFrame = new Float32Array(frameSize)\n        const rightFrame = new Float32Array(frameSize)\n\n        const end = Math.min(i + frameSize, maxLength)\n\n        for (let j = 0; j < end - i; j++) {\n          leftFrame[j] = i + j < left.length ? left[i + j] : 0\n          rightFrame[j] = i + j < right.length ? right[i + j] : 0\n        }\n\n        yield [leftFrame, rightFrame]\n      }\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Collects all frames from a stream into an array\n   * @param {AsyncIterable} frameStream - Stream of frames to collect\n   * @returns {Promise<Array>} Array of all frames from the stream\n   */\n  static async collectFrames(frameStream) {\n    const frames = []\n    for await (const frame of frameStream) {\n      frames.push(frame)\n    }\n    return frames\n  }\n\n  /**\n   * Creates a dummy frame for padding purposes\n   * @returns {Object} Empty frame structure\n   * @private\n   */\n  static _createDummyFrame() {\n    return {\n      nBfu: 0,\n      blockModes: [0, 0, 0],\n      scaleFactorIndices: new Int32Array(0),\n      wordLengthIndices: new Int32Array(0),\n      quantizedCoefficients: [],\n    }\n  }\n\n  /**\n   * Creates an AEA file blob from encoded frames\n   * @param {AsyncIterable<Object>} encodedFrames - Stream of encoded frame data\n   * @param {Object} [options={}] - AEA file options\n   * @param {string} [options.title='encoded by atrac1.js'] - File title\n   * @param {number} [options.channelCount=1] - Number of audio channels\n   * @returns {Promise<Blob>} AEA file blob\n   */\n  static async createAeaBlob(encodedFrames, options = {}) {\n    const { title = 'encoded by atrac1.js', channelCount = 1 } = options\n\n    const frames = []\n    for await (const frame of encodedFrames) {\n      frames.push(serializeFrame(frame))\n    }\n\n    const header = AeaFile.createHeader(title, frames.length, channelCount)\n    const totalSize =\n      header.length + frames.reduce((sum, frame) => sum + frame.length, 0)\n\n    const buffer = new Uint8Array(totalSize)\n    buffer.set(header, 0)\n\n    let offset = header.length\n    for (const frame of frames) {\n      buffer.set(frame, offset)\n      offset += frame.length\n    }\n\n    return new Blob([buffer], { type: 'application/octet-stream' })\n  }\n\n  /**\n   * Creates a WAV file blob from PCM audio frames\n   * @param {Float32Array[]|Float32Array|[Float32Array, Float32Array][]} pcmFrames - PCM audio frames\n   * @param {number} [channelCount=1] - Number of audio channels\n   * @param {number} [sampleRate=SAMPLE_RATE] - Audio sample rate\n   * @returns {Blob} WAV file blob\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static createWavBlob(pcmFrames, channelCount = 1, sampleRate = SAMPLE_RATE) {\n    const frames = Array.isArray(pcmFrames) ? pcmFrames : [pcmFrames]\n\n    if (channelCount === 1) {\n      return AudioProcessor._createMonoWavBlob(frames, sampleRate)\n    } else if (channelCount === 2) {\n      return AudioProcessor._createStereoWavBlob(frames, sampleRate)\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Creates a mono WAV file blob from PCM frames\n   * @param {Float32Array[]} frames - Array of mono PCM frames\n   * @param {number} sampleRate - Audio sample rate\n   * @returns {Blob} Mono WAV file blob\n   * @private\n   */\n  static _createMonoWavBlob(frames, sampleRate) {\n    const totalSamples = frames.reduce((sum, frame) => sum + frame.length, 0)\n    const buffer = new ArrayBuffer(\n      WAV_HEADER_SIZE + totalSamples * WAV_BYTES_PER_SAMPLE\n    )\n    const view = new DataView(buffer)\n\n    // WAV header\n    AudioProcessor._writeWavHeader(view, totalSamples, 1, sampleRate)\n\n    // PCM data using DataView for consistency\n    let offset = WAV_HEADER_SIZE\n    for (const frame of frames) {\n      for (let i = 0; i < frame.length; i++) {\n        const sample = Math.max(-1, Math.min(1, frame[i]))\n        view.setInt16(\n          offset,\n          sample < 0\n            ? sample * WAV_PCM_MAX_NEGATIVE\n            : sample * WAV_PCM_MAX_POSITIVE,\n          true\n        )\n        offset += WAV_BYTES_PER_SAMPLE\n      }\n    }\n\n    return new Blob([buffer], { type: 'audio/wav' })\n  }\n\n  /**\n   * Creates a stereo WAV file blob from PCM frame pairs\n   * @param {[Float32Array, Float32Array][]} framePairs - Array of stereo PCM frame pairs\n   * @param {number} sampleRate - Audio sample rate\n   * @returns {Blob} Stereo WAV file blob\n   * @private\n   */\n  static _createStereoWavBlob(framePairs, sampleRate) {\n    const totalSamples = framePairs.reduce(\n      (sum, [left, right]) => sum + Math.max(left.length, right.length),\n      0\n    )\n\n    const buffer = new ArrayBuffer(\n      WAV_HEADER_SIZE + totalSamples * WAV_BYTES_PER_SAMPLE * 2\n    )\n    const view = new DataView(buffer)\n\n    // WAV header\n    AudioProcessor._writeWavHeader(view, totalSamples, 2, sampleRate)\n\n    // Interleaved PCM data using DataView for consistency\n    let offset = WAV_HEADER_SIZE\n    for (const [leftFrame, rightFrame] of framePairs) {\n      const frameLength = Math.max(leftFrame.length, rightFrame.length)\n      for (let i = 0; i < frameLength; i++) {\n        const leftSample = i < leftFrame.length ? leftFrame[i] : 0\n        const rightSample = i < rightFrame.length ? rightFrame[i] : 0\n        const leftClipped = Math.max(-1, Math.min(1, leftSample))\n        const rightClipped = Math.max(-1, Math.min(1, rightSample))\n        view.setInt16(\n          offset,\n          leftClipped < 0\n            ? leftClipped * WAV_PCM_MAX_NEGATIVE\n            : leftClipped * WAV_PCM_MAX_POSITIVE,\n          true\n        )\n        offset += WAV_BYTES_PER_SAMPLE\n        view.setInt16(\n          offset,\n          rightClipped < 0\n            ? rightClipped * WAV_PCM_MAX_NEGATIVE\n            : rightClipped * WAV_PCM_MAX_POSITIVE,\n          true\n        )\n        offset += WAV_BYTES_PER_SAMPLE\n      }\n    }\n\n    return new Blob([buffer], { type: 'audio/wav' })\n  }\n\n  /**\n   * Writes WAV file header to DataView\n   * @param {DataView} view - DataView to write to\n   * @param {number} totalSamples - Total number of samples\n   * @param {number} channelCount - Number of audio channels\n   * @param {number} sampleRate - Audio sample rate\n   * @private\n   */\n  static _writeWavHeader(view, totalSamples, channelCount, sampleRate) {\n    const byteRate = sampleRate * channelCount * WAV_BYTES_PER_SAMPLE\n    const blockAlign = channelCount * WAV_BYTES_PER_SAMPLE\n    const dataSize = totalSamples * blockAlign\n    const fileSize = WAV_DATA_OFFSET + dataSize\n\n    let pos = 0\n\n    // Helper function to write ASCII strings\n    const writeString = (offset, string) => {\n      for (let i = 0; i < string.length; i++) {\n        view.setUint8(offset + i, string.charCodeAt(i))\n      }\n    }\n\n    // RIFF header\n    writeString(pos, 'RIFF')\n    pos += 4\n    view.setUint32(pos, fileSize, true)\n    pos += 4\n    writeString(pos, 'WAVE')\n    pos += 4\n\n    // fmt chunk\n    writeString(pos, 'fmt ')\n    pos += 4\n    view.setUint32(pos, 16, true) // chunk size\n    pos += 4\n    view.setUint16(pos, 1, true) // PCM format\n    pos += 2\n    view.setUint16(pos, channelCount, true)\n    pos += 2\n    view.setUint32(pos, sampleRate, true)\n    pos += 4\n    view.setUint32(pos, byteRate, true)\n    pos += 4\n    view.setUint16(pos, blockAlign, true)\n    pos += 2\n    view.setUint16(pos, WAV_BITS_PER_SAMPLE, true) // bits per sample\n    pos += 2\n\n    // data chunk\n    writeString(pos, 'data')\n    pos += 4\n    view.setUint32(pos, dataSize, true)\n  }\n\n  /**\n   * Parses an AEA file blob to extract header info and frame data\n   * @param {Blob} blob - AEA file blob to parse\n   * @returns {Promise<Object>} Parsed AEA file data\n   * @returns {Object} returns.info - AEA file header information\n   * @returns {Uint8Array[]} returns.frameData - Array of frame data buffers\n   */\n  static async parseAeaBlob(blob) {\n    const buffer = await blob.arrayBuffer()\n    const header = new Uint8Array(buffer.slice(0, AEA_HEADER_SIZE))\n    const info = AeaFile.parseHeader(header)\n\n    const frameData = []\n    for (let i = AEA_HEADER_SIZE; i < buffer.byteLength; i += SOUND_UNIT_SIZE) {\n      const frame = new Uint8Array(buffer.slice(i, i + SOUND_UNIT_SIZE))\n      if (frame.length === SOUND_UNIT_SIZE) {\n        frameData.push(frame)\n      }\n    }\n\n    return { info, frameData }\n  }\n\n  /**\n   * Creates a stream of deserialized frames from binary frame data\n   * @param {Uint8Array[]} frameData - Array of binary frame data\n   * @returns {Generator<Object>} Stream of deserialized frame objects\n   */\n  static *deserializedFrameStream(frameData) {\n    for (const frame of frameData) {\n      yield deserializeFrame(frame)\n    }\n  }\n\n  /**\n   * Assembles PCM frames into a continuous buffer\n   * @param {Float32Array[]|[Float32Array, Float32Array][]} pcmFrames - Array of PCM frames\n   * @param {number} channelCount - Number of audio channels\n   * @returns {Float32Array} Continuous PCM buffer (interleaved for stereo)\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static assemblePcmFrames(pcmFrames, channelCount) {\n    if (channelCount === 1) {\n      const totalSamples = pcmFrames.reduce(\n        (sum, frame) => sum + frame.length,\n        0\n      )\n      const pcm = new Float32Array(totalSamples)\n      let offset = 0\n      for (const frame of pcmFrames) {\n        pcm.set(frame, offset)\n        offset += frame.length\n      }\n      return pcm\n    } else if (channelCount === 2) {\n      const totalSamples = pcmFrames.reduce(\n        (sum, [left, right]) => sum + Math.max(left.length, right.length),\n        0\n      )\n      const pcm = new Float32Array(totalSamples * 2)\n\n      let pcmOffset = 0\n      for (const [leftFrame, rightFrame] of pcmFrames) {\n        const frameLength = Math.max(leftFrame.length, rightFrame.length)\n        for (let j = 0; j < frameLength; j++) {\n          pcm[pcmOffset * 2] = j < leftFrame.length ? leftFrame[j] : 0\n          pcm[pcmOffset * 2 + 1] = j < rightFrame.length ? rightFrame[j] : 0\n          pcmOffset++\n        }\n      }\n      return pcm\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n}\n","/**\n * Carta1 Web Worker - Self-contained audio codec worker\n *\n * Provides asynchronous encoding and decoding of ATRAC1 audio format\n * in a separate thread to prevent blocking the main UI thread.\n *\n * Usage:\n *   Include this file as a web worker:\n *   new Worker('carta1-worker.min.js')\n */\n\nimport { AeaFile } from '../io/serialization.js'\nimport { EncoderOptions } from '../core/options.js'\nimport { AudioProcessor } from '../io/processor.js'\nimport { SAMPLE_RATE } from '../core/constants.js'\n\n/**\n * Handle messages from the main thread\n *\n * Supported message types:\n * - 'encode': Encode PCM to ATRAC1\n * - 'decode': Decode ATRAC1 to PCM\n * - 'parseAea': Parse AEA blob metadata\n * - 'getEncoderOptions': Get default encoder options\n */\nself.onmessage = async (e) => {\n  const { jobId, type, pcmData, aea, options, title, blob } = e.data\n\n  try {\n    let result\n\n    switch (type) {\n      case 'encode':\n        result = await encodeAtrac(pcmData, options, title)\n        break\n      case 'decode':\n        result = await decodeAtrac(aea)\n        break\n      case 'parseAea':\n        result = await AudioProcessor.parseAeaBlob(blob)\n        break\n      case 'getEncoderOptions':\n        result = getEncoderOptionsMetadata()\n        break\n      default:\n        throw new Error(`Unknown worker message type: ${type}`)\n    }\n\n    self.postMessage({ jobId, result })\n  } catch (error) {\n    console.error('Worker error:', error)\n    self.postMessage({ jobId, error: error.message })\n  }\n}\n\n/**\n * Encode PCM audio data to ATRAC1 format\n *\n * @param {Array<Float32Array>} pcmData - PCM audio data [mono] or [left, right]\n * @param {Object} options - Encoder options\n * @returns {Promise<Object>} Encoded AEA blob\n */\nasync function encodeAtrac(pcmData, options, title) {\n  title = title ?? 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* Carta1 Audio Codec - Core Constants\n */\n\n// Audio format\nexport const SAMPLE_RATE = 44100\nexport const SAMPLES_PER_FRAME = 512\nexport const FRAME_RATE = SAMPLE_RATE / SAMPLES_PER_FRAME\n\n// AEA file format\nexport const AEA_MAGIC = new Uint8Array([0x00, 0x08, 0x00, 0x00])\nexport const AEA_HEADER_SIZE = 2048\nexport const AEA_TITLE_OFFSET = 4\nexport const AEA_TITLE_SIZE = 256\nexport const AEA_FRAME_COUNT_OFFSET = 260\nexport const AEA_CHANNEL_COUNT_OFFSET = 264\n\n// Frame structure\nexport const SOUND_UNIT_SIZE = 212\nexport const FRAME_BITS = SOUND_UNIT_SIZE * 8\nexport const FRAME_OVERHEAD_BITS = 40\nexport const BITRATE_PER_CHANNEL = SOUND_UNIT_SIZE * FRAME_RATE * 8\n\n// BFU (Block Floating Unit) configuration\nexport const NUM_BFUS = 52\nexport const MAX_BFU_SIZE = 20\nexport const BITS_PER_BFU_METADATA = 10\n\nexport const SPECS_PER_BFU = new Int32Array([\n  8, 8, 8, 8, 4, 4, 4, 4, 8, 8, 8, 8, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 7, 7,\n  7, 7, 9, 9, 9, 9, 10, 10, 10, 10, 12, 12, 12, 12, 12, 12, 12, 12, 20, 20, 20,\n  20, 20, 20, 20, 20,\n])\n\nexport const BFU_AMOUNTS_COUNT = 8\nexport const BFU_AMOUNTS = new Int32Array([20, 28, 32, 36, 40, 44, 48, 52])\nexport const BFU_BAND_BOUNDARIES = new Int32Array([20, 36, 52])\n\n// BFU start positions\nexport const BFU_START_LONG = new Int32Array([\n  0, 8, 16, 24, 32, 36, 40, 44, 48, 56, 64, 72, 80, 86, 92, 98, 104, 110, 116,\n  122, 128, 134, 140, 146, 152, 159, 166, 173, 180, 189, 198, 207, 216, 226,\n  236, 246, 256, 268, 280, 292, 304, 316, 328, 340, 352, 372, 392, 412, 432,\n  452, 472, 492,\n])\n\nexport const BFU_START_SHORT = new Int32Array([\n  0, 32, 64, 96, 8, 40, 72, 104, 12, 44, 76, 108, 20, 52, 84, 116, 26, 58, 90,\n  122, 128, 160, 192, 224, 134, 166, 198, 230, 141, 173, 205, 237, 150, 182,\n  214, 246, 256, 288, 320, 352, 384, 416, 448, 480, 268, 300, 332, 364, 396,\n  428, 460, 492,\n])\n\n// Transform sizes\nexport const MDCT_SIZE_SHORT = 64\nexport const MDCT_SIZE_MID = 256\nexport const MDCT_SIZE_LONG = 512\n\n// Window functions\nexport const WINDOW_SHORT = (() => {\n  const table = new Float64Array(32)\n  for (let i = 0; i < 32; i++) {\n    table[i] = Math.sin(((i + 0.5) * Math.PI) / 64)\n  }\n  return table\n})()\n\n// QMF configuration\nexport const QMF_TAPS = 48\nexport const QMF_DELAY = 46\nexport const QMF_HIGH_BAND_DELAY = 39\n\n// QMF prototype filter coefficients\nexport const QMF_COEFFS = new Float32Array([\n  -0.00001461907, -0.00009205479, -0.000056157569, 0.00030117269, 0.0002422519,\n  -0.00085293897, -0.0005205574, 0.0020340169, 0.00078333891, -0.0042153862,\n  -0.00075614988, 0.0078402944, -0.000061169922, -0.01344162, 