cantor-digitalis 0.0.1

package/dist/index.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"index.js","sources":["../src/nodes/gain.ts","../src/nodes/pulse-train.ts","../src/nodes/glottal-formant.ts","../src/nodes/spectral-tilt.ts","../src/nodes/noise-source.ts","../src/nodes/glottal-flow-derivative.ts","../src/nodes/formant-resonator.ts","../src/nodes/formant-bank.ts","../src/nodes/anti-resonance.ts","../src/nodes/vocal-tract.ts","../src/nodes/voice.ts","../src/parameters/vowels.ts","../src/parameters/index.ts"],"sourcesContent":["import type { Node } from \"./types\";\n\nexport type GainParams = {\n  gain: number;\n}\n\nexport class Gain implements Node<GainParams> {\n  private ctx: AudioContext;\n  private gainNode: GainNode;\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(ctx: AudioContext, gainNode: GainNode) {\n    this.ctx = ctx;\n    this.gainNode = gainNode;\n    this.in = gainNode;\n    this.out = gainNode;\n  }\n\n  static async create(ctx: AudioContext, params: GainParams): Promise<Gain> {\n    const gainNode = ctx.createGain();\n    gainNode.gain.setValueAtTime(params.gain, ctx.currentTime);\n    return new Gain(ctx, gainNode);\n  }\n\n  update(params: GainParams) {\n    this.gainNode.gain.setTargetAtTime(params.gain, this.ctx.currentTime, 0.02);\n  }\n\n  destroy() {\n    this.gainNode.disconnect();\n  }\n\n  get gain(): AudioParam {\n    return this.gainNode.gain;\n  }\n}\n","import type { Node } from \"./types\";\nimport { Gain } from \"./gain\";\n\nexport type PulseTrainParams = {\n  /** Fundamental frequency in Hz (derived from P, P₀) */\n  f0: number;\n  /** Jitter depth: maximum f₀ perturbation as a fraction (0-0.3 for ±30%) */\n  jitterDepth: number;\n  /** Shimmer depth: maximum amplitude perturbation as a fraction (0-1 for ±100%) */\n  shimmerDepth: number;\n};\n\n/**\n * Pulse Train AudioWorklet Processor code.\n *\n * Generates a band-limited impulse train at the fundamental frequency f₀,\n * with per-period jitter (frequency perturbation) and shimmer (amplitude\n * perturbation) for roughness simulation.\n *\n * The processor tracks phase internally and generates new random perturbation\n * values at each period boundary, ensuring jitter and shimmer are synchronized\n * to the glottal cycle.\n *\n * Band-limiting is achieved by summing harmonics up to the Nyquist frequency,\n * which avoids aliasing artifacts.\n *\n * Paper reference: Section 3.1 (Dirac comb), Section 4.1.2 (jitter),\n *                  Section 4.2.4 (shimmer)\n */\nconst processorCode = `\nclass PulseTrainProcessor extends AudioWorkletProcessor {\n  static get parameterDescriptors() {\n    return [\n      { name: \"f0\", defaultValue: 220, minValue: 20, maxValue: 20000, automationRate: \"a-rate\" },\n      { name: \"jitterDepth\", defaultValue: 0, minValue: 0, maxValue: 0.3, automationRate: \"k-rate\" },\n      { name: \"shimmerDepth\", defaultValue: 0, minValue: 0, maxValue: 1, automationRate: \"k-rate\" }\n    ];\n  }\n\n  constructor() {\n    super();\n    // Phase accumulator (0 to 1)\n    this.phase = 0;\n\n    // Current perturbation values (updated each period)\n    this.currentJitter = 1;   // Multiplier for f0 (1 = no jitter)\n    this.currentShimmer = 1;  // Multiplier for amplitude (1 = no shimmer)\n\n    // Box-Muller spare for Gaussian random\n    this.hasSpare = false;\n    this.spare = 0;\n\n    // Cached harmonic count for band-limiting\n    this.harmonicCount = 0;\n    this.lastF0ForHarmonics = 220;\n    this.updateHarmonicCount(220);\n  }\n\n  /**\n   * Update the number of harmonics to sum for band-limited impulse.\n   * We include harmonics up to Nyquist to avoid aliasing.\n   */\n  updateHarmonicCount(f0) {\n    const nyquist = sampleRate / 2;\n    this.harmonicCount = Math.max(1, Math.floor(nyquist / f0));\n    this.lastF0ForHarmonics = f0;\n  }\n\n  /**\n   * Generate a Gaussian-distributed random number using Box-Muller transform.\n   * Returns values with mean 0 and standard deviation 1.\n   */\n  gaussianRandom() {\n    if (this.hasSpare) {\n      this.hasSpare = false;\n      return this.spare;\n    }\n\n    let u, v, s;\n    do {\n      u = Math.random() * 2 - 1;\n      v = Math.random() * 2 - 1;\n      s = u * u + v * v;\n    } while (s >= 1 || s === 0);\n\n    const mul = Math.sqrt(-2 * Math.log(s) / s);\n    this.spare = v * mul;\n    this.hasSpare = true;\n    return u * mul;\n  }\n\n  /**\n   * Generate new perturbation values at a period boundary.\n   * Jitter affects the next period's frequency.\n   * Shimmer affects the current pulse's amplitude.\n   */\n  generatePerturbations(jitterDepth, shimmerDepth) {\n    // Jitter: Gaussian perturbation of f0, clipped to ±jitterDepth\n    // Using Gaussian makes small perturbations more likely than large ones\n    const jitterRaw = this.gaussianRandom() * jitterDepth * 0.5;\n    const jitterClamped = Math.max(-jitterDepth, Math.min(jitterDepth, jitterRaw));\n    this.currentJitter = 1 + jitterClamped;\n\n    // Shimmer: Gaussian perturbation of amplitude, clipped to [0, 2]\n    // At shimmerDepth=1, amplitude can vary from 0 to 2 (±100%)\n    const shimmerRaw = this.gaussianRandom() * shimmerDepth * 0.5;\n    const shimmerClamped = Math.max(-shimmerDepth, Math.min(shimmerDepth, shimmerRaw));\n    this.currentShimmer = 1 + shimmerClamped;\n\n    // Clamp shimmer to non-negative (amplitude can't be negative)\n    this.currentShimmer = Math.max(0, this.currentShimmer);\n  }\n\n  /**\n   * Compute a band-limited impulse using additive synthesis.\n   * Sums cosines at all harmonic frequencies up to Nyquist.\n   *\n   * The impulse is normalized so that its peak amplitude is approximately 1.\n   */\n  computeBandLimitedImpulse(phase) {\n    // Sum all harmonics: Σ cos(2π * k * phase) for k = 1 to harmonicCount\n    // This produces a Dirac comb approximation\n    let sum = 0;\n    const twoPiPhase = 2 * Math.PI * phase;\n    for (let k = 1; k <= this.harmonicCount; k++) {\n      sum += Math.cos(k * twoPiPhase);\n    }\n    // Normalize by harmonic count to keep amplitude reasonable\n    return sum / this.harmonicCount;\n  }\n\n  process(inputs, outputs, parameters) {\n    const output = outputs[0]?.[0];\n    if (!output) {\n      return true;\n    }\n\n    const dt = 1 / sampleRate;\n    // k-rate parameters (single value per block)\n    const jitterDepth = parameters.jitterDepth[0];\n    const shimmerDepth = parameters.shimmerDepth[0];\n\n    for (let i = 0; i < output.length; i++) {\n      // a-rate: per-sample value for f0\n      const f0 = parameters.f0.length > 1 ? parameters.f0[i] : parameters.f0[0];\n\n      // Compute effective f0 with current jitter\n      const f0Effective = f0 * this.currentJitter;\n\n      // Advance phase\n      this.phase += f0Effective * dt;\n\n      // Check for period boundary (phase wrapped)\n      if (this.phase >= 1) {\n        this.phase -= 1;\n        // Generate new perturbations for the next period\n        this.generatePerturbations(jitterDepth, shimmerDepth);\n        // Update harmonic count if f0 changed significantly (>10%)\n        if (Math.abs(f0 - this.lastF0ForHarmonics) / this.lastF0ForHarmonics > 0.1) {\n          this.updateHarmonicCount(f0);\n        }\n      }\n\n      // Generate band-limited impulse and apply shimmer\n      output[i] = this.currentShimmer * this.computeBandLimitedImpulse(this.phase);\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('pulse-train-processor', PulseTrainProcessor);\n`;\n\n// Track whether the worklet module has been registered\nlet moduleRegistered = false;\n\n/**\n * Registers the pulse train worklet module with the AudioContext.\n * Only registers once per application lifetime.\n */\nasync function ensureModuleRegistered(ctx: AudioContext): Promise<void> {\n  if (moduleRegistered) {\n    return;\n  }\n\n  const blob = new Blob([processorCode], { type: \"application/javascript\" });\n  const url = URL.createObjectURL(blob);\n\n  try {\n    await ctx.audioWorklet.addModule(url);\n    moduleRegistered = true;\n  } finally {\n    URL.revokeObjectURL(url);\n  }\n}\n\n/**\n * Pulse Train (Voiced Excitation Source)\n *\n * Generates a periodic impulse train at the fundamental frequency f₀,\n * with integrated jitter (f₀ perturbation) and shimmer (amplitude perturbation)\n * for roughness simulation.\n *\n * In the spectral domain, the unperturbed output is a Dirac comb: Σδ(f − n·f₀),\n * representing energy at each harmonic frequency. Jitter spreads energy around\n * harmonics, and shimmer modulates the overall amplitude cycle-to-cycle.\n *\n * The impulse train is the input to the glottal formant filter, which shapes\n * each impulse into the characteristic glottal pulse waveform.