0.0024626821,\n  0.021736089, -0.007801671, -0.034090221, 0.01880949, 0.054326009,\n  -0.043596379, -0.099384367, 0.13207909, 0.46424159,\n])\n\n// QMF window function\nexport const QMF_WINDOW = (() => {\n  const window = new Float32Array(QMF_TAPS)\n  for (let i = 0; i < 24; i++) {\n    window[i] = QMF_COEFFS[i] * 2.0\n    window[47 - i] = QMF_COEFFS[i] * 2.0\n  }\n  return window\n})()\n\n// QMF even/odd taps for optimization\nexport const QMF_EVEN = (() => {\n  const even = new Float32Array(24)\n  for (let i = 0; i < 24; i++) {\n    even[i] = QMF_WINDOW[i * 2]\n  }\n  return even\n})()\n\nexport const QMF_ODD = (() => {\n  const odd = new Float32Array(24)\n  for (let i = 0; i < 24; i++) {\n    odd[i] = QMF_WINDOW[i * 2 + 1]\n  }\n  return odd\n})()\n\n// FFT sizes for transient detection\nexport const FFT_SIZE_LOW = 128\nexport const FFT_SIZE_MID = 128\nexport const FFT_SIZE_HIGH = 256\n\n// MDCT/IMDCT transform configuration\nexport const MDCT_BAND_CONFIGS = [\n  { size: 128, windowStart: 48 }, // Low band (0-5.5kHz)\n  { size: 128, windowStart: 48 }, // Mid band (5.5-11kHz)\n  { size: 256, windowStart: 112 }, // High band (11-22kHz)\n]\n\nexport const MDCT_SHORT_BLOCK_SIZE = 32\nexport const MDCT_OVERLAP_SIZE = 32\nexport const MDCT_TAIL_WINDOW_SIZE = 16\n\n// WAV format\nexport const WAV_HEADER_SIZE = 44\nexport const WAV_BYTES_PER_SAMPLE = 2\nexport const WAV_BITS_PER_SAMPLE = 16\nexport const WAV_PCM_MAX_POSITIVE = 0x7fff\nexport const WAV_PCM_MAX_NEGATIVE = 0x8000\nexport const WAV_DATA_OFFSET = 36\n\n// Frame serialization\nexport const FRAME_HEADER_BITS = 16\nexport const FRAME_WORD_LENGTH_BITS = 4\nexport const FRAME_SCALE_FACTOR_BITS = 6\n\n// Quantization\nexport const QUANTIZATION_SIGN_BIT_SHIFT = 1\nexport const MAX_WORD_LENGTH_INDEX = 15\nexport const WORD_LENGTH_BITS = new Int32Array([\n  0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,\n])\nexport const SCALE_FACTORS = (() => {\n  const table = new Float64Array(64)\n  for (let i = 0; i < 64; i++) {\n    table[i] = Math.pow(2.0, i / 3.0 - 21)\n  }\n  return table\n})()\n\n// Bit allocation\nexport const INV_POWER_OF_TWO = (() => {\n  const maxBits = WORD_LENGTH_BITS[MAX_WORD_LENGTH_INDEX]\n  const table = new Float64Array(maxBits + 1)\n  for (let b = 0; b <= maxBits; b++) {\n    table[b] = Math.pow(2, -b)\n  }\n  return table\n})()\n","/**\n * Carta1 Audio Codec - Bitstream Operations\n *\n * This module provides low-level bitstream packing and unpacking operations\n * for reading and writing ATRAC1 frame data with bit-level precision.\n */\n\n/**\n * Pack bits into a buffer at a specific bit position\n * @param {Uint8Array} buffer - Destination buffer\n * @param {number} bitPosition - Bit position to start writing at\n * @param {number} value - Value to pack\n * @param {number} bitCount - Number of bits to pack\n */\nexport function packBits(buffer, bitPosition, value, bitCount) {\n  if (bitCount === 0) return\n\n  let byteIndex = Math.floor(bitPosition / 8)\n  let bitOffset = bitPosition % 8\n\n  value &= (1 << bitCount) - 1\n\n  let bitsWritten = 0\n  while (bitsWritten < bitCount && byteIndex < buffer.length) {\n    const bitsAvailable = 8 - bitOffset\n    const bitsToWrite = Math.min(bitCount - bitsWritten, bitsAvailable)\n\n    const shift = bitCount - bitsWritten - bitsToWrite\n    const valueBits = (value >> shift) & ((1 << bitsToWrite) - 1)\n\n    const mask = ((1 << bitsToWrite) - 1) << (bitsAvailable - bitsToWrite)\n    buffer[byteIndex] =\n      (buffer[byteIndex] & ~mask) | (valueBits << (bitsAvailable - bitsToWrite))\n\n    bitsWritten += bitsToWrite\n    byteIndex++\n    bitOffset = 0\n  }\n}\n\n/**\n * Unpack bits from a buffer at a specific bit position\n * @param {Uint8Array} buffer - Source buffer\n * @param {number} bitPosition - Bit position to start reading from\n * @param {number} bitCount - Number of bits to unpack\n * @returns {number} Unpacked unsigned value\n */\nexport function unpackBits(buffer, bitPosition, bitCount) {\n  if (bitCount === 0) return 0\n\n  let byteIndex = Math.floor(bitPosition / 8)\n  let bitOffset = bitPosition % 8\n  let value = 0\n\n  for (let bitsRead = 0; bitsRead < bitCount && byteIndex < buffer.length; ) {\n    const bitsAvailable = 8 - bitOffset\n    const bitsToRead = Math.min(bitCount - bitsRead, bitsAvailable)\n\n    const mask = (1 << bitsToRead) - 1\n    const bits = (buffer[byteIndex] >> (bitsAvailable - bitsToRead)) & mask\n\n    value = (value << bitsToRead) | bits\n    bitsRead += bitsToRead\n    byteIndex++\n    bitOffset = 0\n  }\n\n  return value\n}\n\n/**\n * Unpack signed bits from a buffer at a specific bit position\n * @param {Uint8Array} buffer - Source buffer\n * @param {number} bitPosition - Bit position to start reading from\n * @param {number} bitCount - Number of bits to unpack\n * @returns {number} Unpacked signed value (two's complement)\n */\nexport function unpackSignedBits(buffer, bitPosition, bitCount) {\n  const value = unpackBits(buffer, bitPosition, bitCount)\n  const signBit = 1 << (bitCount - 1)\n  return value >= signBit ? value - (1 << bitCount) : value\n}\n","/**\n * Carta1 Audio Codec - Serialization Module\n *\n * Handles serialization and deserialization of ATRAC1 audio frames and AEA file format.\n * Provides functions for packing encoded frame data into binary format and unpacking\n * it back into structured data for processing.\n *\n * The module manages:\n * - Frame data serialization/deserialization with proper bit packing\n * - AEA file header creation and parsing\n * - Binary format compliance for ATRAC1 specification\n */\n\nimport {\n  SOUND_UNIT_SIZE,\n  BFU_AMOUNTS,\n  SPECS_PER_BFU,\n  AEA_HEADER_SIZE,\n  AEA_MAGIC,\n  AEA_TITLE_OFFSET,\n  AEA_TITLE_SIZE,\n  AEA_FRAME_COUNT_OFFSET,\n  AEA_CHANNEL_COUNT_OFFSET,\n  WORD_LENGTH_BITS,\n  FRAME_HEADER_BITS,\n  FRAME_WORD_LENGTH_BITS,\n  FRAME_SCALE_FACTOR_BITS,\n} from '../core/constants.js'\nimport { packBits, unpackBits, unpackSignedBits } from './bitstream.js'\n\n/**\n * Serializes encoded frame data into binary format\n * @param {Object} frameData - The encoded frame data to serialize\n * @param {number} frameData.nBfu - Number of block floating units\n * @param {number[]} frameData.blockModes - Block modes for each band [0-2]\n * @param {Int32Array} frameData.scaleFactorIndices - Scale factor indices for each BFU\n * @param {Int32Array} frameData.wordLengthIndices - Word length indices for each BFU\n * @param {Int32Array[]} frameData.quantizedCoefficients - Quantized spectral coefficients\n * @returns {Uint8Array} Serialized frame data buffer\n */\nexport function serializeFrame(frameData) {\n  const buffer = new Uint8Array(SOUND_UNIT_SIZE)\n  const view = new DataView(buffer.buffer)\n\n  // Pack header (16 bits)\n  const bfuIndex = BFU_AMOUNTS.indexOf(frameData.nBfu)\n  const header =\n    ((2 - frameData.blockModes[0]) << 14) |\n    ((2 - frameData.blockModes[1]) << 12) |\n    ((3 - frameData.blockModes[2]) << 10) |\n    (bfuIndex << 5)\n\n  view.setUint16(0, header, false)\n  let bitPosition = FRAME_HEADER_BITS\n\n  // Pack word lengths (4 bits each)\n  for (let i = 0; i < frameData.nBfu; i++) {\n    packBits(\n      buffer,\n      bitPosition,\n      frameData.wordLengthIndices[i],\n      FRAME_WORD_LENGTH_BITS\n    )\n    bitPosition += FRAME_WORD_LENGTH_BITS\n  }\n\n  // Pack scale factors (6 bits each)\n  for (let i = 0; i < frameData.nBfu; i++) {\n    packBits(\n      buffer,\n      bitPosition,\n      frameData.scaleFactorIndices[i],\n      FRAME_SCALE_FACTOR_BITS\n    )\n    bitPosition += FRAME_SCALE_FACTOR_BITS\n  }\n\n  // Pack quantized coefficients\n  for (let i = 0; i < frameData.nBfu; i++) {\n    const bitsPerSample = WORD_LENGTH_BITS[frameData.wordLengthIndices[i]]\n    if (bitsPerSample > 0) {\n      const coefficients = frameData.quantizedCoefficients[i]\n      for (const coefficient of coefficients) {\n        const value =\n          coefficient < 0 ? coefficient + (1 << bitsPerSample) : coefficient\n        packBits(buffer, bitPosition, value, bitsPerSample)\n        bitPosition += bitsPerSample\n      }\n    }\n  }\n\n  // Zero padding at end\n  buffer[SOUND_UNIT_SIZE - 3] = 0\n  buffer[SOUND_UNIT_SIZE - 2] = 0\n  buffer[SOUND_UNIT_SIZE - 1] = 0\n\n  return buffer\n}\n\n/**\n * Deserializes binary frame data back into structured format\n * @param {Uint8Array} buffer - Binary frame data buffer\n * @returns {Object} Deserialized frame data\n * @returns {number} returns.nBfu - Number of block floating units\n * @returns {number[]} returns.blockModes - Block modes for each band\n * @returns {Int32Array} returns.scaleFactorIndices - Scale factor indices for each BFU\n * @returns {Int32Array} returns.wordLengthIndices - Word length indices for each BFU\n * @returns {Int32Array[]} returns.quantizedCoefficients - Quantized spectral coefficients\n * @throws {Error} If buffer size is invalid\n */\nexport function deserializeFrame(buffer) {\n  if (buffer.length !== SOUND_UNIT_SIZE) {\n    throw new Error(`Frame must be ${SOUND_UNIT_SIZE} bytes`)\n  }\n\n  const view = new DataView(buffer.buffer, buffer.byteOffset, buffer.byteLength)\n\n  // Unpack header\n  const header = view.getUint16(0, false)\n  const blockModes = [\n    2 - ((header >> 14) & 3),\n    2 - ((header >> 12) & 3),\n    3 - ((header >> 10) & 3),\n  ]\n  const bfuIndex = (header >> 5) & 7\n  const nBfu = BFU_AMOUNTS[bfuIndex]\n\n  let bitPosition = FRAME_HEADER_BITS\n\n  // Unpack word lengths\n  const wordLengthIndices = new Int32Array(nBfu)\n  for (let i = 0; i < nBfu; i++) {\n    wordLengthIndices[i] = unpackBits(\n      buffer,\n      bitPosition,\n      FRAME_WORD_LENGTH_BITS\n    )\n    bitPosition += FRAME_WORD_LENGTH_BITS\n  }\n\n  // Unpack scale factors\n  const scaleFactorIndices = new Int32Array(nBfu)\n  for (let i = 0; i < nBfu; i++) {\n    scaleFactorIndices[i] = unpackBits(\n      buffer,\n      bitPosition,\n      FRAME_SCALE_FACTOR_BITS\n    )\n    bitPosition += FRAME_SCALE_FACTOR_BITS\n  }\n\n  // Unpack coefficients\n  const quantizedCoefficients = []\n  for (let i = 0; i < nBfu; i++) {\n    const bitsPerSample = WORD_LENGTH_BITS[wordLengthIndices[i]]\n    const bfuSize = SPECS_PER_BFU[i]\n    const coefficients = new Int32Array(bfuSize)\n\n    if (bitsPerSample > 0) {\n      for (let j = 0; j < bfuSize; j++) {\n        coefficients[j] = unpackSignedBits(buffer, bitPosition, bitsPerSample)\n        bitPosition += bitsPerSample\n      }\n    }\n\n    quantizedCoefficients.push(coefficients)\n  }\n\n  return {\n    nBfu,\n    scaleFactorIndices,\n    wordLengthIndices,\n    quantizedCoefficients,\n    blockModes,\n  }\n}\n\n/**\n * AEA file format handler for ATRAC1 audio files\n * Manages creation and parsing of AEA file headers\n */\nexport class AeaFile {\n  /**\n   * Creates an AEA file header with the specified metadata\n   * @param {string} [title=''] - Audio file title (max 255 characters)\n   * @param {number} [frameCount=0] - Number of audio frames in the file\n   * @param {number} [channelCount=1] - Number of audio channels (1 or 2)\n   * @returns {Uint8Array} AEA file header buffer\n   */\n  static createHeader(title = '', frameCount = 0, channelCount = 1) {\n    const header = new Uint8Array(AEA_HEADER_SIZE)\n    const view = new DataView(header.buffer)\n\n    // Magic number\n    header.set(AEA_MAGIC, 0)\n\n    // Title (null-terminated string)\n    const titleBytes = new TextEncoder().encode(title)\n    header.set(\n      titleBytes.subarray(0, Math.min(titleBytes.length, AEA_TITLE_SIZE - 1)),\n      AEA_TITLE_OFFSET\n    )\n\n    // Frame count\n    view.setUint32(AEA_FRAME_COUNT_OFFSET, frameCount, true)\n\n    // Channel count\n    header[AEA_CHANNEL_COUNT_OFFSET] = channelCount\n\n    return header\n  }\n\n  /**\n   * Parses an AEA file header to extract metadata\n   * @param {Uint8Array} header - AEA file header buffer\n   * @returns {Object} Parsed header information\n   * @returns {string} returns.title - Audio file title\n   * @returns {number} returns.frameCount - Number of audio frames\n   * @returns {number} returns.channelCount - Number of audio channels\n   * @throws {Error} If header size is invalid or magic number doesn't match\n   */\n  static parseHeader(header) {\n    if (header.length !== AEA_HEADER_SIZE) {\n      throw new Error(`Header must be ${AEA_HEADER_SIZE} bytes`)\n    }\n\n    // Check magic number\n    for (let i = 0; i < AEA_MAGIC.length; i++) {\n      if (header[i] !== AEA_MAGIC[i]) {\n        throw new Error('Invalid AEA file')\n      }\n    }\n\n    const view = new DataView(\n      header.buffer,\n      header.byteOffset,\n      header.byteLength\n    )\n\n    // Extract title\n    const titleEnd = header.indexOf(0, AEA_TITLE_OFFSET)\n    const titleLength =\n      titleEnd === -1 ? AEA_TITLE_SIZE : titleEnd - AEA_TITLE_OFFSET\n    const title = new TextDecoder().decode(\n      header.subarray(AEA_TITLE_OFFSET, AEA_TITLE_OFFSET + titleLength)\n    )\n\n    // Extract metadata\n    const frameCount = view.getUint32(AEA_FRAME_COUNT_OFFSET, true)\n    const channelCount = header[AEA_CHANNEL_COUNT_OFFSET]\n\n    return { title, frameCount, channelCount }\n  }\n}\n","/**\n * Carta1 Audio Codec - Encoder Options\n *\n * This module provides configuration management for the ATRAC1 encoder,\n * including parameter validation and metadata for UI controls.\n */\n\n/**\n * Configuration options for the ATRAC1 encoder\n */\nclass EncoderOptions {\n  /**\n   * Create a new EncoderOptions instance\n   * @param {Object} options - Initial option values\n   */\n  constructor(options = {}) {\n    this.values = {\n      transientThresholdLow: 1.0,\n      transientThresholdMid: 1.5,\n      transientThresholdHigh: 2.0,\n      allocationBias: 1.0,\n    }\n\n    this.metadata = {\n      transientThresholdLow: {\n        default: this.values.transientThresholdLow,\n        name: 'Low Band Transient Threshold',\n        description:\n          'Controls the transient detection threshold for the low frequency band (0-5.5kHz). Lower values make detection more sensitive, triggering short blocks more easily. Higher values require stronger transients.',\n        range: [0.01, 2],\n        step: 0.01,\n      },\n      transientThresholdMid: {\n        default: this.values.transientThresholdMid,\n        name: 'Mid Band Transient Threshold',\n        description:\n          'Controls the transient detection threshold for the mid frequency band (5.5-11kHz). Lower values make detection more sensitive, triggering short blocks more easily. Higher values require stronger transients.',\n        range: [0.01, 3],\n        step: 0.01,\n      },\n      transientThresholdHigh: {\n        default: this.values.transientThresholdHigh,\n        name: 'High Band Transient Threshold',\n        description:\n          'Controls the transient detection threshold for the high frequency band (11-22kHz). Lower values make detection more sensitive, triggering short blocks more easily. Higher values require stronger transients.',\n        range: [0.01, 4],\n        step: 0.01,\n      },\n      allocationBias: {\n        default: this.values.allocationBias,\n        name: 'Bit allocation bias',\n        description:\n          'Controls how bits are distributed across the frequency spectrum during allocation. Higher values focus more bits on loud spectral components, while lower values spread bits more evenly across frequencies.',\n        range: [0.5, 3.0],\n        step: 0.01,\n      },\n    }\n\n    // Apply user-provided options\n    if (options) {\n      this.setOptions(options)\n    }\n  }\n\n  /**\n   * Set multiple options at once\n   * @param {Object} options - Object containing option key-value pairs\n   */\n  setOptions(options) {\n    for (const [key, value] of Object.entries(options)) {\n      if (key in this.values) {\n        this.setValue(key, value)\n      }\n    }\n  }\n\n  /**\n   * Set a single option value with validation\n   * @param {string} key - Option key\n   * @param {*} value - Option value\n   * @throws {Error} If key is unknown or value is out of range\n   */\n  setValue(key, value) {\n    if (!(key in this.metadata)) {\n      throw new Error(`Unknown option: ${key}`)\n    }\n\n    const meta = this.metadata[key]\n    const [min, max] = meta.range\n\n    if (value < min || value > max) {\n      throw new Error(\n        `Value for ${key} must be between ${min} and ${max}, got ${value}`\n      )\n    }\n\n    this.values[key] = value\n  }\n\n  /**\n   * Get the value of an option\n   * @param {string} key - Option key\n   * @returns {*} Option value\n   * @throws {Error} If key is unknown\n   */\n  getValue(key) {\n    if (!(key in this.values)) {\n      throw new Error(`Unknown option: ${key}`)\n    }\n    return this.values[key]\n  }\n\n  get transientThresholdLow() {\n    return this.values.transientThresholdLow\n  }\n\n  get transientThresholdMid() {\n    return this.values.transientThresholdMid\n  }\n\n  get transientThresholdHigh() {\n    return this.values.transientThresholdHigh\n  }\n\n  get allocationBias() {\n    return this.values.allocationBias\n  }\n\n  getMetadata(key) {\n    return this.metadata[key]\n  }\n\n  getAllMetadata() {\n    return { ...this.metadata }\n  }\n\n  reset() {\n    for (const [key, meta] of Object.entries(this.metadata)) {\n      this.values[key] = meta.default\n    }\n  }\n\n  toObject() {\n    return {\n      values: { ...this.values },\n      metadata: { ...this.metadata },\n    }\n  }\n}\n\nexport { EncoderOptions }\n","/**\n * Carta1 Audio Codec - Utilities\n *\n * This module provides common utility functions for the ATRAC1 codec including\n * pipeline composition, error handling, and audio stream processing utilities\n * for delay compensation and frame padding.\n */\n\n/**\n * Throws an error with the given message\n * @param {string} msg - Error message\n * @throws {Error} Always throws an error with the provided message\n */\nexport function throwError(msg) {\n  throw new Error(msg)\n}\n\n/**\n * Creates a pipeline by composing multiple processing stages\n * @param {Object} context - Shared context passed to all stages\n * @param {...Function} stages - Stage functions to compose\n * @returns {Function} Composed pipeline function\n */\nexport function pipe(context, ...stages) {\n  const functions = stages.map((stage) => stage(context))\n\n  return (input) => {\n    return functions.reduce((value, fn) => fn(value), input)\n  }\n}\n\n/**\n * Reverses the spectral order of coefficients for mid/high band processing\n *\n * Used in both MDCT and IMDCT stages to handle spectral reversal required\n * by the ATRAC1 format for mid and high frequency bands.\n *\n * @param {Float32Array} spectrum - Input spectrum coefficients\n * @param {Object} reversalBuffers - Pre-allocated buffers for different spectrum sizes\n * @returns {Float32Array} Spectrum with reversed coefficient order\n */\nexport function reverseSpectrum(spectrum, reversalBuffers) {\n  const reversed = reversalBuffers[spectrum.length]\n  for (let i = 0; i < spectrum.length; i++) {\n    reversed[i] = spectrum[spectrum.length - 1 - i]\n  }\n  return reversed\n}\n\n/**\n * Calculates the starting offset for a frequency band in the coefficient array\n *\n * @param {number} bandIndex - Band index (0=low, 1=mid, 2=high)\n * @returns {number} Starting offset in the 512-sample coefficient array\n */\nexport function calculateBandOffset(bandIndex) {\n  return bandIndex === 0 ? 0 : bandIndex === 1 ? 128 : 256\n}\n\n/**\n * Extracts coefficients for a specific frequency band from the full coefficient array\n *\n * @param {Float32Array} coefficients - Full 512-sample coefficient array\n * @param {number} bandIndex - Band index (0=low, 1=mid, 2=high)\n * @returns {Float32Array} Subarray containing coefficients for the specified band\n */\nexport function extractBandCoefficients(coefficients, bandIndex) {\n  const offsets = [0, 128, 256, 512]\n  return coefficients.subarray(offsets[bandIndex], offsets[bandIndex + 1])\n}\n","/**\n * Carta1 Audio Codec - Quadrature Mirror Filter\n *\n * This module implements QMF analysis and synthesis for subband decomposition\n * in the ATRAC1 codec. The QMF splits the audio signal into multiple frequency\n * bands for independent processing and coding.\n */\n\nimport { QMF_DELAY, QMF_EVEN, QMF_ODD } from '../core/constants.js'\nimport { throwError } from '../utils.js'\n\n/**\n * Perform QMF analysis filtering to split input into low and high frequency bands\n * @param {Float32Array} input - Input audio samples\n * @param {Float32Array} delayLine - QMF delay line state\n * @param {Object} qmfWorkBuffers - Optional work buffers for processing\n * @returns {Object} Object containing lowBand, highBand, and newDelay\n */\nexport function qmfAnalysis(input, delayLine, qmfWorkBuffers = null) {\n  const inputLength = input.length\n  const outputLength = inputLength >> 1\n\n  const workBuffer =\n    qmfWorkBuffers?.delay?.[inputLength] ??\n    throwError(`qmfAnalysis: qmfWorkBuffers.delay[${inputLength}] is required`)\n  workBuffer.set(delayLine)\n  workBuffer.set(input, delayLine.length)\n\n  const lowBand = new Float32Array(outputLength)\n  const highBand = new Float32Array(outputLength)\n\n  for (let i = 0; i < outputLength; i++) {\n    let evenSum = 0\n    let oddSum = 0\n    const offset = i * 2\n\n    // Unrolled convolution\n    for (let j = 0; j < 24; j++) {\n      evenSum += workBuffer[offset + 47 - j * 2] * QMF_EVEN[j]\n      oddSum += workBuffer[offset + 46 - j * 2] * QMF_ODD[j]\n    }\n\n    lowBand[i] = evenSum + oddSum\n    highBand[i] = evenSum - oddSum\n  }\n\n  // Update delay line\n  const newDelay = workBuffer.slice(-QMF_DELAY)\n  return { lowBand, highBand, newDelay }\n}\n\n/**\n * Perform QMF synthesis filtering to reconstruct audio from low and high frequency bands\n * @param {Float32Array} lowBand - Low frequency band samples\n * @param {Float32Array} highBand - High frequency band samples\n * @param {Float32Array} delayLine - QMF delay line state\n * @param {Object} qmfWorkBuffers - Optional work buffers for processing\n * @returns {Object} Object containing output and newDelay\n */\nexport function qmfSynthesis(\n  lowBand,\n  highBand,\n  delayLine,\n  qmfWorkBuffers = null\n) {\n  const subbandLength = lowBand.length\n  const outputLength = subbandLength * 2\n\n  const workBuffer =\n    qmfWorkBuffers?.delay?.[outputLength] ??\n    throwError(\n      `qmfSynthesis: qmfWorkBuffers.delay[${outputLength}] is required`\n    )\n  workBuffer.set(delayLine)\n\n  // Upsample and merge\n  for (let i = 0; i < subbandLength; i++) {\n    const low = lowBand[i]\n    const high = highBand[i]\n    const offset = QMF_DELAY + i * 2\n    workBuffer[offset] = 0.5 * (low + high)\n    workBuffer[offset + 1] = 0.5 * (low - high)\n  }\n\n  const output = new Float32Array(outputLength)\n\n  // Convolution\n  for (let i = 0; i < subbandLength; i++) {\n    const offset = i * 2\n    let sample0 = 0\n    let sample1 = 0\n\n    for (let j = 0; j < 24; j++) {\n      const idx = offset + j * 2\n      sample0 += workBuffer[idx] * QMF_EVEN[j]\n      sample1 += workBuffer[idx + 1] * QMF_ODD[j]\n    }\n\n    output[i * 2] = sample1\n    output[i * 2 + 1] = sample0\n  }\n\n  const newDelay = workBuffer.slice(-QMF_DELAY)\n  return { output, newDelay }\n}\n","/**\n * Carta1 Audio Codec - Fast Fourier Transform\n *\n * This module implements a radix-2 Cooley-Tukey FFT algorithm for frequency domain\n * analysis used in psychoacoustic modeling and transient detection.\n */\n\nexport class FFT {\n  /**\n   * Perform in-place FFT on complex data\n   * @param {Float32Array} real - Real part of input/output data\n   * @param {Float32Array} imag - Imaginary part of input/output data\n   */\n  static fft(real, imag) {\n    const size = real.length\n    if (size === 1) return\n\n    const bits = Math.log2(size)\n\n    // Bit reversal\n    for (let i = 0; i < size; i++) {\n      let reversed = 0\n      let temp = i\n      for (let b = 0; b < bits; b++) {\n        reversed = (reversed << 1) | (temp & 1)\n        temp >>= 1\n      }\n      if (reversed > i) {\n        ;[real[i], real[reversed]] = [real[reversed], real[i]]\n        ;[imag[i], imag[reversed]] = [imag[reversed], imag[i]]\n      }\n    }\n\n    // Cooley-Tukey decimation-in-time\n    for (let stride = 2; stride <= size; stride <<= 1) {\n      const halfStride = stride >> 1\n      const angle = (-2 * Math.PI) / stride\n      const wReal = Math.cos(angle)\n      const wImag = Math.sin(angle)\n\n      for (let start = 0; start < size; start += stride) {\n        let twiddleReal = 1\n        let twiddleImag = 0\n\n        for (let k = 0; k < halfStride; k++) {\n          const evenIndex = start + k\n          const oddIndex = evenIndex + halfStride\n\n          const evenReal = real[evenIndex]\n          const evenImag = imag[evenIndex]\n          const oddReal = real[oddIndex]\n          const oddImag = imag[oddIndex]\n\n          const tReal = oddReal * twiddleReal - oddImag * twiddleImag\n          const tImag = oddReal * twiddleImag + oddImag * twiddleReal\n\n          real[evenIndex] = evenReal + tReal\n          imag[evenIndex] = evenImag + tImag\n          real[oddIndex] = evenReal - tReal\n          imag[oddIndex] = evenImag - tImag\n\n          const nextReal = twiddleReal * wReal - twiddleImag * wImag\n          twiddleImag = twiddleReal * wImag + twiddleImag * wReal\n          twiddleReal = nextReal\n        }\n      }\n    }\n  }\n}\n","/**\n * Carta1 Audio Codec - Modified Discrete Cosine Transform\n *\n * This module implements forward and inverse MDCT transforms for the ATRAC1 codec.\n * The MDCT provides time-frequency analysis with perfect reconstruction and\n * 50% overlap between adjacent blocks to avoid blocking artifacts.\n */\n\nimport { FFT } from './fft.js'\nimport {\n  MDCT_SIZE_SHORT,\n  MDCT_SIZE_MID,\n  MDCT_SIZE_LONG,\n} from '../core/constants.js'\nimport { throwError } from '../utils.js'\n\n/**\n * Base class for MDCT and IMDCT transforms\n */\nclass MDCTBase {\n  constructor(size, scale) {\n    this.size = size\n    this.halfSize = size >> 1\n    this.quarterSize = size >> 2\n    this.fftSize = this.halfSize >> 1\n\n    const alpha = (2.0 * Math.PI) / (8.0 * size)\n    const omega = (2.0 * Math.PI) / size\n    const scaleRoot = Math.sqrt(scale / size)\n\n    this.sinCosTable = new Float64Array(this.halfSize)\n    for (let i = 0; i < this.quarterSize; i++) {\n      const angle = omega * i + alpha\n      this.sinCosTable[i * 2] = scaleRoot * Math.cos(angle)\n      this.sinCosTable[i * 2 + 1] = scaleRoot * Math.sin(angle)\n    }\n  }\n}\n\n/**\n * Forward MDCT transform\n */\nexport class MDCT extends MDCTBase {\n  constructor(size, scale) {\n    super(size, scale)\n  }\n\n  /**\n   * Perform forward MDCT transform\n   * @param {Float32Array} input - Time-domain input samples\n   * @param {Object} mdctBuffers - Optional work buffers for FFT\n   * @returns {Float32Array} Frequency-domain MDCT coefficients\n   */\n  transform(input, mdctBuffers = null) {\n    const buffers =\n      mdctBuffers?.[this.fftSize] ??\n      throwError(`MDCT.transform: mdctBuffers[${this.fftSize}] is required`)\n    const real =\n      buffers?.real ??\n      throwError(\n        `MDCT.transform: mdctBuffers[${this.fftSize}].real is required`\n      )\n    const imag =\n      buffers?.imag ??\n      throwError(\n        `MDCT.transform: mdctBuffers[${this.fftSize}].imag is required`\n      )\n\n    const n4 = this.quarterSize\n    const n34 = 3 * n4\n\n    real.fill(0)\n    imag.fill(0)\n\n    // Pre-FFT butterfly\n    for (let i = 0; i < n4; i += 2) {\n      const i1 = n34 - 1 - i\n      const i2 = n34 + i\n      const i3 = n4 + i\n      const i4 = n4 - 1 - i\n\n      const r = input[i1] + input[i2]\n      const im = input[i3] - input[i4]\n      const c = this.sinCosTable[i]\n      const s = this.sinCosTable[i + 1]\n\n      real[i >> 1] = r * c + im * s\n      imag[i >> 1] = im * c - r * s\n    }\n\n    for (let i = n4; i < this.halfSize; i += 2) {\n      const idx = i >> 1\n      const i1 = n34 - 1 - i\n      const i2 = i - n4\n      const i3 = n4 + i\n      const i4 = 5 * n4 - 1 - i\n\n      const r = input[i1] - input[i2]\n      const im = input[i3] + input[i4]\n      const c = this.sinCosTable[i]\n      const s = this.sinCosTable[i + 1]\n\n      real[idx] = r * c + im * s\n      imag[idx] = im * c - r * s\n    }\n\n    FFT.fft(real, imag)\n\n    // Post-FFT processing\n    const output = new Float32Array(this.halfSize)\n    for (let i = 0; i < this.fftSize; i++) {\n      const c = this.sinCosTable[i * 2]\n      const s = this.sinCosTable[i * 2 + 1]\n      const re = real[i]\n      const im = imag[i]\n\n      output[i * 2] = -re * c - im * s\n      output[this.halfSize - 1 - i * 2] = -re * s + im * c\n    }\n\n    return output\n  }\n}\n\n/**\n * Inverse MDCT transform\n */\nexport class IMDCT extends MDCTBase {\n  constructor(size, scale = null) {\n    super(size, scale || size)\n  }\n\n  /**\n   * Perform inverse MDCT transform\n   * @param {Float32Array} input - Frequency-domain MDCT coefficients\n   * @param {Object} mdctBuffers - Optional work buffers for FFT\n   * @returns {Float32Array} Time-domain output samples\n   */\n  transform(input, mdctBuffers = null) {\n    const buffers =\n      mdctBuffers?.[this.fftSize] ??\n      throwError(`IMDCT.transform: mdctBuffers[${this.fftSize}] is required`)\n    const real =\n      buffers?.real ??\n      throwError(\n        `IMDCT.transform: mdctBuffers[${this.fftSize}].real is required`\n      )\n    const imag =\n      buffers?.imag ??