\n *\n * Paper reference: Section 3.1 (spectral equation showing the Dirac comb term),\n *                  Section 4.1.2 (jitter), Section 4.2.4 (shimmer)\n *\n * Input parameters:\n *   - f0: Fundamental frequency in Hz\n *   - jitterDepth: Maximum f₀ perturbation (0-0.3 for up to ±30%)\n *   - shimmerDepth: Maximum amplitude perturbation (0-1 for up to ±100%)\n */\nexport class PulseTrain implements Node<PulseTrainParams> {\n  private ctx: AudioContext;\n  private workletNode: AudioWorkletNode;\n  private gain: Gain;\n\n  public in: null;\n  public out: AudioNode;\n\n  private constructor(ctx: AudioContext, workletNode: AudioWorkletNode, gain: Gain) {\n    this.ctx = ctx;\n    this.workletNode = workletNode;\n    this.gain = gain;\n    this.out = gain.out;\n    this.in = null;\n  }\n\n  /** Fundamental frequency AudioParam (a-rate, 20-20000 Hz) */\n  get f0(): AudioParam {\n    return this.workletNode.parameters.get(\"f0\")!;\n  }\n\n  /** Jitter depth AudioParam (k-rate, 0-0.3) */\n  get jitterDepth(): AudioParam {\n    return this.workletNode.parameters.get(\"jitterDepth\")!;\n  }\n\n  /** Shimmer depth AudioParam (k-rate, 0-1) */\n  get shimmerDepth(): AudioParam {\n    return this.workletNode.parameters.get(\"shimmerDepth\")!;\n  }\n\n  static async create(ctx: AudioContext, params: PulseTrainParams): Promise<PulseTrain> {\n    await ensureModuleRegistered(ctx);\n\n    const workletNode = new AudioWorkletNode(ctx, \"pulse-train-processor\");\n    const gain = await Gain.create(ctx, { gain: 0 });\n\n    workletNode.connect(gain.in);\n\n    const pulseTrain = new PulseTrain(ctx, workletNode, gain);\n    pulseTrain.update(params);\n\n    return pulseTrain;\n  }\n\n  update(params: PulseTrainParams) {\n    this.f0.setTargetAtTime(params.f0, this.ctx.currentTime, 0.02);\n    this.jitterDepth.setTargetAtTime(params.jitterDepth, this.ctx.currentTime, 0.02);\n    this.shimmerDepth.setTargetAtTime(params.shimmerDepth, this.ctx.currentTime, 0.02);\n  }\n\n  destroy() {\n    this.workletNode.disconnect();\n    this.gain.destroy();\n  }\n\n  start() {\n    this.gain.update({ gain: 1 });\n  }\n\n  stop() {\n    this.gain.update({ gain: 0 });\n  }\n}\n","import type { Node } from \"./types\";\n\nexport type GlottalFormantParams = {\n  /** Glottal formant centre frequency in Hz, computed as f0 / (2 * Oq) */\n  Fg: number;\n  /** Glottal formant bandwidth in Hz, computed from f0, Oq, and αm */\n  Bg: number;\n  /** Source amplitude, derived from E, Oq, and R (shimmer) */\n  Ag: number;\n};\n\n/**\n * Glottal Formant AudioWorklet Processor code.\n *\n * Implements the transfer function from Section 3.2.1:\n * GF(z) = -Ag * z^{-1} * (1 - z^{-1}) / (1 - 2R*cos(θ)*z^{-1} + R²*z^{-2})\n *\n * Where:\n *   R = e^(-π * Bg * Ts)  (pole radius, Ts = 1/sampleRate)\n *   θ = 2π * Fg * Ts      (pole angle)\n */\nconst processorCode = `\nclass GlottalFormantProcessor extends AudioWorkletProcessor {\n  static get parameterDescriptors() {\n    return [\n      { name: \"Fg\", defaultValue: 110, minValue: 20, maxValue: 2000, automationRate: \"a-rate\" },\n      { name: \"Bg\", defaultValue: 50, minValue: 10, maxValue: 500, automationRate: \"a-rate\" },\n      { name: \"Ag\", defaultValue: 1, minValue: 0, maxValue: 10, automationRate: \"a-rate\" }\n    ];\n  }\n\n  constructor() {\n    super();\n    // Filter state (delay line)\n    this.x1 = 0;\n    this.x2 = 0;\n    this.y1 = 0;\n    this.y2 = 0;\n\n    // Cached coefficient values to avoid recomputing when unchanged\n    this.lastFg = -1;\n    this.lastBg = -1;\n    this.R = 0;\n    this.cosTheta = 0;\n    this.a1 = 0;\n    this.a2 = 0;\n  }\n\n  process(inputs, outputs, parameters) {\n    const input = inputs[0]?.[0];\n    const output = outputs[0]?.[0];\n\n    if (!input || !output) {\n      return true;\n    }\n\n    const Ts = 1 / sampleRate;\n    // k-rate parameters (single value per block)\n    const Fg = parameters.Fg[0];\n    const Bg = parameters.Bg[0];\n\n    // Recompute filter coefficients if Fg or Bg changed\n    if (Fg !== this.lastFg || Bg !== this.lastBg) {\n      this.R = Math.exp(-Math.PI * Bg * Ts);\n      const theta = 2 * Math.PI * Fg * Ts;\n      this.cosTheta = Math.cos(theta);\n      this.a1 = -2 * this.R * this.cosTheta;\n      this.a2 = this.R * this.R;\n      this.lastFg = Fg;\n      this.lastBg = Bg;\n    }\n\n    for (let i = 0; i < input.length; i++) {\n      // a-rate: per-sample value for Ag\n      const Ag = parameters.Ag.length > 1 ? parameters.Ag[i] : parameters.Ag[0];\n\n      const x0 = input[i];\n\n      // Numerator coefficients (feedforward) depend on Ag\n      // From: -Ag * z^{-1} * (1 - z^{-1}) = -Ag*z^{-1} + Ag*z^{-2}\n      const b1 = -Ag;\n      const b2 = Ag;\n\n      // Direct Form I biquad (b0 is always 0)\n      const y0 = b1 * this.x1 + b2 * this.x2\n                 - this.a1 * this.y1 - this.a2 * this.y2;\n\n      // Update delay line\n      this.x2 = this.x1;\n      this.x1 = x0;\n      this.y2 = this.y1;\n      this.y1 = y0;\n\n      output[i] = y0;\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('glottal-formant-processor', GlottalFormantProcessor);\n`;\n\n// Track whether the worklet module has been registered\nlet moduleRegistered = false;\n\n/**\n * Registers the glottal formant worklet module with the AudioContext.\n * Only registers once per application lifetime.\n */\nasync function ensureModuleRegistered(ctx: AudioContext): Promise<void> {\n  if (moduleRegistered) {\n    return;\n  }\n\n  const blob = new Blob([processorCode], { type: \"application/javascript\" });\n  const url = URL.createObjectURL(blob);\n\n  try {\n    await ctx.audioWorklet.addModule(url);\n    moduleRegistered = true;\n  } finally {\n    URL.revokeObjectURL(url);\n  }\n}\n\n/**\n * Glottal Formant (GF)\n *\n * The glottal formant represents the main spectral peak of the voice source,\n * corresponding to the resonance characteristics of the glottal pulse. It is\n * implemented as a 2-pole 1-zero digital resonant filter in series with a\n * 1-zero differentiation filter.\n *\n * The transfer function (Section 3.2.1) is:\n *\n * GF(z) = -Ag * z^{-1} * (1 - z^{-1}) / (1 - 2R*cos(θ)*z^{-1} + R²*z^{-2})\n *\n * Where:\n *   R = e^(-π * Bg * Ts)  (pole radius, Ts = sampling period)\n *   θ = 2π * Fg * Ts      (pole angle)\n *\n * The filter shapes the periodic impulse train (at frequency f0) to produce\n * the characteristic spectrum of the glottal flow derivative.\n *\n * Paper reference: Section 3.2.1 (filter structure), Section 4.2.2 (parameter mapping)\n *\n * Input parameters:\n *   - Fg (glottal formant centre frequency) — computed as f0 / (2·Oq)\n *   - Bg (glottal formant bandwidth) — computed from f0, Oq, and αm\n *   - Ag (source amplitude) — derived from E, Oq, and R (shimmer)\n *\n * The intermediate parameters Oq (open quotient) and αm (asymmetry coefficient)\n * are computed from T (tenseness), E (vocal effort), and M (laryngeal mechanism).\n */\nexport class GlottalFormant implements Node<GlottalFormantParams> {\n  private ctx: AudioContext;\n  private workletNode: AudioWorkletNode;\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(ctx: AudioContext, workletNode: AudioWorkletNode) {\n    this.ctx = ctx;\n    this.workletNode = workletNode;\n    this.in = workletNode;\n    this.out = workletNode;\n  }\n\n  /** Glottal formant centre frequency AudioParam (a-rate, 20-2000 Hz) */\n  get Fg(): AudioParam {\n    return this.workletNode.parameters.get(\"Fg\")!;\n  }\n\n  /** Glottal formant bandwidth AudioParam (a-rate, 10-500 Hz) */\n  get Bg(): AudioParam {\n    return this.workletNode.parameters.get(\"Bg\")!;\n  }\n\n  /** Source amplitude AudioParam (a-rate, 0-10) */\n  get Ag(): AudioParam {\n    return this.workletNode.parameters.get(\"Ag\")!;\n  }\n\n  /**\n   * Creates a new GlottalFormant node.\n   *\n   * The AudioWorklet module is registered automatically on first use.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: GlottalFormantParams\n  ): Promise<GlottalFormant> {\n    await ensureModuleRegistered(ctx);\n\n    const workletNode = new AudioWorkletNode(ctx, \"glottal-formant-processor\");\n    const node = new GlottalFormant(ctx, workletNode);\n    node.update(params);\n\n    return node;\n  }\n\n  /**\n   * Updates the glottal formant parameters.\n   * Sets AudioParams via setTargetAtTime for smooth transitions.\n   */\n  update(params: GlottalFormantParams): void {\n    this.Fg.setTargetAtTime(params.Fg, this.ctx.currentTime, 0.02);\n    this.Bg.setTargetAtTime(params.Bg, this.ctx.currentTime, 0.02);\n    this.Ag.setTargetAtTime(params.Ag, this.ctx.currentTime, 0.02);\n  }\n\n  destroy(): void {\n    this.workletNode.disconnect();\n  }\n}\n","import type { Node } from \"./types\";\n\nexport type SpectralTiltParams = {\n  /** First stage attenuation in dB at 3000 Hz, derived from E and M */\n  Tl1: number;\n  /** Second stage attenuation in dB at 3000 Hz, derived from E and M */\n  Tl2: number;\n};\n\n/**\n * Spectral Tilt AudioWorklet Processor code.\n *\n * Implements the transfer function from Section 3.2.2:\n * ST(z) = ST₁(z) × ST₂(z)\n *\n * Where each stage is:\n * STᵢ(z) = (1 - aᵢ) / (1 - aᵢ·z⁻¹)\n *\n * And:\n *   aᵢ = νᵢ - √(νᵢ² - 1)\n *   νᵢ = 1 + (1 - cos(2π·3000·Ts)) / (10^(Tlᵢ/10) - 1)\n *\n * This gives unity gain at DC and specified attenuation (Tlᵢ dB) at 3000 Hz.\n */\nconst processorCode = `\nclass SpectralTiltProcessor extends AudioWorkletProcessor {\n  static get parameterDescriptors() {\n    return [\n      { name: \"Tl1\", defaultValue: 0, minValue: 0, maxValue: 50, automationRate: \"k-rate\" },\n      { name: \"Tl2\", defaultValue: 0, minValue: 0, maxValue: 30, automationRate: \"k-rate\" }\n    ];\n  }\n\n  constructor() {\n    super();\n    // Filter coefficients for two stages\n    this.a1 = 0;  // First stage pole\n    this.a2 = 0;  // Second stage pole\n    this.g1 = 1;  // First stage gain (1 - a1)\n    this.g2 = 1;  // Second stage gain (1 - a2)\n\n    // Filter state (one delay element per stage)\n    this.y1_prev = 0;  // Previous output of stage 1\n    this.y2_prev = 0;  // Previous output of stage 2\n\n    // Cache last parameter values for change detection\n    this.lastTl1 = -1;\n    this.lastTl2 = -1;\n\n    // Pre-compute cosOmega since it's constant for a given sample rate\n    const Ts = 1 / sampleRate;\n    this.cosOmega = Math.cos(2 * Math.PI * 3000 * Ts);\n  }\n\n  /**\n   * Compute the pole coefficient for a single stage.\n   * Returns [a, g] where a is the pole and g = 1 - a is the gain.\n   */\n  computeStageCoefficients(Tl) {\n    // Bypass if attenuation is zero or negative\n    if (Tl <= 0) {\n      return [0, 1];\n    }\n\n    // Power ratio at 3000 Hz (10^(Tl/10) is the attenuation factor)\n    const attenuationRatio = Math.pow(10, Tl / 10);\n\n    // Compute ν from Section 3.2.2\n    // νᵢ = 1 + (1 - cos(ω)) / (10^(Tlᵢ/10) - 1)\n    const nu = 1 + (1 - this.cosOmega) / (attenuationRatio - 1);\n\n    // Compute pole coefficient: a = ν - √(ν² - 1)\n    // This ensures 0 < a < 1 for stability\n    const a = nu - Math.sqrt(nu * nu - 1);\n    const g = 1 - a;\n\n    return [a, g];\n  }\n\n  process(inputs, outputs, parameters) {\n    const input = inputs[0]?.