\n      throwError(\n        `IMDCT.transform: mdctBuffers[${this.fftSize}].imag is required`\n      )\n\n    const n4 = this.quarterSize\n    const n34 = 3 * n4\n\n    real.fill(0)\n    imag.fill(0)\n\n    // Pre-FFT processing\n    for (let i = 0; i < this.fftSize; i++) {\n      const i2 = i * 2\n      const r = -input[i2]\n      const im = -input[this.halfSize - 1 - i2]\n      const c = this.sinCosTable[i2]\n      const s = this.sinCosTable[i2 + 1]\n\n      real[i] = im * s + r * c\n      imag[i] = im * c - r * s\n    }\n\n    FFT.fft(real, imag)\n\n    // Post-FFT butterfly\n    const output = new Float32Array(this.size)\n\n    for (let i = 0; i < this.fftSize / 2; i++) {\n      const i2 = i * 2\n      const c = this.sinCosTable[i2]\n      const s = this.sinCosTable[i2 + 1]\n      const re = real[i]\n      const im = imag[i]\n\n      const r1 = re * c + im * s\n      const i1 = re * s - im * c\n\n      output[n34 - 1 - i2] = r1\n      output[n34 + i2] = r1\n      output[n4 + i2] = i1\n      output[n4 - 1 - i2] = -i1\n    }\n\n    for (let i = this.fftSize / 2; i < this.fftSize; i++) {\n      const idx = (i - this.fftSize / 2) * 2 + n4\n      const i2 = i * 2\n      const c = this.sinCosTable[i2]\n      const s = this.sinCosTable[i2 + 1]\n      const re = real[i]\n      const im = imag[i]\n\n      const r1 = re * c + im * s\n      const i1 = re * s - im * c\n\n      output[n34 - 1 - idx] = r1\n      output[idx - n4] = -r1\n      output[n4 + idx] = i1\n      output[5 * n4 - 1 - idx] = i1\n    }\n\n    return output\n  }\n}\n\n// Pre-instantiated transforms\nexport const mdct64 = new MDCT(MDCT_SIZE_SHORT, 0.5)\nexport const mdct256 = new MDCT(MDCT_SIZE_MID, 0.5)\nexport const mdct512 = new MDCT(MDCT_SIZE_LONG, 1.0)\n\nexport const imdct64 = new IMDCT(MDCT_SIZE_SHORT, MDCT_SIZE_SHORT * 8)\nexport const imdct256 = new IMDCT(MDCT_SIZE_MID, MDCT_SIZE_MID * 8)\nexport const imdct512 = new IMDCT(MDCT_SIZE_LONG, MDCT_SIZE_LONG * 4)\n\n/**\n * Perform overlap-add operation for MDCT reconstruction\n * @param {Float32Array} prev - Previous block samples\n * @param {Float32Array} curr - Current block samples\n * @param {Float32Array} window - Window function coefficients\n * @returns {Float32Array} Overlap-added output samples\n */\nexport function overlapAdd(prev, curr, window) {\n  const size = prev.length\n  const output = new Float32Array(size * 2)\n\n  for (let i = 0; i < size; i++) {\n    const w1 = window[i]\n    const w2 = window[2 * size - 1 - i]\n    const p = prev[i]\n    const c = curr[size - 1 - i]\n\n    output[i] = p * w2 - c * w1\n    output[2 * size - 1 - i] = p * w1 + c * w2\n  }\n\n  return output\n}\n","/**\n * Carta1 Audio Codec - Quantization Module\n *\n * This module handles the quantization and dequantization of MDCT frequency coefficients\n * for the Carta1 audio codec. It provides functions to convert floating-point frequency\n * domain data into integer representations suitable for bitstream encoding.\n *\n * Key functions:\n * - Quantization of coefficients using scale factors and bit depths\n * - Dequantization for reconstruction during decoding\n * - BFU (Band Frequency Unit) organization for psychoacoustic processing\n *\n * The quantization process uses configurable scale factors to maintain audio quality\n * while achieving the target compression ratio. The module supports both uniform\n * and non-uniform quantization schemes based on perceptual importance.\n */\n\nimport {\n  SPECS_PER_BFU,\n  BFU_START_LONG,\n  BFU_START_SHORT,\n  BFU_BAND_BOUNDARIES,\n  SCALE_FACTORS,\n  QUANTIZATION_SIGN_BIT_SHIFT,\n} from '../core/constants.js'\n\n/**\n * Quantize coefficients.\n * @param {Float32Array} coefficients\n * @param {number} scaleFactorIndex\n * @param {number} bitsPerSample\n * @returns {Int32Array} out\n */\nexport function quantize(coefficients, scaleFactorIndex, bitsPerSample) {\n  const length = coefficients.length\n  const out = new Int32Array(length)\n  if (bitsPerSample === 0 || scaleFactorIndex === 0) {\n    out.fill(0, 0, length)\n    return out\n  }\n\n  const scaleFactor = SCALE_FACTORS[scaleFactorIndex]\n  const quantRange = (1 << (bitsPerSample - QUANTIZATION_SIGN_BIT_SHIFT)) - 1\n  const normFactor = quantRange / scaleFactor\n\n  const hi = quantRange\n  const lo = -quantRange\n\n  for (let i = 0; i < length; i++) {\n    const x = coefficients[i] * normFactor\n    const y = (x + (x >= 0 ? 0.5 : -0.5)) | 0\n    out[i] = y > hi ? hi : y < lo ? lo : y\n  }\n\n  return out\n}\n\n/**\n * Dequantize coefficients.\n * @param {Int32Array} quantized\n * @param {number} scaleFactorIndex\n * @param {number} bitsPerSample\n * @returns {Float32Array}\n */\nexport function dequantize(quantized, scaleFactorIndex, bitsPerSample) {\n  if (bitsPerSample === 0 || scaleFactorIndex === 0) {\n    return new Float32Array(quantized.length)\n  }\n\n  const scaleFactor = SCALE_FACTORS[scaleFactorIndex]\n  const quantRange = (1 << (bitsPerSample - QUANTIZATION_SIGN_BIT_SHIFT)) - 1\n\n  const deq = new Float32Array(quantized.length)\n  for (let i = 0; i < quantized.length; i++) {\n    deq[i] = (quantized[i] * scaleFactor) / quantRange\n  }\n  return deq\n}\n\n/**\n * Groups MDCT coefficients into Band Frequency Units (BFUs) for bit allocation.\n *\n * This function organizes frequency-domain coefficients into perceptually meaningful\n * frequency bands (BFUs) based on the block modes for each of the three frequency bands.\n * It handles both long and short blocks, with different frequency resolutions and\n * BFU start positions for each block type.\n *\n * The function implements copy-avoidance optimization: when a BFU slice is fully\n * contained within a frequency band, it directly copies the data without buffer\n * filling. For BFUs that span band boundaries or extend beyond the available\n * coefficients, it handles partial copying and zero-padding appropriately.\n *\n * Frequency Band Structure:\n * - Band 0: 0-128 coefficients (low frequencies)\n * - Band 1: 128-256 coefficients (mid frequencies)\n * - Band 2: 256-512 coefficients (high frequencies)\n *\n * Block Types:\n * - Long blocks (mode 0): Higher frequency resolution, fewer temporal segments\n * - Short blocks (mode 1): Lower frequency resolution, more temporal segments\n *\n * @param {Float32Array} coefficients\n * @param {Array<number>} blockModes\n * @returns {{bfuData: Array<Float32Array>, bfuSizes: Array<number>, bfuCount: number}}\n */\nexport function groupIntoBFUs(coefficients, blockModes) {\n  const bfuData = []\n  const bfuSizes = []\n\n  let coeffIndex = 0\n  let bfuIndex = 0\n\n  for (let band = 0; band < 3; band++) {\n    const bandStart = coeffIndex\n    const bandSize = band === 2 ? 256 : 128\n    const bandEnd = band < 2 ? BFU_BAND_BOUNDARIES[band] : SPECS_PER_BFU.length\n    const isLongBlock = blockModes[band] === 0\n    const startPositions = isLongBlock ? BFU_START_LONG : BFU_START_SHORT\n\n    while (bfuIndex < bandEnd) {\n      const size = SPECS_PER_BFU[bfuIndex]\n      const startPos = startPositions[bfuIndex] - bandStart\n      const endPos = startPos + size\n      const bfu = new Float32Array(size)\n\n      if (startPos >= 0 && endPos <= bandSize) {\n        bfu.set(coefficients.subarray(bandStart + startPos, bandStart + endPos))\n      } else {\n        bfu.fill(0)\n        if (startPos < bandSize && endPos > 0) {\n          const srcStart = Math.max(0, startPos)\n          const srcEnd = Math.min(bandSize, endPos)\n          bfu.set(\n            coefficients.subarray(bandStart + srcStart, bandStart + srcEnd),\n            Math.max(0, -startPos)\n          )\n        }\n      }\n\n      bfuData.push(bfu)\n      bfuSizes.push(size)\n      bfuIndex++\n    }\n\n    coeffIndex += bandSize\n  }\n\n  return { bfuData, bfuSizes, bfuCount: bfuIndex }\n}\n","/**\n * Carta1 Audio Codec - Bit Allocation Module\n *\n * This module implements the bit allocation system for the Carta1 audio codec,\n * responsible for distributing available bits across frequency bands to minimize\n * perceptual distortion while maintaining the target bitrate.\n *\n * The module uses Rate-Distortion Optimization (RDO) with a greedy algorithm\n * that allocates bits to Band Frequency Units (BFUs) based on the perceptual\n * benefit per bit spent. It employs psychoacoustic modeling to determine\n * scale factors and uses a priority queue to efficiently find the optimal\n * bit distribution.\n *\n * Key components:\n * - RDO-based bit allocation using exponential distortion modeling\n * - Scale factor computation for each frequency band\n * - Max-heap priority queue for greedy optimization\n * - Support for variable word lengths per BFU\n *\n * The algorithm ensures that bits are allocated where they provide the\n * greatest reduction in audible distortion, resulting in high-quality\n * audio compression at the target bitrate.\n */\n\nimport {\n  NUM_BFUS,\n  FRAME_BITS,\n  FRAME_OVERHEAD_BITS,\n  BITS_PER_BFU_METADATA,\n  MAX_WORD_LENGTH_INDEX,\n  WORD_LENGTH_BITS,\n  INV_POWER_OF_TWO,\n  SCALE_FACTORS,\n} from '../core/constants.js'\n\n/**\n * Allocate bits to BFUs using optimized greedy Lp-RDO\n * @param {Array<Float32Array>} bfuData\n * @param {Int32Array} bfuSizes\n * @param {number} maxBfuCount\n * @param {number} allocationBias\n * @returns {{bfuCount:number, allocation:Int32Array, scaleFactorIndices:Int32Array}}\n */\nexport function allocateBits(bfuData, bfuSizes, maxBfuCount, allocationBias) {\n  const usedBFU = maxBfuCount\n  const avail =\n    FRAME_BITS - FRAME_OVERHEAD_BITS - usedBFU * BITS_PER_BFU_METADATA\n\n  const rdoResult = distributeBitsRDO(\n    usedBFU,\n    bfuData,\n    bfuSizes,\n    avail,\n    allocationBias\n  )\n\n  return {\n    bfuCount: usedBFU,\n    allocation: rdoResult.wordLengths,\n    scaleFactorIndices: rdoResult.scaleFactorIndices,\n  }\n}\n\n/**\n * Distributes available bits across Band Frequency Units (BFUs) using Rate-Distortion Optimization.\n *\n * This function implements an optimized greedy RDO allocator that minimizes perceptual distortion\n * while staying within the bit budget. It uses a priority queue (max-heap) to greedily allocate\n * bits to the BFUs that provide the greatest distortion reduction per bit spent.\n *\n * @param {number} activeBfuCount\n * @param {Array<Float32Array>} bfuData\n * @param {Int32Array} bfuSizes\n * @param {number} remainingBits\n * @param {number} allocationBias\n * @returns {{wordLengths: Int32Array, scaleFactorIndices: Int32Array}}\n */\nfunction distributeBitsRDO(\n  activeBfuCount,\n  bfuData,\n  bfuSizes,\n  remainingBits,\n  allocationBias\n) {\n  const scaleFactorTable = new Int32Array(NUM_BFUS)\n  const wordLengths = new Int32Array(activeBfuCount)\n\n  const heapIndices = []\n  const heapPriorities = []\n\n  const deltaDistPerCoeff = (bfuIndex, currentWl, nextWl) => {\n    const sfi = scaleFactorTable[bfuIndex]\n    const scaleFactor = SCALE_FACTORS[sfi]\n    if (scaleFactor === 0) return 0\n\n    const effectiveScaleFactor = Math.pow(scaleFactor, allocationBias)\n\n    const bits1 = WORD_LENGTH_BITS[currentWl] | 0\n    const bits2 = WORD_LENGTH_BITS[nextWl] | 0\n\n    const f1 = bits1 === 0 ? 2.0 : INV_POWER_OF_TWO[bits1]\n    const f2 = INV_POWER_OF_TWO[bits2]\n\n    return effectiveScaleFactor * (f1 - f2)\n  }\n\n  for (let bfuIndex = 0; bfuIndex < activeBfuCount; bfuIndex++) {\n    const sz = bfuSizes[bfuIndex] | 0\n    if (sz === 0) continue\n\n    const sfi = findScaleFactor(bfuData[bfuIndex].subarray(0, sz))\n    scaleFactorTable[bfuIndex] = sfi\n\n    if (sfi === 0) continue\n\n    const currentWl = 0\n    const nextWl = 1\n    const deltaBits =\n      (WORD_LENGTH_BITS[nextWl] - WORD_LENGTH_BITS[currentWl]) | 0\n    if (deltaBits <= 0) continue\n\n    const ddPerCoeff = deltaDistPerCoeff(bfuIndex, currentWl, nextWl)\n    heapIndices.push(bfuIndex)\n    heapPriorities.push(ddPerCoeff / deltaBits)\n  }\n\n  // Heapify\n  for (let i = (heapIndices.length >> 1) - 1; i >= 0; i--)\n    siftDown(heapIndices, heapPriorities, i)\n\n  // Greedy spending\n  while (remainingBits > 0 && heapIndices.length > 0) {\n    const bfu = heapIndices[0]\n    const cur = wordLengths[bfu] | 0\n    const nxt = cur + 1\n\n    const sz = bfuSizes[bfu] | 0\n    const deltaBits = (WORD_LENGTH_BITS[nxt] - WORD_LENGTH_BITS[cur]) | 0\n    const cost = deltaBits * sz\n\n    if (cost > remainingBits || cost <= 0) {\n      popRoot(heapIndices, heapPriorities)\n      continue\n    }\n\n    remainingBits -= cost\n    wordLengths[bfu] = nxt\n\n    if (nxt < MAX_WORD_LENGTH_INDEX) {\n      const nxt2 = nxt + 1\n      const deltaBits2 = (WORD_LENGTH_BITS[nxt2] - WORD_LENGTH_BITS[nxt]) | 0\n      if (deltaBits2 > 0) {\n        heapPriorities[0] = deltaDistPerCoeff(bfu, nxt, nxt2) / deltaBits2\n        siftDown(heapIndices, heapPriorities, 0)\n      } else {\n        popRoot(heapIndices, heapPriorities)\n      }\n    } else {\n      popRoot(heapIndices, heapPriorities)\n    }\n  }\n\n  return { wordLengths, scaleFactorIndices: scaleFactorTable }\n}\n\n/**\n * Find the optimal scale factor index for a set of coefficients.\n * Same semantics as the original.\n * @param {Float32Array} coefficients\n * @returns {number}\n */\nexport function findScaleFactor(coefficients) {\n  let maxAmplitude = 0.0\n  for (let i = 0; i < coefficients.length; i++) {\n    const a = Math.abs(coefficients[i])\n    if (a > maxAmplitude) maxAmplitude = a\n  }\n  if (maxAmplitude === 0) return 0\n  const index = Math.ceil(3 * (Math.log2(maxAmplitude) + 21))\n  return Math.max(0, Math.min(63, index))\n}\n\n/**\n * Removes the root element from a max-heap and maintains heap property.\n *\n * This function efficiently removes the maximum priority element from the heap\n * by replacing it with the last element and then restoring the heap property\n * through sifting down.\n *\n * @param {Array<number>} heapIndices\n * @param {Array<number>} heapPriorities\n */\nfunction popRoot(heapIndices, heapPriorities) {\n  const last = heapIndices.length - 1\n  heapIndices[0] = heapIndices[last]\n  heapPriorities[0] = heapPriorities[last]\n  heapIndices.pop()\n  heapPriorities.pop()\n  if (heapIndices.length) siftDown(heapIndices, heapPriorities, 0)\n}\n\n/**\n * Restores max-heap property by sifting an element down from a given position.\n *\n * This function maintains the heap invariant where parent nodes have higher\n * priority values than their children. It compares the element at startIndex\n * with its children and swaps with the larger child until the heap property\n * is satisfied, or it reaches a leaf position.\n *\n * @param {Array<number>} heapIndices\n * @param {Array<number>} heapPriorities\n * @param {number} startIndex\n */\nfunction siftDown(heapIndices, heapPriorities, startIndex) {\n  const heapSize = heapIndices.length\n  let i = startIndex\n  const idxVal = heapIndices[i]\n  const prVal = heapPriorities[i]\n\n  while (true) {\n    const l = (i << 1) + 1\n    const r = l + 1\n    let maxI = i\n    let maxP = prVal\n\n    if (l < heapSize && heapPriorities[l] > maxP) {\n      maxI = l\n      maxP = heapPriorities[l]\n    }\n    if (r < heapSize && heapPriorities[r] > maxP) {\n      maxI = r\n    }\n    if (maxI === i) break\n\n    heapIndices[i] = heapIndices[maxI]\n    heapPriorities[i] = heapPriorities[maxI]\n    i = maxI\n  }\n\n  heapIndices[i] = idxVal\n  heapPriorities[i] = prVal\n}\n","/**\n * Carta1 Audio Codec - Buffer Management\n */\n\nimport { QMF_DELAY, QMF_HIGH_BAND_DELAY } from './constants.js'\n\nexport class BufferPool {\n  constructor() {\n    // Transform buffers\n    this.transformBuffers = {\n      64: new Float32Array(64),\n      128: new Float32Array(128),\n      256: new Float32Array(256),\n      512: new Float32Array(512),\n    }\n\n    // QMF work buffers\n    this.qmfWorkBuffers = {\n      delay: {\n        128: new Float32Array(QMF_DELAY + 128),\n        256: new Float32Array(QMF_DELAY + 256),\n        512: new Float32Array(QMF_DELAY + 512),\n      },\n      highBandDelay: {\n        128: new Float32Array(QMF_HIGH_BAND_DELAY + 128),\n        256: new Float32Array(QMF_HIGH_BAND_DELAY + 256),\n      },\n    }\n\n    // QMF delay lines\n    this.qmfDelays = {\n      lowBand: new Float32Array(QMF_DELAY),\n      midBand: new Float32Array(QMF_DELAY),\n      highBand: new Float32Array(QMF_HIGH_BAND_DELAY),\n    }\n\n    // Transient detector\n    this.transientDetection = [\n      new Float32Array(64),\n      new Float32Array(64),\n      new Float32Array(128),\n    ]\n\n    // MDCT/IMDCT work buffers\n    this.mdctBuffers = {\n      16: {\n        real: new Float32Array(16),\n        imag: new Float32Array(16),\n      },\n      64: {\n        real: new Float32Array(64),\n        imag: new Float32Array(64),\n      },\n      128: {\n        real: new Float32Array(128),\n        imag: new Float32Array(128),\n      },\n    }\n\n    // MDCT overlap buffers (encoder)\n    this.mdctOverlap = [\n      new Float32Array(32),\n      new Float32Array(32),\n      new Float32Array(32),\n    ]\n\n    // IMDCT overlap buffers (decoder)\n    this.imdctOverlap = [\n      new Float32Array(256),\n      new Float32Array(256),\n      new Float32Array(512),\n    ]\n\n    // Spectrum reversal buffers\n    this.reversalBuffers = {\n      32: new Float32Array(32),\n      128: new Float32Array(128),\n      256: new Float32Array(256),\n    }\n  }\n}\n","/**\n * Carta1 Audio Codec - Transient Detection Module\n *\n * Detects transients using spectral flux analysis to prevent pre-echo artifacts.\n * Combines multiple spectral features to identify sudden changes in signal\n * characteristics that require special handling during encoding.\n */\n\nimport { FFT } from '../transforms/fft.js'\n\n/**\n * Perform FFT on time-domain samples and return magnitude spectrum\n * @param {Float32Array} samples - Time-domain samples\n * @param {number} fftSize - FFT size (must be power of 2)\n * @returns {Float32Array} Magnitude spectrum (positive frequencies only)\n */\nexport function performFFT(samples, fftSize) {\n  const real = new Float32Array(fftSize)\n  const imag = new Float32Array(fftSize)\n\n  // Copy samples with zero-padding if necessary\n  const copyLen = Math.min(samples.length, fftSize)\n  real.set(samples.subarray(0, copyLen))\n\n  // Perform FFT\n  FFT.fft(real, imag)\n\n  // Calculate magnitude spectrum for positive frequencies\n  const magnitude = new Float32Array(fftSize / 2)\n  for (let i = 0; i < fftSize / 2; i++) {\n    magnitude[i] = Math.sqrt(real[i] * real[i] + imag[i] * imag[i])\n  }\n\n  return magnitude\n}\n\n/**\n * Detect transients by analyzing spectral changes between frames\n * @param {Float32Array} currentCoeffs - Current frame coefficients\n * @param {Float32Array} prevCoeffs - Previous frame coefficients\n * @param {number} threshold - Detection threshold\n * @returns {boolean} True if transient detected\n */\nexport function detectTransient(currentCoeffs, prevCoeffs, threshold) {\n  // Can't detect transients without previous frame\n  if (!prevCoeffs) return false\n\n  // Calculate all spectral features\n  const features = calculateSpectralFeatures(currentCoeffs, prevCoeffs)\n\n  // Combine features using perceptually-motivated weighting\n  const transientScore = calculateTransientScore(features)\n\n  return transientScore > threshold\n}\n\n/**\n * Calculate all spectral features for transient detection\n * @param {Float32Array} currentCoeffs - Current frame coefficients\n * @param {Float32Array} prevCoeffs - Previous frame coefficients\n * @returns {Object} Computed spectral features\n */\nfunction calculateSpectralFeatures(currentCoeffs, prevCoeffs) {\n  // Calculate spectral flux (positive differences only)\n  const spectralFlux = calculateSpectralFlux(currentCoeffs, prevCoeffs)\n\n  // Calculate spectral flatness for both frames\n  const currentFlatness = calculateSpectralFlatness(currentCoeffs)\n  const prevFlatness = calculateSpectralFlatness(prevCoeffs)\n  const flatnessChange = Math.abs(currentFlatness - prevFlatness)\n\n  // Calculate high-frequency content ratio\n  const currentHfRatio = calculateHighFrequencyRatio(currentCoeffs)\n  const prevHfRatio = calculateHighFrequencyRatio(prevCoeffs)\n  const hfChange = Math.abs(currentHfRatio - prevHfRatio)\n\n  // Calculate energy change in dB\n  const energyChange = calculateEnergyChange(currentCoeffs, prevCoeffs)\n\n  return {\n    spectralFlux,\n    flatnessChange,\n    hfChange,\n    energyChange,\n  }\n}\n\n/**\n * Calculate normalized spectral flux (positive spectral differences)\n * High values indicate sudden onset of new spectral components\n */\nfunction calculateSpectralFlux(currentCoeffs, prevCoeffs) {\n  let flux = 0\n  let currentEnergy = 0\n\n  for (let i = 0; i < currentCoeffs.length; i++) {\n    const currentMag = Math.abs(currentCoeffs[i])\n    const prevMag = Math.abs(prevCoeffs[i])\n\n    // Only count positive differences (new energy)\n    const diff = currentMag - prevMag\n    if (diff > 0) {\n      flux += diff\n    }\n\n    currentEnergy += currentMag * currentMag\n  }\n\n  // Normalize by current frame energy to make threshold independent of level\n  const normalization = Math.sqrt(currentEnergy) || 1e-6\n  return flux / normalization\n}\n\n/**\n * Calculate spectral flatness (geometric mean / arithmetic mean)\n * Measures how noise-like vs tonal the spectrum is\n * @param {Float32Array} coeffs - Spectral coefficients\n * @returns {number} Flatness measure (0 = tonal, 1 = noise-like)\n */\nfunction calculateSpectralFlatness(coeffs) {\n  const EPSILON = 1e-10\n  let sumLog = 0\n  let sumLinear = 0\n  let validBins = 0\n\n  for (let i = 0; i < coeffs.length; i++) {\n    const magnitude = Math.abs(coeffs[i])\n    if (magnitude > EPSILON) {\n      sumLog += Math.log(magnitude)\n      sumLinear += magnitude\n      validBins++\n    }\n  }\n\n  if (validBins === 0) return 0\n\n  const geometricMean = Math.exp(sumLog / validBins)\n  const arithmeticMean = sumLinear / validBins\n\n  return arithmeticMean > EPSILON ? geometricMean / arithmeticMean : 0\n}\n\n/**\n * Calculate ratio of high-frequency to total energy\n * High values indicate bright/harsh sounds that are perceptually important\n * @param {Float32Array} coeffs - Spectral coefficients\n * @returns {number} High-frequency energy ratio (0-1)\n */\nfunction calculateHighFrequencyRatio(coeffs) {\n  const midPoint = Math.floor(coeffs.length / 2)\n  let lowEnergy = 0\n  let highEnergy = 0\n\n  for (let i = 0; i < midPoint; i++) {\n    lowEnergy += coeffs[i] * coeffs[i]\n  }\n\n  for (let i = midPoint; i < coeffs.length; i++) {\n    highEnergy += coeffs[i] * coeffs[i]\n  }\n\n  const totalEnergy = lowEnergy + highEnergy\n  return totalEnergy > 0 ? highEnergy / totalEnergy : 0\n}\n\n/**\n * Calculate energy change between frames in dB\n * @param {Float32Array} currentCoeffs - Current frame coefficients\n * @param {Float32Array} prevCoeffs - Previous frame coefficients\n * @returns {number} Energy change in dB (positive values only)\n */\nfunction calculateEnergyChange(currentCoeffs, prevCoeffs) {\n  let currentEnergy = 0\n  let prevEnergy = 0\n\n  for (let i = 0; i < currentCoeffs.length; i++) {\n    currentEnergy += currentCoeffs[i] * currentCoeffs[i]\n    prevEnergy += prevCoeffs[i] * prevCoeffs[i]\n  }\n\n  // Avoid log of zero and ensure minimum energy\n  currentEnergy = Math.max(currentEnergy, 1e-10)\n  prevEnergy = Math.max(prevEnergy, 1e-10)\n\n  const changeDb = 10 * Math.log10(currentEnergy / prevEnergy)\n\n  // Only interested in energy increases (potential transients)\n  return Math.max(0, changeDb)\n}\n\n/**\n * Calculate unified transient score from spectral features\n * Uses perceptually-motivated combination instead of magic numbers\n * @param {Object} features - Spectral features\n * @returns {number} Transient score\n */\nfunction calculateTransientScore(features) {\n  const { spectralFlux, flatnessChange, hfChange, energyChange } = features\n\n  // Spectral flux is the primary indicator - it directly measures\n  // sudden onset of new frequency components\n  const fluxContribution = spectralFlux\n\n  // Flatness change indicates transition between tonal and noise-like\n  // signals (e.g., pitched to unpitched percussion)\n  // Scale by sqrt to reduce dominance\n  const flatnessContribution = Math.sqrt(flatnessChange)\n\n  // High-frequency changes are perceptually important but shouldn't\n  // dominate the decision. Use log scaling for perceptual relevance\n  const hfContribution = Math.log1p(hfChange * 10) / Math.log1p(10)\n\n  // Energy changes in dB are already perceptually scaled\n  // Normalize to 0-1 range assuming 30dB is a strong transient\n  const energyContribution = Math.min(energyChange / 30, 1)\n\n  // Combine features with equal perceptual importance\n  // Each contribution is normalized to roughly 0-1 range\n  return (\n    (fluxContribution +\n      flatnessContribution +\n      hfContribution +\n      energyContribution) /\n    4\n  )\n}\n","/**\n * Carta1 Audio Codec - Encoding Pipeline\n *\n * The encoding pipeline transforms PCM audio samples into ATRAC1 compressed format\n * through a series of processing stages:\n *\n * 1. QMF Analysis: Splits audio into frequency bands using quadrature mirror filters\n * 2. Block Selection: Detects transients to choose appropriate transform block sizes\n * 3. MDCT Transform: Applies Modified Discrete Cosine Transform to frequency bands\n * 4. Quantization: Performs psychoacoustic analysis, bit allocation, and quantization\n *\n * Each stage is implemented as a functional pipeline component that can be composed\n * using the pipe utility. The pipeline maintains state through a shared context\n * containing buffer pools and encoding options.\n */\n\nimport {\n  pipe,\n  throwError,\n  reverseSpectrum,\n  calculateBandOffset,\n} from '../utils.js'\nimport { qmfAnalysis } from '../transforms/qmf.js'\nimport { mdct64, mdct256, mdct512 } from '../transforms/mdct.js'\nimport { groupIntoBFUs, quantize } from '../coding/quantization.js'\nimport { allocateBits } from '../coding/bitallocation.js'\nimport { BufferPool } from '../core/buffers.js'\nimport { EncoderOptions } from '../core/options.js'\nimport {\n  WORD_LENGTH_BITS,\n  WINDOW_SHORT,\n  FFT_SIZE_LOW,\n  FFT_SIZE_MID,\n  FFT_SIZE_HIGH,\n  MDCT_SIZE_SHORT,\n  MDCT_SIZE_MID,\n  MDCT_SIZE_LONG,\n  MDCT_BAND_CONFIGS,\n  MDCT_SHORT_BLOCK_SIZE,\n  MDCT_OVERLAP_SIZE,\n} from '../core/constants.js'\nimport { performFFT, detectTransient } from '../analysis/transient.js'\n\n/**\n * QMF Analysis Stage - Splits audio into frequency bands\n *\n * Performs two-stage QMF analysis to decompose the input signal into three bands:\n * - Low band (0-5.5kHz): processed at full resolution\n * - Mid band (5.5-11kHz): processed at half resolution\n * - High band (11-22kHz): processed at quarter resolution with delay compensation\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes PCM samples\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function qmfAnalysisStage(context) {\n  const bufferPool =\n    context?.bufferPool ??\n    throwError('qmfAnalysisStage: bufferPool is required')\n  const delays = bufferPool.qmfDelays\n\n  /**\n   * Process PCM samples through QMF analysis\n   * @param {Float32Array} pcmSamples - Input PCM samples\n   * @returns {Object} Analysis result containing frequency bands\n   * @returns {Array<Float32Array>} returns.bands - Three frequency bands [low, mid, high]\n   */\n  return (pcmSamples) => {\n    const stage1 = qmfAnalysis(\n      pcmSamples,\n      delays.lowBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.lowBand = stage1.newDelay\n\n    const stage2 = qmfAnalysis(\n      stage1.lowBand,\n      delays.midBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.midBand = stage2.newDelay\n\n    const delayedHigh =\n      bufferPool.qmfWorkBuffers.highBandDelay[stage1.highBand.length]\n    delayedHigh.set(delays.highBand)\n    delayedHigh.set(stage1.highBand, delays.highBand.length)\n\n    const highBand = delayedHigh.slice(0, stage1.highBand.length)\n    delays.highBand = delayedHigh.slice(stage1.highBand.length)\n\n    return {\n      bands: [stage2.lowBand, stage2.highBand, highBand],\n    }\n  }\n}\n\n/**\n * Block Selection Stage - Detects transients to choose transform block sizes\n *\n * Analyzes frequency bands using FFT and transient detection to determine\n * appropriate MDCT block sizes. Short blocks are used for transient signals\n * to prevent pre-echo artifacts, while long blocks provide better frequency\n * resolution for steady-state signals.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes QMF analysis results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function blockSelectorStage(context) {\n  const bufferPool =\n    context?.bufferPool ??\n    throwError('blockSelectorStage: bufferPool is required')\n  const options =\n    context?.options ?? throwError('blockSelectorStage: options is required')\n\n  /**\n   * Analyze frequency bands and determine block modes\n   * @param {Object} input - QMF analysis results\n   * @param {Array<Float32Array>} input.bands - Three frequency bands [low, mid, high]\n   * @returns {Object} Block selection results\n   * @returns {Array<Float32Array>} returns.bands - Original frequency bands\n   * @returns {Array<number>} returns.blockModes - Block modes for each band\n   */\n  return (input) => {\n    const { bands } = input\n    const fftSizes = [FFT_SIZE_LOW, FFT_SIZE_MID, FFT_SIZE_HIGH]\n    const blockModes = bands.map((bandSamples, bandIndex) => {\n      const coeffs = performFFT(bandSamples, fftSizes[bandIndex])\n      const transient = detectTransient(\n        coeffs,\n        bufferPool.transientDetection[bandIndex],\n        options.transientThresholdLow\n      )\n      bufferPool.transientDetection[bandIndex] = coeffs\n      return transient * Math.max(bandIndex + 1, 2)\n    })\n\n    return {\n      bands,\n      blockModes,\n    }\n  }\n}\n\n/**\n * MDCT Transform Stage - Applies Modified Discrete Cosine Transform\n *\n * Transforms time-domain samples to frequency-domain coefficients using MDCT.