[0];\n    const output = outputs[0]?.[0];\n\n    if (!input || !output) {\n      return true;\n    }\n\n    // k-rate parameters (single value per block)\n    const Tl1 = parameters.Tl1[0];\n    const Tl2 = parameters.Tl2[0];\n\n    // Recompute coefficients if parameters changed\n    if (Tl1 !== this.lastTl1) {\n      [this.a1, this.g1] = this.computeStageCoefficients(Tl1);\n      this.lastTl1 = Tl1;\n    }\n    if (Tl2 !== this.lastTl2) {\n      [this.a2, this.g2] = this.computeStageCoefficients(Tl2);\n      this.lastTl2 = Tl2;\n    }\n\n    for (let i = 0; i < input.length; i++) {\n      // Stage 1: y1[n] = g1 * x[n] + a1 * y1[n-1]\n      const y1 = this.g1 * input[i] + this.a1 * this.y1_prev;\n      this.y1_prev = y1;\n\n      // Stage 2: y2[n] = g2 * y1[n] + a2 * y2[n-1]\n      const y2 = this.g2 * y1 + this.a2 * this.y2_prev;\n      this.y2_prev = y2;\n\n      output[i] = y2;\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('spectral-tilt-processor', SpectralTiltProcessor);\n`;\n\n// Track whether the worklet module has been registered\nlet moduleRegistered = false;\n\n/**\n * Registers the spectral tilt worklet module with the AudioContext.\n * Only registers once per application lifetime.\n */\nasync function ensureModuleRegistered(ctx: AudioContext): Promise<void> {\n  if (moduleRegistered) {\n    return;\n  }\n\n  const blob = new Blob([processorCode], { type: \"application/javascript\" });\n  const url = URL.createObjectURL(blob);\n\n  try {\n    await ctx.audioWorklet.addModule(url);\n    moduleRegistered = true;\n  } finally {\n    URL.revokeObjectURL(url);\n  }\n}\n\n/**\n * Spectral Tilt (ST)\n *\n * The spectral tilt filter models the high-frequency roll-off of the glottal\n * source spectrum. Louder, more pressed phonation has less spectral tilt\n * (brighter sound), while softer or breathier phonation has more tilt\n * (darker sound).\n *\n * It is implemented as a cascade of two 1-pole 1-zero low-pass filters, each\n * providing attenuation specified in dB at 3000 Hz.\n *\n * The transfer function (Section 3.2.2) is:\n *\n * ST(z) = ST₁(z) × ST₂(z)\n *\n * Where each stage is:\n * STᵢ(z) = (1 - aᵢ) / (1 - aᵢ·z⁻¹)\n *\n * With:\n *   aᵢ = νᵢ - √(νᵢ² - 1)\n *   νᵢ = 1 + (1 - cos(2π·3000·Ts)) / (10^(Tlᵢ/10) - 1)\n *\n * Paper reference: Section 3.2.2 (filter structure), Section 4.2.3 (parameter mapping)\n *\n * Input parameters:\n *   - Tl₁ (first tilt stage attenuation in dB at 3 kHz) — derived from E and M\n *   - Tl₂ (second tilt stage attenuation in dB at 3 kHz) — derived from E and M\n *\n * Parameter mapping from Section 4.2.3:\n *   For M=1 (chest voice):\n *     Tl₁ = 27 - 21·Ep dB\n *     Tl₂ = 11 - 11·Ep dB\n *   For M=2 (falsetto):\n *     Tl₁ = 45 - 36·Ep dB\n *     Tl₂ = 20 - 18.5·Ep dB\n *\n * Where Ep is the perturbed vocal effort (E with heartbeat and slow perturbations).\n */\nexport class SpectralTilt implements Node<SpectralTiltParams> {\n  private ctx: AudioContext;\n  private workletNode: AudioWorkletNode;\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(ctx: AudioContext, workletNode: AudioWorkletNode) {\n    this.ctx = ctx;\n    this.workletNode = workletNode;\n    this.in = workletNode;\n    this.out = workletNode;\n  }\n\n  /** First stage spectral tilt attenuation AudioParam (k-rate, 0-50 dB at 3kHz) */\n  get Tl1(): AudioParam {\n    return this.workletNode.parameters.get(\"Tl1\")!;\n  }\n\n  /** Second stage spectral tilt attenuation AudioParam (k-rate, 0-30 dB at 3kHz) */\n  get Tl2(): AudioParam {\n    return this.workletNode.parameters.get(\"Tl2\")!;\n  }\n\n  /**\n   * Creates a new SpectralTilt node.\n   *\n   * The AudioWorklet module is registered automatically on first use.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: SpectralTiltParams\n  ): Promise<SpectralTilt> {\n    await ensureModuleRegistered(ctx);\n\n    const workletNode = new AudioWorkletNode(ctx, \"spectral-tilt-processor\");\n    const node = new SpectralTilt(ctx, workletNode);\n    node.update(params);\n\n    return node;\n  }\n\n  /**\n   * Updates the spectral tilt parameters.\n   * Sets AudioParams via setTargetAtTime for smooth transitions.\n   */\n  update(params: SpectralTiltParams): void {\n    this.Tl1.setTargetAtTime(params.Tl1, this.ctx.currentTime, 0.02);\n    this.Tl2.setTargetAtTime(params.Tl2, this.ctx.currentTime, 0.02);\n  }\n\n  destroy(): void {\n    this.workletNode.disconnect();\n  }\n}\n","import type { Node } from \"./types\";\n\nexport type NoiseSourceParams = {\n  /** Noise amplitude, derived directly from B (breathiness) */\n  An: number;\n};\n\n/**\n * Noise Source AudioWorklet Processor code.\n *\n * Generates Gaussian white noise using the Box-Muller transform,\n * scaled by an amplitude parameter.\n */\nconst processorCode = `\nclass NoiseSourceProcessor extends AudioWorkletProcessor {\n  static get parameterDescriptors() {\n    return [\n      { name: \"An\", defaultValue: 0, minValue: 0, maxValue: 1, automationRate: \"a-rate\" }\n    ];\n  }\n\n  constructor() {\n    super();\n    // Box-Muller spare value\n    this.hasSpare = false;\n    this.spare = 0;\n  }\n\n  /**\n   * Generate a Gaussian-distributed random number using Box-Muller transform.\n   * Returns values with mean 0 and standard deviation 1.\n   */\n  gaussianRandom() {\n    if (this.hasSpare) {\n      this.hasSpare = false;\n      return this.spare;\n    }\n\n    // Generate two uniform random numbers in (0, 1)\n    let u, v, s;\n    do {\n      u = Math.random() * 2 - 1;\n      v = Math.random() * 2 - 1;\n      s = u * u + v * v;\n    } while (s >= 1 || s === 0);\n\n    const mul = Math.sqrt(-2 * Math.log(s) / s);\n    this.spare = v * mul;\n    this.hasSpare = true;\n    return u * mul;\n  }\n\n  process(inputs, outputs, parameters) {\n    const output = outputs[0]?.[0];\n\n    if (!output) {\n      return true;\n    }\n\n    for (let i = 0; i < output.length; i++) {\n      // a-rate: per-sample value for An\n      const An = parameters.An.length > 1 ? parameters.An[i] : parameters.An[0];\n      output[i] = An * this.gaussianRandom();\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('noise-source-processor', NoiseSourceProcessor);\n`;\n\n// Track whether the worklet module has been registered\nlet moduleRegistered = false;\n\n/**\n * Registers the noise source worklet module with the AudioContext.\n * Only registers once per application lifetime.\n */\nasync function ensureModuleRegistered(ctx: AudioContext): Promise<void> {\n  if (moduleRegistered) {\n    return;\n  }\n\n  const blob = new Blob([processorCode], { type: \"application/javascript\" });\n  const url = URL.createObjectURL(blob);\n\n  try {\n    await ctx.audioWorklet.addModule(url);\n    moduleRegistered = true;\n  } finally {\n    URL.revokeObjectURL(url);\n  }\n}\n\n/**\n * Noise Source (NS)\n *\n * The noise source provides the aspiration component for breathy voice qualities.\n * It consists of Gaussian white noise filtered through a bandpass filter\n * (Butterworth, 1000–6000 Hz cutoff frequencies) and scaled by an amplitude factor.\n *\n * The noise can be modulated by the glottal flow derivative for mixed voice\n * qualities (not implemented in this basic version).\n *\n * Paper reference: Section 3.2.3 (filter structure), Section 4.2.5 (parameter mapping)\n *\n * Input parameters:\n *   - An (noise amplitude) — derived directly from B (breathiness), with\n *     scaling by E when voicing is off\n *\n * Signal flow:\n *   Gaussian Noise → Highpass (1000 Hz) → Lowpass (6000 Hz) → Output\n *\n * The bandpass is implemented as a cascade of 2nd-order Butterworth highpass\n * and lowpass filters using native BiquadFilterNodes.\n */\nexport class NoiseSource implements Node<NoiseSourceParams> {\n  private ctx: AudioContext;\n  private workletNode: AudioWorkletNode;\n  private highpassFilter: BiquadFilterNode;\n  private lowpassFilter: BiquadFilterNode;\n\n  public in: AudioNode | null = null; // Noise source has no input\n  public out: AudioNode;\n\n  private constructor(\n    ctx: AudioContext,\n    workletNode: AudioWorkletNode,\n    highpassFilter: BiquadFilterNode,\n    lowpassFilter: BiquadFilterNode\n  ) {\n    this.ctx = ctx;\n    this.workletNode = workletNode;\n    this.highpassFilter = highpassFilter;\n    this.lowpassFilter = lowpassFilter;\n    this.out = lowpassFilter;\n  }\n\n  /** Noise amplitude AudioParam (a-rate, 0-1) */\n  get An(): AudioParam {\n    return this.workletNode.parameters.get(\"An\")!;\n  }\n\n  /**\n   * Creates a new NoiseSource node.\n   *\n   * The AudioWorklet module is registered automatically on first use.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: NoiseSourceParams\n  ): Promise<NoiseSource> {\n    await ensureModuleRegistered(ctx);\n\n    // Create noise generator worklet\n    const workletNode = new AudioWorkletNode(ctx, \"noise-source-processor\");\n\n    // Create Butterworth bandpass as highpass + lowpass cascade\n    // Highpass at 1000 Hz\n    const highpassFilter = ctx.createBiquadFilter();\n    highpassFilter.type = \"highpass\";\n    highpassFilter.frequency.setValueAtTime(1000, ctx.currentTime);\n    highpassFilter.Q.setValueAtTime(Math.SQRT1_2, ctx.currentTime); // Butterworth Q = 1/√2\n\n    // Lowpass at 6000 Hz\n    const lowpassFilter = ctx.createBiquadFilter();\n    lowpassFilter.type = \"lowpass\";\n    lowpassFilter.frequency.setValueAtTime(6000, ctx.currentTime);\n    lowpassFilter.Q.setValueAtTime(Math.SQRT1_2, ctx.currentTime); // Butterworth Q = 1/√2\n\n    // Connect signal chain: noise → highpass → lowpass\n    workletNode.connect(highpassFilter);\n    highpassFilter.connect(lowpassFilter);\n\n    const node = new NoiseSource(ctx, workletNode, highpassFilter, lowpassFilter);\n    node.update(params);\n\n    return node;\n  }\n\n  /**\n   * Updates the noise source amplitude.\n   * Sets AudioParam via setTargetAtTime for smooth transitions.\n   */\n  update(params: NoiseSourceParams): void {\n    this.An.setTargetAtTime(params.An, this.ctx.currentTime, 0.02);\n  }\n\n  destroy(): void {\n    this.workletNode.disconnect();\n    this.highpassFilter.disconnect();\n    this.lowpassFilter.disconnect();\n  }\n}\n","import type { Node } from \"./types\";\nimport { PulseTrain } from \"./pulse-train\";\nimport { GlottalFormant } from \"./glottal-formant\";\nimport { SpectralTilt } from \"./spectral-tilt\";\nimport { NoiseSource } from \"./noise-source\";\nimport { Gain } from \"./gain\";\n\nexport type GlottalFlowDerivativeParams = {\n  /** Fundamental frequency in Hz (derived from P, P₀) */\n  f0: number;\n  /** Glottal formant centre frequency in Hz, computed as f0 / (2 * Oq) */\n  Fg: number;\n  /** Glottal formant bandwidth in Hz, computed from f0, Oq, and αm */\n  Bg: number;\n  /** Source amplitude, derived from E and Oq */\n  Ag: number;\n  /** First stage spectral tilt attenuation in dB at 3 kHz, derived from E and M */\n  Tl1: number;\n  /** Second stage spectral tilt attenuation in dB at 3 kHz, derived from E and M */\n  Tl2: number;\n  /** Noise amplitude, derived from B (breathiness) */\n  An: number;\n  /** Jitter depth: maximum f₀ perturbation as a fraction (0-0.3 for ±30%), derived from R */\n  jitterDepth: number;\n  /** Shimmer depth: maximum amplitude perturbation as a fraction (0-1 for ±100%), derived from R */\n  shimmerDepth: number;\n};\n\n/**\n * Glottal Flow Derivative (GFD)\n *\n * The glottal flow derivative model represents the sound source produced by\n * the vibrating vocal folds. It consists of a voiced component (periodic pulses\n * shaped by the glottal formant and spectral tilt) and an unvoiced component\n * (filtered noise for breathiness).\n *\n * The noise is modulated by the glottal flow derivative, so aspiration noise\n * follows the glottal cycle - loudest during the open phase, quiet when closed.\n *\n * Signal flow:\n *\n *   PulseTrain(f0) → GlottalFormant(Fg, Bg, Ag) → SpectralTilt(Tl1, Tl2) ──┬────→ Output\n *                                                                          │        ↑\n *                                                                          │ (mod)  │\n *                                                                          ↓        │\n *   NoiseSource(An) ─────────────────────────────────────────────────→ [Multiply] ──┘\n *\n * Paper reference: Section 3.2 (overall structure)\n *\n * Input parameters:\n *   - f0 (fundamental frequency) — controls the pulse train period\n *   - Fg (glottal formant centre frequency) — computed as f0 / (2·Oq)\n *   - Bg (glottal formant bandwidth) — computed from f0, Oq, and αm\n *   - Ag (source amplitude) — derived from E, Oq, and R (shimmer)\n *   - Tl1 (first tilt stage attenuation in dB at 3 kHz) — derived from E and M\n *   - Tl2 (second tilt stage attenuation in dB at 3 kHz) — derived from E and M\n *   - An (noise amplitude) — derived from B (breathiness)\n */\nexport class GlottalFlowDerivative implements Node<GlottalFlowDerivativeParams> {\n  private pulseTrain: PulseTrain;\n  private glottalFormant: GlottalFormant;\n  private spectralTilt: SpectralTilt;\n  private noiseSource: NoiseSource;\n  private noiseModulator: GainNode; // Multiplies noise by GFD signal\n  private outputGain: Gain;\n\n  public in: AudioNode | null = null; // Source node has no input\n  public out: AudioNode;\n\n  private constructor(\n    pulseTrain: PulseTrain,\n    glottalFormant: GlottalFormant,\n    spectralTilt: SpectralTilt,\n    noiseSource: NoiseSource,\n    noiseModulator: GainNode,\n    outputGain: Gain\n  ) {\n    this.pulseTrain = pulseTrain;\n    this.glottalFormant = glottalFormant;\n    this.spectralTilt = spectralTilt;\n    this.noiseSource = noiseSource;\n    this.noiseModulator = noiseModulator;\n    this.outputGain = outputGain;\n    this.out = outputGain.out;\n  }\n\n  /**\n   * Creates a new GlottalFlowDerivative node.\n   *\n   * Sets up the voiced path (pulse train → glottal formant → spectral tilt)\n   * and the noise modulation path where noise is multiplied by the GFD signal.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: GlottalFlowDerivativeParams\n  ): Promise<GlottalFlowDerivative> {\n    // Create all sub-nodes in parallel\n    const [pulseTrain, glottalFormant, spectralTilt, noiseSource, outputGain] =\n      await Promise.all([\n        PulseTrain.create(ctx, {\n          f0: params.f0,\n          jitterDepth: params.jitterDepth,\n          shimmerDepth: params.shimmerDepth,\n        }),\n        GlottalFormant.create(ctx, { Fg: params.Fg, Bg: params.Bg, Ag: params.Ag }),\n        SpectralTilt.create(ctx, { Tl1: params.Tl1, Tl2: params.Tl2 }),\n        NoiseSource.create(ctx, { An: params.An }),\n        Gain.create(ctx, { gain: 1 }),\n      ]);\n\n    // Create noise modulator - a GainNode whose gain is modulated by the GFD signal\n    // This implements the multiplication: noise × GFD\n    const noiseModulator = ctx.createGain();\n    noiseModulator.gain.setValueAtTime(0, ctx.currentTime); // Base gain is 0, modulated by audio signal\n\n    // Connect voiced path: PulseTrain → GlottalFormant → SpectralTilt → Output\n    pulseTrain.out.connect(glottalFormant.in);\n    glottalFormant.out.connect(spectralTilt.in);\n    spectralTilt.out.connect(outputGain.in);\n\n    // Connect GFD signal to modulate the noise gain (audio-rate modulation)\n    spectralTilt.out.connect(noiseModulator.gain);\n\n    // Connect noise through modulator to output: NoiseSource → [×GFD] → Output\n    noiseSource.out.connect(noiseModulator);\n    noiseModulator.connect(outputGain.in);\n\n    return new GlottalFlowDerivative(\n      pulseTrain,\n      glottalFormant,\n      spectralTilt,\n      noiseSource,\n      noiseModulator,\n      outputGain\n    );\n  }\n\n  /**\n   * Updates all glottal flow derivative parameters.\n   * Each sub-node receives its relevant parameters.\n   */\n  update(params: GlottalFlowDerivativeParams): void {\n    this.pulseTrain.update({\n      f0: params.f0,\n      jitterDepth: params.jitterDepth,\n      shimmerDepth: params.shimmerDepth,\n    });\n    this.glottalFormant.update({ Fg: params.Fg, Bg: params.Bg, Ag: params.Ag });\n    this.spectralTilt.update({ Tl1: params.Tl1, Tl2: params.Tl2 });\n    this.noiseSource.update({ An: params.An });\n  }\n\n  /**\n   * Returns the pulse train node for direct AudioParam access.\n   *\n   * Example usage:\n   *   gfd.pulseTrainNode.f0.setValueAtTime(440, ctx.currentTime);\n   *   gfd.pulseTrainNode.jitterDepth.linearRampToValueAtTime(0.1, ctx.currentTime + 0.5);\n   */\n  get pulseTrainNode(): PulseTrain {\n    return this.pulseTrain;\n  }\n\n  /**\n   * Returns the glottal formant node for direct AudioParam access.\n   *\n   * Example usage:\n   *   gfd.glottalFormantNode.Fg.setValueAtTime(220, ctx.currentTime);\n   *   gfd.glottalFormantNode.Ag.setTargetAtTime(0.5, ctx.currentTime, 0.1);\n   */\n  get glottalFormantNode(): GlottalFormant {\n    return this.glottalFormant;\n  }\n\n  /**\n   * Returns the spectral tilt node for direct AudioParam access.\n   *\n   * Example usage:\n   *   gfd.spectralTiltNode.Tl1.setValueAtTime(10, ctx.currentTime);\n   */\n  get spectralTiltNode(): SpectralTilt {\n    return this.spectralTilt;\n  }\n\n  /**\n   * Returns the noise source node for direct AudioParam access.\n   *\n   * Example usage:\n   *   gfd.noiseSourceNode.An.setValueAtTime(0.3, ctx.currentTime);\n   */\n  get noiseSourceNode(): NoiseSource {\n    return this.noiseSource;\n  }\n\n  /**\n   * Starts the voiced excitation (pulse train).\n   */\n  start(): void {\n    this.pulseTrain.start();\n  }\n\n  /**\n   * Stops the voiced excitation (pulse train).\n   * Note: Noise source continues to produce output based on An parameter.\n   */\n  stop(): void {\n    this.pulseTrain.stop();\n  }\n\n  destroy(): void {\n    this.pulseTrain.destroy();\n    this.glottalFormant.destroy();\n    this.spectralTilt.destroy();\n    this.noiseSource.destroy();\n    this.noiseModulator.disconnect();\n    this.outputGain.destroy();\n  }\n}\n","import type { Node } from \"./types\";\n\nexport type FormantResonatorParams = {\n  /** Formant centre frequency in Hz */\n  F: number;\n  /** Formant bandwidth in Hz */\n  B: number;\n  /** Formant amplitude (linear gain) */\n  A: number;\n};\n\n/**\n * Formant Resonator AudioWorklet Processor code.\n *\n * Implements the transfer function from Section 3.3.1:\n * R(z) = A * (1 - R) * (1 - R*z^{-2}) / (1 - 2R*cos(θ)*z^{-1} + R²*z^{-2})\n *\n * Where:\n *   R = e^(-π * B * Ts)  (pole radius, Ts = 1/sampleRate)\n *   θ = 2π * F * Ts      (pole angle)\n *\n * This is a 2-pole 2-zero digital resonator (bandpass filter).\n */\nconst processorCode = `\nclass FormantResonatorProcessor extends AudioWorkletProcessor {\n  static get parameterDescriptors() {\n    return [\n      { name: \"F\", defaultValue: 500, minValue: 100, maxValue: 8000, automationRate: \"a-rate\" },\n      { name: \"B\", defaultValue: 100, minValue: 20, maxValue: 1000, automationRate: \"a-rate\" },\n      { name: \"A\", defaultValue: 1, minValue: 0, maxValue: 10, automationRate: \"a-rate\" }\n    ];\n  }\n\n  constructor() {\n    super();\n    // Filter state (delay line)\n    this.x1 = 0;\n    this.x2 = 0;\n    this.y1 = 0;\n    this.y2 = 0;\n\n    // Cached parameter values for change detection\n    this.lastF = -1;\n    this.lastB = -1;\n    this.lastA = -1;\n\n    // Cached coefficients\n    this.b0 = 0;\n    this.b2 = 0;\n    this.a1 = 0;\n    this.a2 = 0;\n  }\n\n  process(inputs, outputs, parameters) {\n    const input = inputs[0]?.[0];\n    const output = outputs[0]?.[0];\n\n    if (!input || !output) {\n      return true;\n    }\n\n    // a-rate parameters (per-sample values)\n    const F = parameters.F[0];\n    const B = parameters.B[0];\n    const A = parameters.A[0];\n\n    // Recompute coefficients if any parameter changed\n    if (F !