\n * The transform size depends on the block mode determined by transient detection:\n * - Long blocks: 512/256 samples for better frequency resolution\n * - Short blocks: 64 samples for better time resolution\n *\n * Applies windowing and overlap-add processing to prevent blocking artifacts.\n * Mid and high bands are spectrally reversed to match ATRAC1 format.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes block selection results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function mdctStage(context) {\n  const bufferPool =\n    context?.bufferPool ?? throwError('mdctStage: bufferPool is required')\n  const overlapBuffers = bufferPool.mdctOverlap\n\n  // Transform function mapping for each band\n  const TRANSFORM_FUNCS = [mdct256, mdct256, mdct512]\n\n  /**\n   * Transform a single frequency band using MDCT.\n   * @param {Float32Array} samples\n   * @param {number} bandIndex\n   * @param {number} blockMode\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @returns {Float32Array}\n   */\n  function transformBand(\n    samples,\n    bandIndex,\n    blockMode,\n    overlapBuffer,\n    bufferPool\n  ) {\n    const config = MDCT_BAND_CONFIGS[bandIndex]\n    const transformFunc = TRANSFORM_FUNCS[bandIndex]\n    const isLongBlock = blockMode === 0\n\n    if (isLongBlock) {\n      return transformLongBlock(\n        samples,\n        bandIndex,\n        config,\n        transformFunc,\n        overlapBuffer,\n        bufferPool\n      )\n    } else {\n      return transformShortBlocks(\n        samples,\n        bandIndex,\n        config,\n        overlapBuffer,\n        bufferPool\n      )\n    }\n  }\n\n  /**\n   * Transform using long block (better frequency resolution).\n   * @param {Float32Array} samples\n   * @param {number} bandIndex\n   * @param {Object} config\n   * @param {Object} transformFunc\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @returns {Float32Array}\n   */\n  function transformLongBlock(\n    samples,\n    bandIndex,\n    config,\n    transformFunc,\n    overlapBuffer,\n    bufferPool\n  ) {\n    const mdctSize = bandIndex === 2 ? MDCT_SIZE_LONG : MDCT_SIZE_MID\n    const mdctInput = bufferPool.transformBuffers[mdctSize]\n    mdctInput.fill(0)\n\n    // Apply overlap from previous frame\n    mdctInput.set(overlapBuffer, config.windowStart)\n\n    // Window and save tail for next frame\n    applyTailWindowing(samples, overlapBuffer, config.size)\n\n    // Add current frame samples\n    mdctInput.set(samples, config.windowStart + MDCT_OVERLAP_SIZE)\n\n    // Transform\n    let spectrum = transformFunc.transform(mdctInput, bufferPool.mdctBuffers)\n\n    // Apply spectral reversal for mid/high bands\n    if (bandIndex > 0) {\n      spectrum = reverseSpectrum(spectrum, bufferPool.reversalBuffers)\n    }\n\n    return spectrum\n  }\n\n  /**\n   * Transform using short blocks (better time resolution for transients).\n   * @param {Float32Array} samples\n   * @param {number} bandIndex\n   * @param {Object} config\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @returns {Float32Array}\n   */\n  function transformShortBlocks(\n    samples,\n    bandIndex,\n    config,\n    overlapBuffer,\n    bufferPool\n  ) {\n    const numBlocks = 1 << (config.size === 256 ? 3 : 2) // 8 for band 2, 4 for bands 0-1\n    const output = new Float32Array(config.size)\n\n    for (let block = 0; block < numBlocks; block++) {\n      const blockStart = block * MDCT_SHORT_BLOCK_SIZE\n      const blockSamples = samples.subarray(\n        blockStart,\n        blockStart + MDCT_SHORT_BLOCK_SIZE\n      )\n\n      // Prepare MDCT input\n      const mdctInput = bufferPool.transformBuffers[MDCT_SIZE_SHORT]\n      mdctInput.fill(0)\n      mdctInput.set(overlapBuffer, 0)\n\n      // Window and save tail\n      applyTailWindowing(blockSamples, overlapBuffer, MDCT_SHORT_BLOCK_SIZE)\n      mdctInput.set(blockSamples, MDCT_OVERLAP_SIZE)\n\n      // Transform\n      let spectrum = mdct64.transform(mdctInput, bufferPool.mdctBuffers)\n\n      // Apply spectral reversal for mid/high bands\n      if (bandIndex > 0) {\n        spectrum = reverseSpectrum(spectrum, bufferPool.reversalBuffers)\n      }\n\n      output.set(spectrum, blockStart)\n    }\n\n    return output\n  }\n\n  function applyTailWindowing(samples, overlapBuffer, blockSize) {\n    const tailStart = blockSize - MDCT_OVERLAP_SIZE\n    for (let i = 0; i < MDCT_OVERLAP_SIZE; i++) {\n      const tailValue = samples[tailStart + i]\n      overlapBuffer[i] = WINDOW_SHORT[i] * tailValue\n      samples[tailStart + i] = tailValue * WINDOW_SHORT[31 - i]\n    }\n  }\n\n  /**\n   * Transform frequency bands using MDCT\n   * @param {Object} input - Block selection results\n   * @param {Array<Float32Array>} input.bands - Three frequency bands\n   * @param {Array<number>} input.blockModes - Block modes for each band\n   * @param {any} input.originalFrame - Original frame data (passed through)\n   * @returns {Object} MDCT transform results\n   * @returns {Array<Float32Array>} returns.bands - Original frequency bands\n   * @returns {Float32Array} returns.coefficients - MDCT coefficients (512 samples)\n   * @returns {Array<number>} returns.blockModes - Block modes\n   * @returns {any} returns.originalFrame - Original frame data\n   */\n  return (input) => {\n    const { bands, blockModes, originalFrame } = input\n    const coefficients = new Float32Array(512)\n\n    bands.forEach((bandSamples, bandIndex) => {\n      const transformed = transformBand(\n        bandSamples,\n        bandIndex,\n        blockModes[bandIndex],\n        overlapBuffers[bandIndex],\n        bufferPool\n      )\n\n      const offset = calculateBandOffset(bandIndex)\n      coefficients.set(transformed, offset)\n    })\n\n    return { bands, coefficients, blockModes, originalFrame }\n  }\n}\n\n/**\n * Quantization Stage - Performs RDO bit allocation and quantization\n *\n * The final encoding stage that:\n * 1. Groups coefficients into Block Floating Units (BFUs)\n * 2. Allocates bits using Rate-Distortion Optimization (RDO)\n * 3. Quantizes coefficients using scale factors and word lengths\n *\n * Uses RDO to optimize the allocation of bits for maximum quality\n * within the available bit budget.\n *\n * @returns {Function} Stage function that processes MDCT results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function quantizationStage(context) {\n  const options =\n    context?.options ?? throwError('quantizationStage: options is required')\n\n  /**\n   * Perform RDO bit allocation and quantization\n   * @param {Object} input - MDCT transform results\n   * @param {Float32Array} input.coefficients - MDCT coefficients\n   * @param {Array<number>} input.blockModes - Block modes\n   * @returns {Object} Quantization results ready for bitstream encoding\n   * @returns {number} returns.nBfu - Number of active BFUs\n   * @returns {Int32Array} returns.scaleFactorIndices - Scale factor indices for each BFU\n   * @returns {Int32Array} returns.wordLengthIndices - Word length indices for each BFU\n   * @returns {Array<Int32Array>} returns.quantizedCoefficients - Quantized coefficient data\n   * @returns {Array<number>} returns.blockModes - Block modes for each band\n   */\n  return (input) => {\n    const { coefficients, blockModes } = input\n\n    const { bfuData, bfuSizes, bfuCount } = groupIntoBFUs(\n      coefficients,\n      blockModes\n    )\n\n    const {\n      bfuCount: selectedBfuCount,\n      allocation,\n      scaleFactorIndices,\n    } = allocateBits(bfuData, bfuSizes, bfuCount, options.allocationBias)\n\n    const quantizedCoefficients = []\n\n    for (let bfu = 0; bfu < selectedBfuCount; bfu++) {\n      const data = bfuData[bfu].subarray(0, bfuSizes[bfu])\n      const wordLength = allocation[bfu]\n      const bitsPerSample = WORD_LENGTH_BITS[wordLength]\n      const quantized = quantize(data, scaleFactorIndices[bfu], bitsPerSample)\n      quantizedCoefficients.push(quantized)\n    }\n\n    return {\n      nBfu: selectedBfuCount,\n      scaleFactorIndices,\n      wordLengthIndices: allocation.slice(0, selectedBfuCount),\n      quantizedCoefficients,\n      blockModes,\n    }\n  }\n}\n\n/**\n * Create ATRAC1 encoding pipeline\n *\n * Constructs a complete encoding pipeline that transforms PCM audio samples\n * into ATRAC1 compressed format. The pipeline processes audio through QMF\n * analysis, transient detection, MDCT transform, and psychoacoustic quantization.\n *\n * The returned function can be called repeatedly to encode audio frames,\n * maintaining state through the shared buffer pool for efficient processing.\n *\n * @param {EncoderOptions} [options=new EncoderOptions()] - Encoding configuration\n * @param {BufferPool} [bufferPool=new BufferPool()] - Shared buffer pool for state\n * @returns {Function} Encoding pipeline function that processes PCM samples\n *\n * @example\n * const encoder = encode(new EncoderOptions(), new BufferPool())\n * const result = encoder(pcmSamples) // Returns quantized ATRAC1 frame data\n */\nexport function encode(\n  options = new EncoderOptions(),\n  bufferPool = new BufferPool()\n) {\n  const context = { options, bufferPool }\n  return pipe(\n    context,\n    qmfAnalysisStage,\n    blockSelectorStage,\n    mdctStage,\n    quantizationStage\n  )\n}\n","/**\n * Carta1 Audio Codec - Decoding Pipeline\n *\n * The decoding pipeline transforms ATRAC1 compressed format back into PCM audio samples\n * through a series of processing stages:\n *\n * 1. Dequantization: Reconstructs MDCT coefficients from quantized data using scale factors\n * 2. Inverse MDCT: Applies Inverse Modified Discrete Cosine Transform to frequency bands\n * 3. QMF Synthesis: Reconstructs full-spectrum audio using quadrature mirror filters\n *\n * Each stage is implemented as a functional pipeline component that can be composed\n * using the pipe utility. The pipeline maintains state through a shared context\n * containing buffer pools and decoding options.\n */\n\nimport {\n  pipe,\n  throwError,\n  reverseSpectrum,\n  extractBandCoefficients,\n} from '../utils.js'\nimport { qmfSynthesis } from '../transforms/qmf.js'\nimport { imdct64, imdct256, imdct512, overlapAdd } from '../transforms/mdct.js'\nimport { dequantize } from '../coding/quantization.js'\nimport { BufferPool } from '../core/buffers.js'\nimport {\n  BFU_BAND_BOUNDARIES,\n  BFU_START_LONG,\n  BFU_START_SHORT,\n  WORD_LENGTH_BITS,\n  WINDOW_SHORT,\n  MDCT_SIZE_LONG,\n  MDCT_BAND_CONFIGS,\n  MDCT_SHORT_BLOCK_SIZE,\n  MDCT_TAIL_WINDOW_SIZE,\n} from '../core/constants.js'\n\n/**\n * Dequantization Stage - Reconstructs MDCT coefficients from quantized data\n *\n * Reverses the quantization process by:\n * - Reconstructing coefficient values using scale factors and word lengths\n * - Organizing coefficients back into Block Floating Units (BFUs)\n * - Preparing coefficient array for inverse MDCT transform\n *\n * Uses the same BFU organization as the encoder but reconstructs the original\n * coefficient values from the quantized representation.\n *\n * @returns {Function} Stage function that processes quantized frame data\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function dequantizationStage() {\n  /**\n   * Reconstruct MDCT coefficients from quantized data\n   * @param {Object} frameData - Quantized frame data\n   * @param {number} frameData.nBfu - Number of active BFUs\n   * @param {Int32Array} frameData.scaleFactorIndices - Scale factor indices for each BFU\n   * @param {Int32Array} frameData.wordLengthIndices - Word length indices for each BFU\n   * @param {Array<Int32Array>} frameData.quantizedCoefficients - Quantized coefficient data\n   * @param {Array<number>} frameData.blockModes - Block modes for each band\n   * @returns {Object} Dequantization results\n   * @returns {Float32Array} returns.coefficients - Reconstructed MDCT coefficients (512 samples)\n   * @returns {Array<number>} returns.blockModes - Block modes for each band\n   */\n  return (frameData) => {\n    const coefficients = new Float32Array(512)\n    const {\n      nBfu,\n      scaleFactorIndices,\n      wordLengthIndices,\n      quantizedCoefficients,\n      blockModes,\n    } = frameData\n\n    for (let bfu = 0; bfu < nBfu; bfu++) {\n      const bitsPerSample = WORD_LENGTH_BITS[wordLengthIndices[bfu]]\n\n      let band = 0\n      if (bfu >= BFU_BAND_BOUNDARIES[0]) band = 1\n      if (bfu >= BFU_BAND_BOUNDARIES[1]) band = 2\n\n      const isLongBlock = blockModes[band] === 0\n      const position = isLongBlock ? BFU_START_LONG[bfu] : BFU_START_SHORT[bfu]\n\n      if (bitsPerSample > 0) {\n        const quantized = quantizedCoefficients[bfu]\n        const dequantized = dequantize(\n          quantized,\n          scaleFactorIndices[bfu],\n          bitsPerSample\n        )\n        coefficients.set(dequantized, position)\n      }\n    }\n\n    return { coefficients, blockModes }\n  }\n}\n\n/**\n * Inverse MDCT Stage - Transforms frequency coefficients back to time domain\n *\n * Applies Inverse Modified Discrete Cosine Transform to reconstruct time-domain\n * samples from frequency coefficients. The transform size depends on block modes:\n * - Long blocks: 512/256 samples for steady-state signals\n * - Short blocks: 64 samples for transient signals\n *\n * Applies windowing and overlap-add processing to prevent blocking artifacts.\n * Mid and high bands are spectrally unreversed from ATRAC1 format.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes dequantization results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function imdctStage(context) {\n  const bufferPool =\n    context?.bufferPool ?? throwError('imdctStage: bufferPool is required')\n  const overlapBuffers = bufferPool.imdctOverlap\n\n  // Transform function mapping for each band\n  const TRANSFORM_FUNCS = [imdct256, imdct256, imdct512]\n\n  /**\n   * Inverse transform a single frequency band.\n   * @param {Float32Array} coeffs\n   * @param {number} bandIndex\n   * @param {number} blockMode\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @param {Object} config\n   * @returns {Float32Array}\n   */\n  function transformBand(\n    coeffs,\n    bandIndex,\n    blockMode,\n    overlapBuffer,\n    bufferPool,\n    config\n  ) {\n    const isLongBlock = blockMode === 0\n    const transformFunc = TRANSFORM_FUNCS[bandIndex]\n\n    if (isLongBlock) {\n      return inverseLongBlock(\n        coeffs,\n        bandIndex,\n        overlapBuffer,\n        bufferPool,\n        config,\n        transformFunc\n      )\n    } else {\n      return inverseShortBlocks(\n        coeffs,\n        bandIndex,\n        overlapBuffer,\n        bufferPool,\n        config\n      )\n    }\n  }\n\n  /**\n   * Inverse transform using long block.