== this.lastF || B !== this.lastB || A !== this.lastA) {\n      const Ts = 1 / sampleRate;\n\n      // Pole radius and angle\n      const R = Math.exp(-Math.PI * B * Ts);\n      const theta = 2 * Math.PI * F * Ts;\n\n      // Gain scaling factor\n      const g = 1 - R;\n\n      // Numerator coefficients (feedforward)\n      // From: A * g * (1 - R*z^{-2}) = A*g + 0*z^{-1} - A*g*R*z^{-2}\n      this.b0 = A * g;\n      this.b2 = -A * g * R;\n\n      // Denominator coefficients (feedback)\n      // From: 1 - 2R*cos(θ)*z^{-1} + R²*z^{-2}\n      this.a1 = -2 * R * Math.cos(theta);\n      this.a2 = R * R;\n\n      this.lastF = F;\n      this.lastB = B;\n      this.lastA = A;\n    }\n\n    for (let i = 0; i < input.length; i++) {\n      const x0 = input[i];\n\n      // Direct Form I biquad (b1 is always 0)\n      const y0 = this.b0 * x0 + this.b2 * this.x2\n                 - this.a1 * this.y1 - this.a2 * this.y2;\n\n      // Update delay line\n      this.x2 = this.x1;\n      this.x1 = x0;\n      this.y2 = this.y1;\n      this.y1 = y0;\n\n      output[i] = y0;\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('formant-resonator-processor', FormantResonatorProcessor);\n`;\n\n// Track whether the worklet module has been registered\nlet moduleRegistered = false;\n\n/**\n * Registers the formant resonator worklet module with the AudioContext.\n * Only registers once per application lifetime.\n */\nasync function ensureModuleRegistered(ctx: AudioContext): Promise<void> {\n  if (moduleRegistered) {\n    return;\n  }\n\n  const blob = new Blob([processorCode], { type: \"application/javascript\" });\n  const url = URL.createObjectURL(blob);\n\n  try {\n    await ctx.audioWorklet.addModule(url);\n    moduleRegistered = true;\n  } finally {\n    URL.revokeObjectURL(url);\n  }\n}\n\n/**\n * Formant Resonator (Ri)\n *\n * A single formant filter representing one resonance of the vocal tract.\n * Implemented as a 2-pole 2-zero digital resonator (bandpass filter) with\n * controllable centre frequency (F), bandwidth (B), and amplitude (A).\n *\n * The transfer function (Section 3.3.1) is:\n *\n * R(z) = A * (1 - R) * (1 - R*z^{-2}) / (1 - 2R*cos(θ)*z^{-1} + R²*z^{-2})\n *\n * Where:\n *   R = e^(-π * B * Ts)  (pole radius, Ts = sampling period)\n *   θ = 2π * F * Ts      (pole angle)\n *\n * The vocal tract model uses six formant resonators (R1-R6) in parallel:\n *   - F1-F3: Primary formants that determine vowel identity\n *   - F4-F6: Higher formants that contribute to voice timbre\n *\n * Grouping F3, F4, and F5 closely together can produce the \"singer's formant\"\n * characteristic of trained classical voices.\n *\n * Paper reference: Section 3.3.1 (filter structure), Section 4.3.1 (vowel table),\n *                  Section 4.3.6 (bandwidths), Section 4.3.7 (amplitudes)\n *\n * Input parameters:\n *   - F (formant centre frequency in Hz) — interpolated from vowel table,\n *     scaled by αS (vocal tract size) and K (larynx position factor)\n *   - B (formant bandwidth in Hz) — interpolated from vowel table\n *   - A (formant amplitude, linear) — interpolated from vowel table,\n *     with correction when harmonics coincide with formant frequencies\n */\nexport class FormantResonator implements Node<FormantResonatorParams> {\n  private ctx: AudioContext;\n  private workletNode: AudioWorkletNode;\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(ctx: AudioContext, workletNode: AudioWorkletNode) {\n    this.ctx = ctx;\n    this.workletNode = workletNode;\n    this.in = workletNode;\n    this.out = workletNode;\n  }\n\n  /** Formant centre frequency AudioParam (a-rate, 100-8000 Hz) */\n  get F(): AudioParam {\n    return this.workletNode.parameters.get(\"F\")!;\n  }\n\n  /** Formant bandwidth AudioParam (a-rate, 20-1000 Hz) */\n  get B(): AudioParam {\n    return this.workletNode.parameters.get(\"B\")!;\n  }\n\n  /** Formant amplitude AudioParam (a-rate, 0-10) */\n  get A(): AudioParam {\n    return this.workletNode.parameters.get(\"A\")!;\n  }\n\n  /**\n   * Creates a new FormantResonator node.\n   *\n   * The AudioWorklet module is registered automatically on first use.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: FormantResonatorParams\n  ): Promise<FormantResonator> {\n    await ensureModuleRegistered(ctx);\n\n    const workletNode = new AudioWorkletNode(ctx, \"formant-resonator-processor\");\n    const node = new FormantResonator(ctx, workletNode);\n    node.update(params);\n\n    return node;\n  }\n\n  /**\n   * Updates the formant resonator parameters.\n   * Sets AudioParams via setTargetAtTime for smooth transitions.\n   */\n  update(params: FormantResonatorParams): void {\n    this.F.setTargetAtTime(params.F, this.ctx.currentTime, 0.02);\n    this.B.setTargetAtTime(params.B, this.ctx.currentTime, 0.02);\n    this.A.setTargetAtTime(params.A, this.ctx.currentTime, 0.02);\n  }\n\n  destroy(): void {\n    this.workletNode.disconnect();\n  }\n}\n","import type { Node } from \"./types\";\nimport { FormantResonator, FormantResonatorParams } from \"./formant-resonator\";\n\nexport type FormantBankParams = {\n  /** Array of formant parameters, one per resonator */\n  formants: FormantResonatorParams[];\n};\n\n/**\n * Formant Bank\n *\n * A parallel bank of formant resonators representing the vocal tract resonances.\n * Each resonator is a 2-pole 2-zero bandpass filter (see FormantResonator).\n *\n * The input signal is fed to all resonators in parallel, and their outputs\n * are summed to produce a single output.\n *\n * Signal flow:\n *\n *              ┌─→ R1(F1, B1, A1) ─┐\n *              ├─→ R2(F2, B2, A2) ─┤\n *   Input ─────┼─→ R3(F3, B3, A3) ─┼─────→ Output (sum)\n *              ├─→ ...            ─┤\n *              └─→ Rn(Fn, Bn, An) ─┘\n *\n * The paper uses six formants (R1-R6):\n *   - F1-F3: Primary formants that determine vowel identity\n *   - F4-F6: Higher formants that contribute to voice timbre\n *\n * This implementation accepts any number of formants, allowing flexibility\n * for experimentation or simplified models.\n *\n * Paper reference: Section 3.3.1 (parallel formant structure)\n *\n * Input parameters:\n *   - formants: Array of {F, B, A} objects defining each resonator\n *\n * Note: The number of formants is fixed at creation time. Calling update()\n * with a different array length will throw an error. To change the number\n * of formants, destroy this node and create a new one.\n */\nexport class FormantBank implements Node<FormantBankParams> {\n  private resonators: FormantResonator[];\n  private inputGain: GainNode;\n  private outputGain: GainNode;\n\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(\n    resonators: FormantResonator[],\n    inputGain: GainNode,\n    outputGain: GainNode\n  ) {\n    this.resonators = resonators;\n    this.inputGain = inputGain;\n    this.outputGain = outputGain;\n    this.in = inputGain;\n    this.out = outputGain;\n  }\n\n  /**\n   * Creates a new FormantBank node with the specified formants.\n   *\n   * All formant resonators are created in parallel for efficiency.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: FormantBankParams\n  ): Promise<FormantBank> {\n    // Create input and output gain nodes for routing\n    const inputGain = ctx.createGain();\n    inputGain.gain.setValueAtTime(1, ctx.currentTime);\n\n    const outputGain = ctx.createGain();\n    outputGain.gain.setValueAtTime(1, ctx.currentTime);\n\n    // Create all formant resonators in parallel\n    const resonators = await Promise.all(\n      params.formants.map((formant) => FormantResonator.create(ctx, formant))\n    );\n\n    // Wire up parallel structure: input → each resonator → output\n    for (const resonator of resonators) {\n      inputGain.connect(resonator.in);\n      resonator.out.connect(outputGain);\n    }\n\n    return new FormantBank(resonators, inputGain, outputGain);\n  }\n\n  /**\n   * Updates all formant parameters.\n   *\n   * The formants array must have the same length as the original.\n   * Each resonator receives its corresponding parameters.\n   *\n   * @throws Error if the formants array length doesn't match\n   */\n  update(params: FormantBankParams): void {\n    if (params.formants.length !== this.resonators.length) {\n      throw new Error(\n        `FormantBank: Cannot change number of formants. ` +\n          `Expected ${this.resonators.length}, got ${params.formants.length}. ` +\n          `Destroy and recreate the node to change formant count.`\n      );\n    }\n\n    for (let i = 0; i < this.resonators.length; i++) {\n      this.resonators[i].update(params.formants[i]);\n    }\n  }\n\n  /**\n   * Returns the number of formant resonators in this bank.\n   */\n  get count(): number {\n    return this.resonators.length;\n  }\n\n  /**\n   * Returns the array of formant resonators for direct AudioParam access.\n   *\n   * Example usage:\n   *   formantBank.formants[0].F.setValueAtTime(800, ctx.currentTime);\n   *   formantBank.formants[1].B.linearRampToValueAtTime(100, ctx.currentTime + 0.1);\n   */\n  get formants(): readonly FormantResonator[] {\n    return this.resonators;\n  }\n\n  destroy(): void {\n    // Disconnect routing nodes\n    this.inputGain.disconnect();\n    this.outputGain.disconnect();\n\n    // Destroy all resonators\n    for (const resonator of this.resonators) {\n      resonator.destroy();\n    }\n\n    this.resonators = [];\n  }\n}\n","import type { Node } from \"./types\";\n\nexport type AntiResonanceParams = {\n  /** Anti-formant centre frequency in Hz (nominally 4700 Hz, scaled by αS) */\n  F: number;\n  /** Anti-formant quality factor (nominally 2.5) */\n  Q: number;\n};\n\n/**\n * Anti-Resonance (Notch Filter) AudioWorklet Processor code.\n *\n * Implements the transfer function from Section 3.3.2:\n * BQ(z) = (1 + β*z⁻¹ + z⁻²) / ((1 + α) + β*z⁻¹ + (1 - α)*z⁻²)\n *\n * Where:\n *   α = sin(2π * F * Ts) / (2 * Q)\n *   β = -2 * cos(2π * F * Ts)\n *\n * This creates a notch (anti-resonance) at frequency F with width controlled by Q.\n */\nconst processorCode = `\nclass AntiResonanceProcessor extends AudioWorkletProcessor {\n  static get parameterDescriptors() {\n    return [\n      { name: \"F\", defaultValue: 4700, minValue: 1000, maxValue: 10000, automationRate: \"k-rate\" },\n      { name: \"Q\", defaultValue: 2.5, minValue: 0.5, maxValue: 20, automationRate: \"k-rate\" }\n    ];\n  }\n\n  constructor() {\n    super();\n    // Filter state (delay line)\n    this.x1 = 0;\n    this.x2 = 0;\n    this.y1 = 0;\n    this.y2 = 0;\n\n    // Cached parameter values for change detection\n    this.lastF = -1;\n    this.lastQ = -1;\n\n    // Cached coefficients\n    this.b0 = 1;\n    this.b1 = 0;\n    this.b2 = 1;\n    this.a1 = 0;\n    this.a2 = 0;\n  }\n\n  process(inputs, outputs, parameters) {\n    const input = inputs[0]?.[0];\n    const output = outputs[0]?.[0];\n\n    if (!input || !output) {\n      return true;\n    }\n\n    // k-rate parameters (single value per block)\n    const F = parameters.F[0];\n    const Q = parameters.Q[0];\n\n    // Recompute coefficients if any parameter changed\n    if (F !== this.lastF || Q !== this.lastQ) {\n      const Ts = 1 / sampleRate;\n      const omega = 2 * Math.PI * F * Ts;\n\n      // Intermediate coefficients from the paper\n      const alpha = Math.sin(omega) / (2 * Q);\n      const beta = -2 * Math.cos(omega);\n\n      // Normalize by dividing by (1 + alpha) so that a0 = 1\n      const a0 = 1 + alpha;\n\n      // Numerator coefficients (feedforward)\n      // From: (1 + β*z⁻¹ + z⁻²) / a0\n      this.b0 = 1 / a0;\n      this.b1 = beta / a0;\n      this.b2 = 1 / a0;\n\n      // Denominator coefficients (feedback), normalized\n      // From: ((1 + α) + β*z⁻¹ + (1 - α)*z⁻²) / a0\n      // a0/a0 = 1 (implicit), a1 = β/a0, a2 = (1 - α)/a0\n      this.a1 = beta / a0;\n      this.a2 = (1 - alpha) / a0;\n\n      this.lastF = F;\n      this.lastQ = Q;\n    }\n\n    for (let i = 0; i < input.length; i++) {\n      const x0 = input[i];\n\n      // Direct Form I biquad\n      const y0 = this.b0 * x0 + this.b1 * this.x1 + this.b2 * this.x2\n                 - this.a1 * this.y1 - this.a2 * this.y2;\n\n      // Update delay line\n      this.x2 = this.x1;\n      this.x1 = x0;\n      this.y2 = this.y1;\n      this.y1 = y0;\n\n      output[i] = y0;\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('anti-resonance-processor', AntiResonanceProcessor);\n`;\n\n// Track whether the worklet module has been registered\nlet moduleRegistered = false;\n\n/**\n * Registers the anti-resonance worklet module with the AudioContext.\n * Only registers once per application lifetime.\n */\nasync function ensureModuleRegistered(ctx: AudioContext): Promise<void> {\n  if (moduleRegistered) {\n    return;\n  }\n\n  const blob = new Blob([processorCode], { type: \"application/javascript\" });\n  const url = URL.createObjectURL(blob);\n\n  try {\n    await ctx.audioWorklet.addModule(url);\n    moduleRegistered = true;\n  } finally {\n    URL.revokeObjectURL(url);\n  }\n}\n\n/**\n * Hypo-pharynx Anti-Resonance Filter (BQ)\n *\n * A notch filter in cascade with the parallel formants that models the\n * anti-resonances caused by the hypo-pharynx cavity (the space between\n * the larynx and the pharynx). This creates spectral dips at approximately\n * 2.5–3.5 kHz and 4–5 kHz that contribute to natural voice timbre.\n *\n * The transfer function (Section 3.3.2) is:\n *\n * BQ(z) = (1 + β*z⁻¹ + z⁻²) / ((1 + α) + β*z⁻¹ + (1 - α)*z⁻²)\n *\n * Where:\n *   α = sin(2π * F * Ts) / (2 * Q)\n *   β = -2 * cos(2π * F * Ts)\n *\n * This is a bi-quadratic second-order notch filter that attenuates\n * frequencies near the centre frequency F, with the notch width\n * controlled by the quality factor Q.\n *\n * Paper reference: Section 3.3.2 (filter structure), Section 4.3.8 (parameter values)\n *\n * Input parameters:\n *   - F (anti-formant centre frequency in Hz) — nominally 4700 Hz,\n *     scaled by αS (vocal tract size factor)\n *   - Q (anti-formant quality factor) — fixed at 2.5 in the paper\n */\nexport class AntiResonance implements Node<AntiResonanceParams> {\n  private ctx: AudioContext;\n  private workletNode: AudioWorkletNode;\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(ctx: AudioContext, workletNode: AudioWorkletNode) {\n    this.ctx = ctx;\n    this.workletNode = workletNode;\n    this.in = workletNode;\n    this.out = workletNode;\n  }\n\n  /** Anti-formant centre frequency AudioParam (k-rate, 1000-10000 Hz) */\n  get F(): AudioParam {\n    return this.workletNode.parameters.get(\"F\")!;\n  }\n\n  /** Anti-formant quality factor AudioParam (k-rate, 0.5-20) */\n  get Q(): AudioParam {\n    return this.workletNode.parameters.get(\"Q\")!;\n  }\n\n  /**\n   * Creates a new AntiResonance node.\n   *\n   * The AudioWorklet module is registered automatically on first use.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: AntiResonanceParams\n  ): Promise<AntiResonance> {\n    await ensureModuleRegistered(ctx);\n\n    const workletNode = new AudioWorkletNode(ctx, \"anti-resonance-processor\");\n    const node = new AntiResonance(ctx, workletNode);\n    node.update(params);\n\n    return node;\n  }\n\n  /**\n   * Updates the anti-resonance parameters.\n   * Sets AudioParams via setTargetAtTime for smooth transitions.\n   */\n  update(params: AntiResonanceParams): void {\n    this.F.setTargetAtTime(params.F, this.ctx.currentTime, 0.02);\n    this.Q.setTargetAtTime(params.Q, this.ctx.currentTime, 0.02);\n  }\n\n  destroy(): void {\n    this.workletNode.disconnect();\n  }\n}\n","import type { Node } from \"./types\";\nimport { FormantBank } from \"./formant-bank\";\nimport { FormantResonator, FormantResonatorParams } from \"./formant-resonator\";\nimport { AntiResonance } from \"./anti-resonance\";\n\nexport type VocalTractParams = {\n  /** Array of formant parameters (F, B, A) for each resonator */\n  formants: FormantResonatorParams[];\n  /** Anti-formant centre frequency in Hz (nominally 4700 Hz, scaled by αS) */\n  F_BQ: number;\n  /** Anti-formant quality factor (nominally 2.5) */\n  Q_BQ: number;\n};\n\n/**\n * Vocal Tract Model\n *\n * The vocal tract model shapes the glottal source spectrum to produce\n * recognisable vowels and voice timbres. It consists of parallel formant\n * resonators (simulating the resonances of the oral cavity) followed by\n * a cascaded anti-resonance filter (simulating the effect of the hypo-pharynx).\n *\n * Signal flow:\n *\n *              ┌─→ R1(F1, B1, A1) ─┐\n *              ├─→ R2(F2, B2, A2) ─┤\n *   Input ─────┼─→ R3(F3, B3, A3) ─┼─────→ BQ(F_BQ, Q_BQ) ─────→ Output\n *              ├─→ ...            ─┤\n *              └─→ Rn(Fn, Bn, An) ─┘\n *\n * The paper uses six formants (R1-R6):\n *   - F1-F3: Primary formants that determine vowel identity\n *   - F4-F6: Higher formants that contribute to voice timbre\n *\n * The anti-resonance (BQ) models spectral dips caused by the hypo-pharynx\n * cavity at approximately 2.5–3.5 kHz and 4–5 kHz.\n *\n * Paper reference: Section 3.3 (overall structure), Section 3.3.1 (formants),\n *                  Section 3.3.2 (anti-resonance)\n *\n * Input parameters:\n *   - formants: Array of {F, B, A} objects for each resonator\n *   - F_BQ: Anti-formant centre frequency (nominally 4700 Hz, scaled by αS)\n *   - Q_BQ: Anti-formant quality factor (nominally 2.5)\n */\nexport class VocalTract implements Node<VocalTractParams> {\n  private formantBank: FormantBank;\n  private antiResonance: AntiResonance;\n\n  public in: AudioNode;\n  public out: AudioNode;\n\n  private constructor(\n    formantBank: FormantBank,\n    antiResonance: AntiResonance\n  ) {\n    this.formantBank = formantBank;\n    this.antiResonance = antiResonance;\n    this.in = formantBank.in;\n    this.out = antiResonance.out;\n  }\n\n  /**\n   * Creates a new VocalTract node.\n   *\n   * Sets up the parallel formant bank followed by the anti-resonance filter.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: VocalTractParams\n  ): Promise<VocalTract> {\n    // Create sub-nodes in parallel\n    const [formantBank, antiResonance] = await Promise.all([\n      FormantBank.create(ctx, { formants: params.formants }),\n      AntiResonance.create(ctx, { F: params.F_BQ, Q: params.Q_BQ }),\n    ]);\n\n    // Connect in cascade: FormantBank → AntiResonance\n    formantBank.out.connect(antiResonance.in);\n\n    return new VocalTract(formantBank, antiResonance);\n  }\n\n  /**\n   * Updates all vocal tract parameters.\n   *\n   * @throws Error if the formants array length doesn't match the original\n   */\n  update(params: VocalTractParams): void {\n    this.formantBank.update({ formants: params.formants });\n    this.antiResonance.update({ F: params.F_BQ, Q: params.Q_BQ });\n  }\n\n  /**\n   * Returns the number of formant resonators in the vocal tract.\n   */\n  get formantCount(): number {\n    return this.formantBank.count;\n  }\n\n  /**\n   * Returns the array of formant resonators for direct AudioParam access.\n   *\n   * Example usage:\n   *   vocalTract.formants[0].F.setValueAtTime(800, ctx.currentTime);\n   *   vocalTract.formants[2].B.linearRampToValueAtTime(100, ctx.currentTime + 0.1);\n   */\n  get formants(): readonly FormantResonator[] {\n    return this.formantBank.formants;\n  }\n\n  /**\n   * Returns the anti-resonance node for direct AudioParam access.