\n   * @param {Float32Array} coeffs\n   * @param {number} bandIndex\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @param {Object} config\n   * @param {Object} transformFunc\n   * @returns {Float32Array}\n   */\n  function inverseLongBlock(\n    coeffs,\n    bandIndex,\n    overlapBuffer,\n    bufferPool,\n    config,\n    transformFunc\n  ) {\n    // Unreverse spectral reversal for mid/high bands\n    let blockSpecs = coeffs\n    if (bandIndex > 0) {\n      blockSpecs = reverseSpectrum(coeffs, bufferPool.reversalBuffers)\n    }\n\n    // Apply inverse transform\n    const inv = transformFunc.transform(blockSpecs, bufferPool.mdctBuffers)\n    const invStart = inv.length / 4\n\n    // Prepare output buffer\n    const invBuf = bufferPool.transformBuffers[MDCT_SIZE_LONG]\n    invBuf.fill(0)\n\n    // Copy transformed samples\n    for (let i = 0; i < config.size; i++) {\n      invBuf[i] = inv[invStart + i]\n    }\n\n    // Handle overlap-add with previous frame\n    const prevBuf = overlapBuffer.slice(\n      config.size * 2 - MDCT_TAIL_WINDOW_SIZE,\n      config.size * 2\n    )\n    const dstPart = overlapAdd(\n      prevBuf.slice(0, MDCT_TAIL_WINDOW_SIZE),\n      invBuf.slice(0, MDCT_TAIL_WINDOW_SIZE),\n      WINDOW_SHORT\n    )\n\n    overlapBuffer.set(dstPart, 0)\n\n    // Save tail for next frame\n    for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n      prevBuf[i] = invBuf[MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    // Copy main samples\n    const copyLen = bandIndex === 2 ? 240 : 112\n    for (let i = 0; i < copyLen; i++) {\n      overlapBuffer[32 + i] = invBuf[MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    // Copy final samples\n    for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n      overlapBuffer[config.size * 2 - MDCT_TAIL_WINDOW_SIZE + i] =\n        invBuf[config.size - MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    return overlapBuffer.slice(0, config.size)\n  }\n\n  /**\n   * Inverse transform using short blocks.\n   * @param {Float32Array} coeffs\n   * @param {number} bandIndex\n   * @param {Float32Array} overlapBuffer\n   * @param {Object} bufferPool\n   * @param {Object} config\n   * @returns {Float32Array}\n   */\n  function inverseShortBlocks(\n    coeffs,\n    bandIndex,\n    overlapBuffer,\n    bufferPool,\n    config\n  ) {\n    const numBlocks = 1 << (config.size === 256 ? 3 : 2)\n    const invBuf = bufferPool.transformBuffers[MDCT_SIZE_LONG]\n    invBuf.fill(0)\n\n    const prevBuf = overlapBuffer.slice(\n      config.size * 2 - MDCT_TAIL_WINDOW_SIZE,\n      config.size * 2\n    )\n\n    let start = 0\n    let pos = 0\n\n    for (let block = 0; block < numBlocks; block++) {\n      // Extract block coefficients\n      let blockSpecs = coeffs.slice(pos, pos + MDCT_SHORT_BLOCK_SIZE)\n\n      // Unreverse spectral reversal for mid/high bands\n      if (bandIndex > 0) {\n        blockSpecs = reverseSpectrum(blockSpecs, bufferPool.reversalBuffers)\n      }\n\n      // Apply inverse transform\n      const inv = imdct64.transform(blockSpecs, bufferPool.mdctBuffers)\n      const invStart = inv.length / 4\n\n      // Copy to intermediate buffer\n      for (let i = 0; i < MDCT_SHORT_BLOCK_SIZE; i++) {\n        invBuf[start + i] = inv[invStart + i]\n      }\n\n      // Overlap-add with previous block\n      const dstPart = overlapAdd(\n        prevBuf.slice(0, MDCT_TAIL_WINDOW_SIZE),\n        invBuf.slice(start, start + MDCT_TAIL_WINDOW_SIZE),\n        WINDOW_SHORT\n      )\n\n      overlapBuffer.set(dstPart, start)\n\n      // Save tail for next block\n      for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n        prevBuf[i] = invBuf[start + MDCT_TAIL_WINDOW_SIZE + i]\n      }\n\n      start += MDCT_SHORT_BLOCK_SIZE\n      pos += MDCT_SHORT_BLOCK_SIZE\n    }\n\n    // Copy final tail samples\n    for (let i = 0; i < MDCT_TAIL_WINDOW_SIZE; i++) {\n      overlapBuffer[config.size * 2 - MDCT_TAIL_WINDOW_SIZE + i] =\n        invBuf[config.size - MDCT_TAIL_WINDOW_SIZE + i]\n    }\n\n    return overlapBuffer.slice(0, config.size)\n  }\n\n  /**\n   * Transform coefficients back to time domain using inverse MDCT\n   * @param {Object} input - Dequantization results\n   * @param {Float32Array} input.coefficients - MDCT coefficients (512 samples)\n   * @param {Array<number>} input.blockModes - Block modes for each band\n   * @returns {Array<Float32Array>} Three reconstructed frequency bands [low, mid, high]\n   */\n  return (input) => {\n    const { coefficients, blockModes } = input\n\n    return MDCT_BAND_CONFIGS.map((config, bandIndex) => {\n      const bandCoeffs = extractBandCoefficients(coefficients, bandIndex)\n      return transformBand(\n        bandCoeffs,\n        bandIndex,\n        blockModes[bandIndex],\n        overlapBuffers[bandIndex],\n        bufferPool,\n        config\n      )\n    })\n  }\n}\n\n/**\n * QMF Synthesis Stage - Reconstructs full-spectrum audio from frequency bands\n *\n * Performs two-stage QMF synthesis to reconstruct the full-spectrum signal\n * from three frequency bands:\n * - Low band (0-5.5kHz): processed at full resolution\n * - Mid band (5.5-11kHz): processed at half resolution\n * - High band (11-22kHz): processed at quarter resolution with delay compensation\n *\n * This is the inverse operation of the QMF analysis performed during encoding,\n * combining the separate frequency bands back into a single wideband signal.\n *\n * @param {Object} context - Pipeline context containing bufferPool\n * @param {BufferPool} context.bufferPool - Shared buffer pool for efficient memory management\n * @returns {Function} Stage function that processes inverse MDCT results\n * @throws {Error} If bufferPool is not provided in context\n */\nexport function qmfSynthesisStage(context) {\n  const bufferPool =\n    context?.bufferPool ??\n    throwError('qmfSynthesisStage: bufferPool is required')\n  const delays = bufferPool.qmfDelays\n\n  /**\n   * Reconstruct full-spectrum PCM samples from frequency bands\n   * @param {Array<Float32Array>} bands - Three frequency bands [low, mid, high]\n   * @returns {Float32Array} Reconstructed PCM samples\n   */\n  return (bands) => {\n    // Apply high band delay compensation\n    const delayedHigh = bufferPool.qmfWorkBuffers.highBandDelay[bands[2].length]\n    delayedHigh.set(delays.highBand)\n    delayedHigh.set(bands[2], delays.highBand.length)\n\n    const highBand = delayedHigh.slice(0, bands[0].length * 2)\n    delays.highBand = delayedHigh.slice(bands[0].length * 2)\n\n    // Second stage synthesis: combine low and mid bands\n    const stage2 = qmfSynthesis(\n      bands[0],\n      bands[1],\n      delays.midBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.midBand = stage2.newDelay\n\n    // First stage synthesis: combine result with high band\n    const stage1 = qmfSynthesis(\n      stage2.output,\n      highBand,\n      delays.lowBand,\n      bufferPool.qmfWorkBuffers\n    )\n    delays.lowBand = stage1.newDelay\n\n    return stage1.output\n  }\n}\n\n/**\n * Create ATRAC1 decoding pipeline\n *\n * Constructs a complete decoding pipeline that transforms ATRAC1 compressed format\n * back into PCM audio samples. The pipeline processes quantized data through\n * dequantization, inverse MDCT transform, and QMF synthesis.\n *\n * The returned function can be called repeatedly to decode audio frames,\n * maintaining state through the shared buffer pool for efficient processing.\n *\n * @param {BufferPool} [bufferPool=new BufferPool()] - Shared buffer pool for state\n * @returns {Function} Decoding pipeline function that processes quantized frame data\n *\n * @example\n * const decoder = decode(new BufferPool())\n * const result = decoder(quantizedFrame) // Returns reconstructed PCM samples\n */\nexport function decode(bufferPool = new BufferPool()) {\n  const context = { bufferPool }\n  return pipe(context, dequantizationStage, imdctStage, qmfSynthesisStage)\n}\n","/**\n * Carta1 Audio Codec - Audio Processing Module\n *\n * High-level audio processing interface that provides stream-based encoding and decoding\n * for ATRAC1 audio format. Handles both mono and stereo audio processing with proper\n * delay compensation and frame management.\n *\n * Key features:\n * - Stream-based processing for large audio files\n * - Automatic delay compensation for codec processing\n * - Support for mono and stereo audio\n * - Progress tracking callbacks\n * - AEA file format creation and parsing\n * - WAV file generation from decoded audio\n */\n\nimport { encode } from '../pipeline/encoder.js'\nimport { decode } from '../pipeline/decoder.js'\nimport { EncoderOptions } from '../core/options.js'\nimport {\n  SAMPLES_PER_FRAME,\n  SAMPLE_RATE,\n  AEA_HEADER_SIZE,\n  SOUND_UNIT_SIZE,\n  WAV_HEADER_SIZE,\n  WAV_BYTES_PER_SAMPLE,\n  WAV_BITS_PER_SAMPLE,\n  WAV_PCM_MAX_POSITIVE,\n  WAV_PCM_MAX_NEGATIVE,\n  WAV_DATA_OFFSET,\n} from '../core/constants.js'\nimport { serializeFrame, deserializeFrame, AeaFile } from './serialization.js'\n\n/**\n * Audio processing utility class providing high-level encoding and decoding operations\n */\nexport class AudioProcessor {\n  /**\n   * Encodes a stream of audio frames to ATRAC1 format\n   * @param {AsyncIterable<Float32Array>|AsyncIterable<[Float32Array, Float32Array]>} audioFrames - Audio frame stream\n   * @param {Object} [options={}] - Encoding options\n   * @param {number} [options.channelCount=1] - Number of audio channels (1 or 2)\n   * @param {Function} [options.onProgress] - Progress callback function\n   * @param {EncoderOptions} [options.encoderOptions] - Encoder configuration options\n   * @returns {AsyncGenerator<Object>} Stream of encoded frame data\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static async *encodeStream(audioFrames, options = {}) {\n    const { channelCount = 1, onProgress, encoderOptions } = options\n\n    if (channelCount === 1) {\n      yield* AudioProcessor._encodeMonoStream(\n        audioFrames,\n        onProgress,\n        encoderOptions\n      )\n    } else if (channelCount === 2) {\n      yield* AudioProcessor._encodeStereoStream(\n        audioFrames,\n        onProgress,\n        encoderOptions\n      )\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Internal method for encoding mono audio streams\n   * @param {AsyncIterable<Float32Array>} audioFrames - Mono audio frame stream\n   * @param {Function} onProgress - Progress callback function\n   * @param {EncoderOptions} encoderOptions - Encoder configuration\n   * @returns {AsyncGenerator<Object>} Stream of encoded frame data\n   * @private\n   */\n  static async *_encodeMonoStream(audioFrames, onProgress, encoderOptions) {\n    const options = encoderOptions || new EncoderOptions()\n    const encoder = encode(options)\n    let frameIndex = 0\n    for await (const frame of audioFrames) {\n      const result = encoder(frame)\n      yield result\n\n      if (onProgress) {\n        onProgress(frameIndex++)\n      }\n    }\n  }\n\n  /**\n   * Internal method for encoding stereo audio streams\n   * @param {AsyncIterable<[Float32Array, Float32Array]>} audioFrames - Stereo audio frame stream\n   * @param {Function} onProgress - Progress callback function\n   * @param {EncoderOptions} encoderOptions - Encoder configuration\n   * @returns {AsyncGenerator<Object>} Stream of encoded frame data (left then right)\n   * @private\n   */\n  static async *_encodeStereoStream(audioFrames, onProgress, encoderOptions) {\n    const options = encoderOptions || new EncoderOptions()\n    const leftEncoder = encode(options)\n    const rightEncoder = encode(options)\n    let frameIndex = 0\n\n    for await (const [leftFrame, rightFrame] of audioFrames) {\n      const leftResult = leftEncoder(leftFrame)\n      const rightResult = rightEncoder(rightFrame)\n\n      yield leftResult\n      yield rightResult\n\n      if (onProgress) {\n        onProgress(frameIndex++)\n      }\n    }\n  }\n\n  /**\n   * Decodes a stream of ATRAC1 encoded frames back to PCM audio\n   * @param {AsyncIterable<Object>} encodedFrames - Stream of encoded frame data\n   * @param {Object} [options={}] - Decoding options\n   * @param {number} [options.channelCount=1] - Number of audio channels (1 or 2)\n   * @param {Function} [options.onProgress] - Progress callback function\n   * @returns {AsyncGenerator<Float32Array>|AsyncGenerator<[Float32Array, Float32Array]>} Stream of decoded PCM frames\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static async *decodeStream(encodedFrames, options = {}) {\n    const { channelCount = 1, onProgress } = options\n\n    if (channelCount === 1) {\n      yield* AudioProcessor._decodeMonoStream(encodedFrames, onProgress)\n    } else if (channelCount === 2) {\n      yield* AudioProcessor._decodeStereoStream(encodedFrames, onProgress)\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Internal method for decoding mono audio streams\n   * @param {AsyncIterable<Object>} encodedFrames - Mono encoded frame stream\n   * @param {Function} onProgress - Progress callback function\n   * @returns {AsyncGenerator<Float32Array>} Stream of decoded mono PCM frames\n   * @private\n   */\n  static async *_decodeMonoStream(encodedFrames, onProgress) {\n    const decoder = decode()\n    let frameIndex = 0\n\n    // Create raw PCM frame stream\n    const rawPcmFrames = async function* () {\n      for await (const frame of encodedFrames) {\n        const result = decoder(frame)\n        yield result\n\n        if (onProgress) {\n          onProgress(frameIndex++)\n        }\n      }\n    }\n\n    // Always apply delay compensation\n    yield* rawPcmFrames()\n  }\n\n  /**\n   * Internal method for decoding stereo audio streams\n   * @param {AsyncIterable<Object>} encodedFrames - Stereo encoded frame stream (interleaved left/right)\n   * @param {Function} onProgress - Progress callback function\n   * @returns {AsyncGenerator<[Float32Array, Float32Array]>} Stream of decoded stereo PCM frame pairs\n   * @private\n   */\n  static async *_decodeStereoStream(encodedFrames, onProgress) {\n    const leftDecoder = decode()\n    const rightDecoder = decode()\n    let frameIndex = 0\n\n    const rawPcmFrames = async function* () {\n      let leftFrame = null\n      let isLeftChannel = true\n\n      for await (const frame of encodedFrames) {\n        if (isLeftChannel) {\n          leftFrame = frame\n        } else {\n          const rightFrame = frame\n\n          // Decode and yield the pair\n          const leftResult = leftDecoder(leftFrame)\n          const rightResult = rightDecoder(rightFrame)\n\n          yield [leftResult, rightResult]\n\n          if (onProgress) {\n            onProgress(frameIndex++)\n          }\n        }\n\n        isLeftChannel = !isLeftChannel\n      }\n\n      if (!isLeftChannel && leftFrame) {\n        const rightFrame = AudioProcessor._createDummyFrame()\n        const leftResult = leftDecoder(leftFrame)\n        const rightResult = rightDecoder(rightFrame)\n\n        yield [leftResult, rightResult]\n\n        if (onProgress) {\n          onProgress(frameIndex++)\n        }\n      }\n    }\n\n    // Always apply stereo delay compensation\n    yield* rawPcmFrames()\n  }\n\n  /**\n   * Converts channel buffer arrays to frame-based stream format\n   * @param {Float32Array[]} buffers - Array of channel buffers (1 for mono, 2 for stereo)\n   * @param {number} [frameSize=SAMPLES_PER_FRAME] - Frame size in samples\n   * @returns {Generator<Float32Array>|Generator<[Float32Array, Float32Array]>} Stream of audio frames\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static *frameBufferToFrames(buffers, frameSize = SAMPLES_PER_FRAME) {\n    const channelCount = buffers.length\n\n    if (channelCount === 1) {\n      const [left] = buffers\n      for (let i = 0; i < left.length; i += frameSize) {\n        const frame = new Float32Array(frameSize)\n        const end = Math.min(i + frameSize, left.length)\n        for (let j = 0; j < end - i; j++) {\n          frame[j] = left[i + j]\n        }\n        yield frame\n      }\n    } else if (channelCount === 2) {\n      const [left, right] = buffers\n      const maxLength = Math.max(left.length, right.length)\n\n      for (let i = 0; i < maxLength; i += frameSize) {\n        const leftFrame = new Float32Array(frameSize)\n        const rightFrame = new Float32Array(frameSize)\n\n        const end = Math.min(i + frameSize, maxLength)\n\n        for (let j = 0; j < end - i; j++) {\n          leftFrame[j] = i + j < left.length ? left[i + j] : 0\n          rightFrame[j] = i + j < right.length ? right[i + j] : 0\n        }\n\n        yield [leftFrame, rightFrame]\n      }\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Collects all frames from a stream into an array\n   * @param {AsyncIterable} frameStream - Stream of frames to collect\n   * @returns {Promise<Array>} Array of all frames from the stream\n   */\n  static async collectFrames(frameStream) {\n    const frames = []\n    for await (const frame of frameStream) {\n      frames.push(frame)\n    }\n    return frames\n  }\n\n  /**\n   * Creates a dummy frame for padding purposes\n   * @returns {Object} Empty frame structure\n   * @private\n   */\n  static _createDummyFrame() {\n    return {\n      nBfu: 0,\n      blockModes: [0, 0, 0],\n      scaleFactorIndices: new Int32Array(0),\n      wordLengthIndices: new Int32Array(0),\n      quantizedCoefficients: [],\n    }\n  }\n\n  /**\n   * Creates an AEA file blob from encoded frames\n   * @param {AsyncIterable<Object>} encodedFrames - Stream of encoded frame data\n   * @param {Object} [options={}] - AEA file options\n   * @param {string} [options.title='encoded by atrac1.js'] - File title\n   * @param {number} [options.channelCount=1] - Number of audio channels\n   * @returns {Promise<Blob>} AEA file blob\n   */\n  static async createAeaBlob(encodedFrames, options = {}) {\n    const { title = 'encoded by atrac1.js', channelCount = 1 } = options\n\n    const frames = []\n    for await (const frame of encodedFrames) {\n      frames.push(serializeFrame(frame))\n    }\n\n    const header = AeaFile.createHeader(title, frames.length, channelCount)\n    const totalSize =\n      header.length + frames.reduce((sum, frame) => sum + frame.length, 0)\n\n    const buffer = new Uint8Array(totalSize)\n    buffer.set(header, 0)\n\n    let offset = header.length\n    for (const frame of frames) {\n      buffer.set(frame, offset)\n      offset += frame.length\n    }\n\n    return new Blob([buffer], { type: 'application/octet-stream' })\n  }\n\n  /**\n   * Creates a WAV file blob from PCM audio frames\n   * @param {Float32Array[]|Float32Array|[Float32Array, Float32Array][]} pcmFrames - PCM audio frames\n   * @param {number} [channelCount=1] - Number of audio channels\n   * @param {number} [sampleRate=SAMPLE_RATE] - Audio sample rate\n   * @returns {Blob} WAV file blob\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static createWavBlob(pcmFrames, channelCount = 1, sampleRate = SAMPLE_RATE) {\n    const frames = Array.isArray(pcmFrames) ? pcmFrames : [pcmFrames]\n\n    if (channelCount === 1) {\n      return AudioProcessor._createMonoWavBlob(frames, sampleRate)\n    } else if (channelCount === 2) {\n      return AudioProcessor._createStereoWavBlob(frames, sampleRate)\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n\n  /**\n   * Creates a mono WAV file blob from PCM frames\n   * @param {Float32Array[]} frames - Array of mono PCM frames\n   * @param {number} sampleRate - Audio sample rate\n   * @returns {Blob} Mono WAV file blob\n   * @private\n   */\n  static _createMonoWavBlob(frames, sampleRate) {\n    const totalSamples = frames.reduce((sum, frame) => sum + frame.length, 0)\n    const buffer = new ArrayBuffer(\n      WAV_HEADER_SIZE + totalSamples * WAV_BYTES_PER_SAMPLE\n    )\n    const view = new DataView(buffer)\n\n    // WAV header\n    AudioProcessor._writeWavHeader(view, totalSamples, 1, sampleRate)\n\n    // PCM data using DataView for consistency\n    let offset = WAV_HEADER_SIZE\n    for (const frame of frames) {\n      for (let i = 0; i < frame.length; i++) {\n        const sample = Math.max(-1, Math.min(1, frame[i]))\n        view.setInt16(\n          offset,\n          sample < 0\n            ? sample * WAV_PCM_MAX_NEGATIVE\n            : sample * WAV_PCM_MAX_POSITIVE,\n          true\n        )\n        offset += WAV_BYTES_PER_SAMPLE\n      }\n    }\n\n    return new Blob([buffer], { type: 'audio/wav' })\n  }\n\n  /**\n   * Creates a stereo WAV file blob from PCM frame pairs\n   * @param {[Float32Array, Float32Array][]} framePairs - Array of stereo PCM frame pairs\n   * @param {number} sampleRate - Audio sample rate\n   * @returns {Blob} Stereo WAV file blob\n   * @private\n   */\n  static _createStereoWavBlob(framePairs, sampleRate) {\n    const totalSamples = framePairs.reduce(\n      (sum, [left, right]) => sum + Math.max(left.length, right.length),\n      0\n    )\n\n    const buffer = new ArrayBuffer(\n      WAV_HEADER_SIZE + totalSamples * WAV_BYTES_PER_SAMPLE * 2\n    )\n    const view = new DataView(buffer)\n\n    // WAV header\n    AudioProcessor._writeWavHeader(view, totalSamples, 2, sampleRate)\n\n    // Interleaved PCM data using DataView for consistency\n    let offset = WAV_HEADER_SIZE\n    for (const [leftFrame, rightFrame] of framePairs) {\n      const frameLength = Math.max(leftFrame.length, rightFrame.length)\n      for (let i = 0; i < frameLength; i++) {\n        const leftSample = i < leftFrame.length ? leftFrame[i] : 0\n        const rightSample = i < rightFrame.length ? rightFrame[i] : 0\n        const leftClipped = Math.max(-1, Math.min(1, leftSample))\n        const rightClipped = Math.max(-1, Math.min(1, rightSample))\n        view.setInt16(\n          offset,\n          leftClipped < 0\n            ? leftClipped * WAV_PCM_MAX_NEGATIVE\n            : leftClipped * WAV_PCM_MAX_POSITIVE,\n          true\n        )\n        offset += WAV_BYTES_PER_SAMPLE\n        view.setInt16(\n          offset,\n          rightClipped < 0\n            ? rightClipped * WAV_PCM_MAX_NEGATIVE\n            : rightClipped * WAV_PCM_MAX_POSITIVE,\n          true\n        )\n        offset += WAV_BYTES_PER_SAMPLE\n      }\n    }\n\n    return new Blob([buffer], { type: 'audio/wav' })\n  }\n\n  /**\n   * Writes WAV file header to DataView\n   * @param {DataView} view - DataView to write to\n   * @param {number} totalSamples - Total number of samples\n   * @param {number} channelCount - Number of audio channels\n   * @param {number} sampleRate - Audio sample rate\n   * @private\n   */\n  static _writeWavHeader(view, totalSamples, channelCount, sampleRate) {\n    const byteRate = sampleRate * channelCount * WAV_BYTES_PER_SAMPLE\n    const blockAlign = channelCount * WAV_BYTES_PER_SAMPLE\n    const dataSize = totalSamples * blockAlign\n    const fileSize = WAV_DATA_OFFSET + dataSize\n\n    let pos = 0\n\n    // Helper function to write ASCII strings\n    const writeString = (offset, string) => {\n      for (let i = 0; i < string.length; i++) {\n        view.setUint8(offset + i, string.charCodeAt(i))\n      }\n    }\n\n    // RIFF header\n    writeString(pos, 'RIFF')\n    pos += 4\n    view.setUint32(pos, fileSize, true)\n    pos += 4\n    writeString(pos, 'WAVE')\n    pos += 4\n\n    // fmt chunk\n    writeString(pos, 'fmt ')\n    pos += 4\n    view.setUint32(pos, 16, true) // chunk size\n    pos += 4\n    view.setUint16(pos, 1, true) // PCM format\n    pos += 2\n    view.setUint16(pos, channelCount, true)\n    pos += 2\n    view.setUint32(pos, sampleRate, true)\n    pos += 4\n    view.setUint32(pos, byteRate, true)\n    pos += 4\n    view.setUint16(pos, blockAlign, true)\n    pos += 2\n    view.setUint16(pos, WAV_BITS_PER_SAMPLE, true) // bits per sample\n    pos += 2\n\n    // data chunk\n    writeString(pos, 'data')\n    pos += 4\n    view.setUint32(pos, dataSize, true)\n  }\n\n  /**\n   * Parses an AEA file blob to extract header info and frame data\n   * @param {Blob} blob - AEA file blob to parse\n   * @returns {Promise<Object>} Parsed AEA file data\n   * @returns {Object} returns.info - AEA file header information\n   * @returns {Uint8Array[]} returns.frameData - Array of frame data buffers\n   */\n  static async parseAeaBlob(blob) {\n    const buffer = await blob.arrayBuffer()\n    const header = new Uint8Array(buffer.slice(0, AEA_HEADER_SIZE))\n    const info = AeaFile.parseHeader(header)\n\n    const frameData = []\n    for (let i = AEA_HEADER_SIZE; i < buffer.byteLength; i += SOUND_UNIT_SIZE) {\n      const frame = new Uint8Array(buffer.slice(i, i + SOUND_UNIT_SIZE))\n      if (frame.length === SOUND_UNIT_SIZE) {\n        frameData.push(frame)\n      }\n    }\n\n    return { info, frameData }\n  }\n\n  /**\n   * Creates a stream of deserialized frames from binary frame data\n   * @param {Uint8Array[]} frameData - Array of binary frame data\n   * @returns {Generator<Object>} Stream of deserialized frame objects\n   */\n  static *deserializedFrameStream(frameData) {\n    for (const frame of frameData) {\n      yield deserializeFrame(frame)\n    }\n  }\n\n  /**\n   * Assembles PCM frames into a continuous buffer\n   * @param {Float32Array[]|[Float32Array, Float32Array][]} pcmFrames - Array of PCM frames\n   * @param {number} channelCount - Number of audio channels\n   * @returns {Float32Array} Continuous PCM buffer (interleaved for stereo)\n   * @throws {Error} If unsupported channel count is provided\n   */\n  static assemblePcmFrames(pcmFrames, channelCount) {\n    if (channelCount === 1) {\n      const totalSamples = pcmFrames.reduce(\n        (sum, frame) => sum + frame.length,\n        0\n      )\n      const pcm = new Float32Array(totalSamples)\n      let offset = 0\n      for (const frame of pcmFrames) {\n        pcm.set(frame, offset)\n        offset += frame.length\n      }\n      return pcm\n    } else if (channelCount === 2) {\n      const totalSamples = pcmFrames.reduce(\n        (sum, [left, right]) => sum + Math.max(left.length, right.length),\n        0\n      )\n      const pcm = new Float32Array(totalSamples * 2)\n\n      let pcmOffset = 0\n      for (const [leftFrame, rightFrame] of pcmFrames) {\n        const frameLength = Math.max(leftFrame.length, rightFrame.length)\n        for (let j = 0; j < frameLength; j++) {\n          pcm[pcmOffset * 2] = j < leftFrame.length ? leftFrame[j] : 0\n          pcm[pcmOffset * 2 + 1] = j < rightFrame.length ? rightFrame[j] : 0\n          pcmOffset++\n        }\n      }\n      return pcm\n    } else {\n      throw new Error(`Unsupported channel count: ${channelCount}`)\n    }\n  }\n}\n","/**\n * Carta1 Web Worker - Self-contained audio codec worker\n *\n * Provides asynchronous encoding and decoding of ATRAC1 audio format\n * in a separate thread to prevent blocking the main UI thread.\n *\n * Usage:\n *   Include this file as a web worker:\n *   new Worker('carta1-worker.min.js')\n */\n\nimport { AeaFile } from '../io/serialization.js'\nimport { EncoderOptions } from '../core/options.js'\nimport { AudioProcessor } from '../io/processor.js'\nimport { SAMPLE_RATE } from '../core/constants.js'\n\n/**\n * Handle messages from the main thread\n *\n * Supported message types:\n * - 'encode': Encode PCM to ATRAC1\n * - 'decode': Decode ATRAC1 to PCM\n * - 'parseAea': Parse AEA blob metadata\n * - 'getEncoderOptions': Get default encoder options\n */\nself.onmessage = async (e) => {\n  const { jobId, type, pcmData, aea, options, title, blob } = e.data\n\n  try {\n    let result\n\n    switch (type) {\n      case 'encode':\n        result = await encodeAtrac(pcmData, options, title)\n        break\n      case 'decode':\n        result = await decodeAtrac(aea)\n        break\n      case 'parseAea':\n        result = await AudioProcessor.parseAeaBlob(blob)\n        break\n      case 'getEncoderOptions':\n        result = getEncoderOptionsMetadata()\n        break\n      default:\n        throw new Error(`Unknown worker message type: ${type}`)\n    }\n\n    self.postMessage({ jobId, result })\n  } catch (error) {\n    console.error('Worker error:', error)\n    self.postMessage({ jobId, error: error.message })\n  }\n}\n\n/**\n * Encode PCM audio data to ATRAC1 format\n *\n * @param {Array<Float32Array>} pcmData - PCM audio data [mono] or [left, right]\n * @param {Object} options - Encoder options\n * @returns {Promise<Object>} Encoded AEA blob\n */\nasync function encodeAtrac(pcmData, options, title) {\n  title = title ?? 'encoded by carta1'\n  try {\n    const audioFrames = AudioProcessor.frameBufferToFrames(pcmData)\n    const channelCount = pcmData.length\n\n    // Create encoder options from provided settings\n    const encoderOptions = new EncoderOptions(options)\n\n    const encodedFrames = AudioProcessor.encodeStream(audioFrames, {\n      channelCount,\n      encoderOptions,\n    })\n\n    const frameArray = await AudioProcessor.collectFrames(encodedFrames)\n\n    const aeaBlob = await AudioProcessor.createAeaBlob(frameArray, {\n      title,\n      channelCount,\n    })\n\n    // Return the blob\n    return { aeaBlob }\n  } catch (error) {\n    console.error('Worker encode error:', error)\n    throw error\n  }\n}\n\n/**\n * Get default encoder options metadata\n *\n * @returns {Object} Default encoder options as plain object\n */\nfunction getEncoderOptionsMetadata() {\n  const defaultOptions = new EncoderOptions()\n  return defaultOptions.toObject()\n}\n\n/**\n * Decode ATRAC1 data to PCM WAV 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