\n   *\n   * Example usage:\n   *   vocalTract.antiResonanceNode.F.setValueAtTime(4500, ctx.currentTime);\n   *   vocalTract.antiResonanceNode.Q.setValueAtTime(3, ctx.currentTime);\n   */\n  get antiResonanceNode(): AntiResonance {\n    return this.antiResonance;\n  }\n\n  destroy(): void {\n    this.formantBank.destroy();\n    this.antiResonance.destroy();\n  }\n}\n","import type { Node } from \"./types\";\nimport { GlottalFlowDerivative } from \"./glottal-flow-derivative\";\nimport { VocalTract } from \"./vocal-tract\";\nimport { Gain } from \"./gain\";\nimport { SynthParams } from \"../parameters\";\n\n/**\n * Voice Synthesizer\n *\n * The complete voice synthesis pipeline combining the glottal flow derivative\n * (voice source) with the vocal tract model (formant filtering). This implements\n * the full source-filter model described in the paper.\n *\n * Signal flow:\n *\n *   ┌─────────────────────────────────────────────────────────────────────────┐\n *   │ Glottal Flow Derivative (Source)                                        │\n *   │                                                                         │\n *   │   PulseTrain(f0) → GlottalFormant(Fg,Bg,Ag) → SpectralTilt(Tl1,Tl2) ─┬─┤\n *   │                                                          ↓ (mod)     │ │\n *   │   NoiseSource(An) ──────────────────────────────────→ [×] ───────────┘ │\n *   └─────────────────────────────────────────────────────────────────────────┘\n *                                        │\n *                                        ↓\n *   ┌─────────────────────────────────────────────────────────────────────────┐\n *   │ Vocal Tract (Filter)                                                    │\n *   │                                                                         │\n *   │            ┌─→ R1(F1, B1, A1) ─┐                                        │\n *   │            ├─→ R2(F2, B2, A2) ─┤                                        │\n *   │   Input ───┼─→ R3(F3, B3, A3) ─┼───→ BQ(F_BQ, Q_BQ) ───→ Output        │\n *   │            ├─→ ...            ─┤                                        │\n *   │            └─→ Rn(Fn, Bn, An) ─┘                                        │\n *   └─────────────────────────────────────────────────────────────────────────┘\n *                                        │\n *                                        ↓\n *                                   OutputGain ───→ Output\n *\n * Paper reference: Section 3.1 (overall model), Section 3.2 (voice source),\n *                  Section 3.3 (vocal tract)\n *\n * Input parameters:\n *   - source: GlottalFlowDerivativeParams (f0, Fg, Bg, Ag, Tl1, Tl2, An)\n *   - tract: VocalTractParams (formants array, F_BQ, Q_BQ)\n *   - outputGain: Overall output level (optional, defaults to 1)\n */\nexport class Voice implements Node<SynthParams> {\n  private glottalFlowDerivative: GlottalFlowDerivative;\n  private vocalTract: VocalTract;\n  private outputGainNode: Gain;\n\n  public in: AudioNode | null = null; // Source node has no input\n  public out: AudioNode;\n\n  private constructor(\n    glottalFlowDerivative: GlottalFlowDerivative,\n    vocalTract: VocalTract,\n    outputGainNode: Gain\n  ) {\n    this.glottalFlowDerivative = glottalFlowDerivative;\n    this.vocalTract = vocalTract;\n    this.outputGainNode = outputGainNode;\n    this.out = outputGainNode.out;\n  }\n\n  /**\n   * Creates a new Voice synthesizer node.\n   *\n   * Sets up the complete source-filter synthesis pipeline.\n   */\n  static async create(\n    ctx: AudioContext,\n    params: SynthParams\n  ): Promise<Voice> {\n    // Create all sub-nodes in parallel\n    const [glottalFlowDerivative, vocalTract, outputGainNode] = await Promise.all([\n      GlottalFlowDerivative.create(ctx, params),\n      VocalTract.create(ctx, params),\n      Gain.create(ctx, { gain: params.outputGain ?? 1 }),\n    ]);\n\n    // Connect the pipeline: GFD → VocalTract → OutputGain\n    glottalFlowDerivative.out.connect(vocalTract.in);\n    vocalTract.out.connect(outputGainNode.in);\n\n    return new Voice(glottalFlowDerivative, vocalTract, outputGainNode);\n  }\n\n  /**\n   * Updates all voice parameters.\n   *\n   * @throws Error if the formants array length doesn't match the original\n   */\n  update(params: SynthParams): void {\n    this.glottalFlowDerivative.update(params);\n    this.vocalTract.update(params);\n    if (params.outputGain !== undefined) {\n      this.outputGainNode.update({ gain: params.outputGain });\n    }\n  }\n\n  /**\n   * Starts the voice (begins voiced excitation).\n   */\n  start(): void {\n    this.glottalFlowDerivative.start();\n  }\n\n  /**\n   * Stops the voice (stops voiced excitation).\n   * Note: Noise component continues based on An parameter.\n   */\n  stop(): void {\n    this.glottalFlowDerivative.stop();\n  }\n\n  /**\n   * Returns the number of formant resonators in the vocal tract.\n   */\n  get formantCount(): number {\n    return this.vocalTract.formantCount;\n  }\n\n  /**\n   * Returns the glottal flow derivative node for direct AudioParam access.\n   *\n   * Example usage:\n   *   voice.source.pulseTrainNode.f0.setValueAtTime(440, ctx.currentTime);\n   *   voice.source.glottalFormantNode.Ag.setTargetAtTime(0.5, ctx.currentTime, 0.1);\n   */\n  get source(): GlottalFlowDerivative {\n    return this.glottalFlowDerivative;\n  }\n\n  /**\n   * Returns the vocal tract node for direct AudioParam access.\n   *\n   * Example usage:\n   *   voice.tract.formants[0].F.linearRampToValueAtTime(800, ctx.currentTime + 0.1);\n   *   voice.tract.antiResonanceNode.F.setValueAtTime(4500, ctx.currentTime);\n   */\n  get tract(): VocalTract {\n    return this.vocalTract;\n  }\n\n  /**\n   * Returns the output gain node for direct AudioParam access.\n   *\n   * Example usage:\n   *   voice.outputGain.gain.linearRampToValueAtTime(0.5, ctx.currentTime + 1);\n   */\n  get outputGain(): Gain {\n    return this.outputGainNode;\n  }\n\n  destroy(): void {\n    this.glottalFlowDerivative.destroy();\n    this.vocalTract.destroy();\n    this.outputGainNode.destroy();\n  }\n}\n","/**\n * Vowel formant data from Cantor Digitalis Table 3\n *\n * Reference: Section 4.3.1 \"Generic Formant Values\"\n *\n * The sixth formant is derived: F6 = 2*F4, A6 = -15 dB, B6 = 150 Hz\n */\n\nexport interface Formant {\n  frequency: number; // Hz\n  amplitude: number; // dB\n  bandwidth: number; // Hz\n}\n\nexport interface VowelData {\n  ipa: string;\n  v: number; // backness: 0 = back, 1 = front\n  h: number; // height: 0 = close, 1 = open\n  formants: [Formant, Formant, Formant, Formant, Formant, Formant];\n}\n\n// Grid structure for interpolation\n// V levels: 0 (back), 0.5 (central), 1 (front)\n// H levels: 0 (close), 1/3, 2/3, 1 (open)\nconst V_LEVELS = [0, 0.5, 1];\nconst H_LEVELS = [0, 1 / 3, 2 / 3, 1];\n\n// Vowel grid indexed by [h_index][v_index]\n// At H=1, only /a/ exists (V=0.5), so we treat it as spanning the full width\nconst VOWEL_GRID: (string | null)[][] = [\n  [\"u\", \"y\", \"i\"], // H = 0\n  [\"o\", \"œ\", \"e\"], // H = 1/3\n  [\"ɔ\", \"ø\", \"ɛ\"], // H = 2/3\n  [\"a\", \"a\", \"a\"], // H = 1 (all map to /a/)\n];\n\nfunction findVowelByIpa(ipa: string): VowelData | undefined {\n  return vowels.find((v) => v.ipa === ipa);\n}\n\nfunction lerp(a: number, b: number, t: number): number {\n  return a + (b - a) * t;\n}\n\nfunction lerpFormant(a: Formant, b: Formant, t: number): Formant {\n  return {\n    frequency: lerp(a.frequency, b.frequency, t),\n    amplitude: lerp(a.amplitude, b.amplitude, t),\n    bandwidth: lerp(a.bandwidth, b.bandwidth, t),\n  };\n}\n\nfunction lerpFormants(\n  a: Formant[],\n  b: Formant[],\n  t: number\n): [Formant, Formant, Formant, Formant, Formant, Formant] {\n  return a.map((f, i) => lerpFormant(f, b[i], t)) as [\n    Formant,\n    Formant,\n    Formant,\n    Formant,\n    Formant,\n    Formant,\n  ];\n}\n\n/**\n * Interpolate formants for arbitrary vowel coordinates using bilinear interpolation.\n *\n * @param v - Vowel backness (0 = back, 1 = front)\n * @param h - Vowel height (0 = close, 1 = open)\n * @returns Interpolated formant array\n */\nexport function interpolateFormants(\n  v: number,\n  h: number\n): [Formant, Formant, Formant, Formant, Formant, Formant] {\n  // Clamp inputs to valid range\n  v = Math.max(0, Math.min(1, v));\n  h = Math.max(0, Math.min(1, h));\n\n  // Find surrounding grid indices\n  let hLow = 0;\n  let hHigh = 1;\n  for (let i = 0; i < H_LEVELS.length - 1; i++) {\n    if (h >= H_LEVELS[i] && h <= H_LEVELS[i + 1]) {\n      hLow = i;\n      hHigh = i + 1;\n      break;\n    }\n  }\n\n  let vLow = 0;\n  let vHigh = 1;\n  for (let i = 0; i < V_LEVELS.length - 1; i++) {\n    if (v >= V_LEVELS[i] && v <= V_LEVELS[i + 1]) {\n      vLow = i;\n      vHigh = i + 1;\n      break;\n    }\n  }\n\n  // Get the four corner vowels\n  const bottomLeft = findVowelByIpa(VOWEL_GRID[hLow][vLow]!)!;\n  const bottomRight = findVowelByIpa(VOWEL_GRID[hLow][vHigh]!)!;\n  const topLeft = findVowelByIpa(VOWEL_GRID[hHigh][vLow]!)!;\n  const topRight = findVowelByIpa(VOWEL_GRID[hHigh][vHigh]!)!;\n\n  // Calculate interpolation factors\n  const hRange = H_LEVELS[hHigh] - H_LEVELS[hLow];\n  const vRange = V_LEVELS[vHigh] - V_LEVELS[vLow];\n\n  const tH = hRange > 0 ? (h - H_LEVELS[hLow]) / hRange : 0;\n  const tV = vRange > 0 ? (v - V_LEVELS[vLow]) / vRange : 0;\n\n  // Bilinear interpolation\n  const bottom = lerpFormants(bottomLeft.formants, bottomRight.formants, tV);\n  const top = lerpFormants(topLeft.formants, topRight.formants, tV);\n\n  return lerpFormants(bottom, top, tH);\n}\n\nexport const vowels: VowelData[] = [\n  {\n    ipa: \"i\",\n    v: 1,\n    h: 0,\n    formants: [\n      { frequency: 215, amplitude: -10, bandwidth: 10 },\n      { frequency: 1900, amplitude: -10, bandwidth: 18 },\n      { frequency: 2630, amplitude: -8, bandwidth: 20 },\n      { frequency: 3170, amplitude: -4, bandwidth: 30 },\n      { frequency: 3710, amplitude: -15, bandwidth: 40 },\n      { frequency: 6340, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"e\",\n    v: 1,\n    h: 1 / 3,\n    formants: [\n      { frequency: 410, amplitude: -1, bandwidth: 10 },\n      { frequency: 2000, amplitude: -3, bandwidth: 15 },\n      { frequency: 2570, amplitude: -2, bandwidth: 20 },\n      { frequency: 2980, amplitude: -2, bandwidth: 30 },\n      { frequency: 3900, amplitude: -5, bandwidth: 40 },\n      { frequency: 5960, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"ɛ\",\n    v: 1,\n    h: 2 / 3,\n    formants: [\n      { frequency: 590, amplitude: 0, bandwidth: 10 },\n      { frequency: 1700, amplitude: -4, bandwidth: 15 },\n      { frequency: 2540, amplitude: -5, bandwidth: 30 },\n      { frequency: 2800, amplitude: -12, bandwidth: 50 },\n      { frequency: 3900, amplitude: -24, bandwidth: 40 },\n      { frequency: 5600, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"y\",\n    v: 0.5,\n    h: 0,\n    formants: [\n      { frequency: 250, amplitude: -12, bandwidth: 10 },\n      { frequency: 1750, amplitude: -9, bandwidth: 10 },\n      { frequency: 2160, amplitude: -14, bandwidth: 20 },\n      { frequency: 3060, amplitude: -11, bandwidth: 30 },\n      { frequency: 3900, amplitude: -11, bandwidth: 40 },\n      { frequency: 6120, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"œ\",\n    v: 0.5,\n    h: 1 / 3,\n    formants: [\n      { frequency: 350, amplitude: -6, bandwidth: 10 },\n      { frequency: 1350, amplitude: -3, bandwidth: 10 },\n      { frequency: 2250, amplitude: -8, bandwidth: 20 },\n      { frequency: 3170, amplitude: -8, bandwidth: 30 },\n      { frequency: 3900, amplitude: -10, bandwidth: 40 },\n      { frequency: 6340, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"ø\",\n    v: 0.5,\n    h: 2 / 3,\n    formants: [\n      { frequency: 620, amplitude: -3, bandwidth: 10 },\n      { frequency: 1300, amplitude: -3, bandwidth: 10 },\n      { frequency: 2520, amplitude: -3, bandwidth: 20 },\n      { frequency: 3310, amplitude: -7, bandwidth: 30 },\n      { frequency: 3900, amplitude: -14, bandwidth: 40 },\n      { frequency: 6620, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"u\",\n    v: 0,\n    h: 0,\n    formants: [\n      { frequency: 290, amplitude: -6, bandwidth: 10 },\n      { frequency: 750, amplitude: -8, bandwidth: 10 },\n      { frequency: 2300, amplitude: -13, bandwidth: 20 },\n      { frequency: 3080, amplitude: -8, bandwidth: 30 },\n      { frequency: 3900, amplitude: -9, bandwidth: 40 },\n      { frequency: 6160, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"o\",\n    v: 0,\n    h: 1 / 3,\n    formants: [\n      { frequency: 440, amplitude: -6, bandwidth: 10 },\n      { frequency: 750, amplitude: -1, bandwidth: 12 },\n      { frequency: 2160, amplitude: -10, bandwidth: 20 },\n      { frequency: 2860, amplitude: -6, bandwidth: 30 },\n      { frequency: 3900, amplitude: -28, bandwidth: 40 },\n      { frequency: 5720, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"ɔ\",\n    v: 0,\n    h: 2 / 3,\n    formants: [\n      { frequency: 610, amplitude: -3, bandwidth: 10 },\n      { frequency: 950, amplitude: 0, bandwidth: 12 },\n      { frequency: 2510, amplitude: -12, bandwidth: 20 },\n      { frequency: 2830, amplitude: -15, bandwidth: 30 },\n      { frequency: 3900, amplitude: -20, bandwidth: 40 },\n      { frequency: 5660, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n  {\n    ipa: \"a\",\n    v: 0.5,\n    h: 1,\n    formants: [\n      { frequency: 700, amplitude: 0, bandwidth: 13 },\n      { frequency: 1200, amplitude: 0, bandwidth: 13 },\n      { frequency: 2500, amplitude: -5, bandwidth: 40 },\n      { frequency: 2800, amplitude: -7, bandwidth: 60 },\n      { frequency: 3600, amplitude: -24, bandwidth: 40 },\n      { frequency: 5600, amplitude: -15, bandwidth: 150 },\n    ],\n  },\n];\n","/**\n * Perceptual and Synth Parameters\n *\n * There are high-level, user-controllable parameters for the voice synthesiser.\n * They map to the perceptually meaningful dimensions described in Section 2.1 of the paper.\n * These are converted to the low-level synthesis parameters required by\n * the voice engine using the generateSynthParams function.\n *\n * All normalised parameters range from 0 to 1 unless otherwise noted.\n */\n\nimport { interpolateFormants, Formant } from \"./vowels\";\n\nexport type SynthFormant = {\n  /** Formant centre frequency in Hz */\n  F: number;\n  /** Formant bandwidth in Hz */\n  B: number;\n  /** Formant amplitude (linear gain) */\n  A: number;\n};\n\nexport type SynthParams = {\n  /** Fundamental frequency in Hz */\n  f0: number;\n  /** Glottal formant centre frequency in Hz */\n  Fg: number;\n  /** Glottal formant bandwidth in Hz */\n  Bg: number;\n  /** Source amplitude */\n  Ag: number;\n  /** First stage spectral tilt attenuation in dB at 3 kHz */\n  Tl1: number;\n  /** Second stage spectral tilt attenuation in dB at 3 kHz */\n  Tl2: number;\n  /** Noise amplitude */\n  An: number;\n  /** Jitter depth: maximum f₀ perturbation as a fraction (0-0.3 for ±30%) */\n  jitterDepth: number;\n  /** Shimmer depth: maximum amplitude perturbation as a fraction (0-1 for ±100%) */\n  shimmerDepth: number;\n  /** Array of formant parameters (F, B, A) for each resonator */\n  formants: SynthFormant[];\n  /** Anti-formant centre frequency in Hz */\n  F_BQ: number;\n  /** Anti-formant quality factor */\n  Q_BQ: number;\n  /** Overall output gain (linear) */\n  outputGain?: number;\n}\n\nexport type PerceptualParams = {\n  /** (P) Normalised melodic position across the pitch range (0-1) */\n  pitch: number;\n  /** (P₀) Base MIDI note number (e.g., 48 for C3, 60 for C4) */\n  pitchOffset: number;\n  /** (E) Perceived force/dynamics of the voice (0-1) */\n  vocalEffort: number;\n  /** (H) Vertical tongue position: 0 = close (e.g., /i/, /u/), 1 = open (e.g., /a/) */\n  vowelHeight: number;\n  /** (V) Horizontal tongue position: 0 = back (e.g., /u/), 1 = front (e.g., /i/) */\n  vowelBackness: number;\n  /** (T) Degree of vocal fold adduction: 0 = lax, 1 = tense */\n  tenseness: number;\n  /** (B) Amount of aspiration noise (0-1) */\n  breathiness: number;\n  /** (R) Structural aperiodicities causing jitter/shimmer (0-1) */\n  roughness: number;\n  /** (S) Apparent size of the vocal tract: 0 = small/child, 1 = large/giant */\n  vocalTractSize: number;\n  /** (M) Vocal fold vibration mode: false = chest voice (M1), true = falsetto (M2) */\n  isFalsetto: boolean;\n};\n\n/**\n * Feature flags for the voice parameter conversion.\n * All features are enabled by default (true). 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package/dist/nodes/anti-resonance.d.ts

@@ -0,0 +1,58 @@
+import type { Node } from "./types";
+export type AntiResonanceParams = {
+    /** Anti-formant centre frequency in Hz (nominally 4700 Hz, scaled by αS) */
+    F: number;
+    /** Anti-formant quality factor (nominally 2.5) */
+    Q: number;
+};
+/**
+ * Hypo-pharynx Anti-Resonance Filter (BQ)
+ *
+ * A notch filter in cascade with the parallel formants that models the
+ * anti-resonances caused by the hypo-pharynx cavity (the space between
+ * the larynx and the pharynx). This creates spectral dips at approximately
+ * 2.5–3.5 kHz and 4–5 kHz that contribute to natural voice timbre.
+ *
+ * The transfer function (Section 3.3.2) is:
+ *
+ * BQ(z) = (1 + β*z⁻¹ + z⁻²) / ((1 + α) + β*z⁻¹ + (1 - α)*z⁻²)
+ *
+ * Where:
+ *   α = sin(2π * F * Ts) / (2 * Q)
+ *   β = -2 * cos(2π * F * Ts)
+ *
+ * This is a bi-quadratic second-order notch filter that attenuates
+ * frequencies near the centre frequency F, with the notch width
+ * controlled by the quality factor Q.
+ *
+ * Paper reference: Section 3.3.2 (filter structure), Section 4.3.8 (parameter values)
+ *
+ * Input parameters:
+ *   - F (anti-formant centre frequency in Hz) — nominally 4700 Hz,
+ *     scaled by αS (vocal tract size factor)
+ *   - Q (anti-formant quality factor) — fixed at 2.5 in the paper
+ */
+export declare class AntiResonance implements Node<AntiResonanceParams> {
+    private ctx;
+    private workletNode;
+    in: AudioNode;
+    out: AudioNode;
+    private constructor();
+    /** Anti-formant centre frequency AudioParam (k-rate, 1000-10000 Hz) */
+    get F(): AudioParam;
+    /** Anti-formant quality factor AudioParam (k-rate, 0.5-20) */
+    get Q(): AudioParam;
+    /**
+     * Creates a new AntiResonance node.
+     *
+     * The AudioWorklet module is registered automatically on first use.
+     */
+    static create(ctx: AudioContext, params: AntiResonanceParams): Promise<AntiResonance>;
+    /**
+     * Updates the anti-resonance parameters.
+     * Sets AudioParams via setTargetAtTime for smooth transitions.
+     */
+    update(params: AntiResonanceParams): void;
+    destroy(): void;
+}
+//# sourceMappingURL=anti-resonance.d.ts.map

package/dist/nodes/anti-resonance.d.ts.map

@@ -0,0 +1 @@
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package/dist/nodes/formant-bank.d.ts

@@ -0,0 +1,76 @@
+import type { Node } from "./types";
+import { FormantResonator, FormantResonatorParams } from "./formant-resonator";
+export type FormantBankParams = {
+    /** Array of formant parameters, one per resonator */
+    formants: FormantResonatorParams[];
+};
+/**
+ * Formant Bank
+ *
+ * A parallel bank of formant resonators representing the vocal tract resonances.
+ * Each resonator is a 2-pole 2-zero bandpass filter (see FormantResonator).
+ *
+ * The input signal is fed to all resonators in parallel, and their outputs
+ * are summed to produce a single output.
+ *
+ * Signal flow:
+ *
+ *              ┌─→ R1(F1, B1, A1) ─┐
+ *              ├─→ R2(F2, B2, A2) ─┤
+ *   Input ─────┼─→ R3(F3, B3, A3) ─┼─────→ Output (sum)
+ *              ├─→ ...            ─┤
+ *              └─→ Rn(Fn, Bn, An) ─┘
+ *
+ * The paper uses six formants (R1-R6):
+ *   - F1-F3: Primary formants that determine vowel identity
+ *   - F4-F6: Higher formants that contribute to voice timbre
+ *
+ * This implementation accepts any number of formants, allowing flexibility
+ * for experimentation or simplified models.
+ *
+ * Paper reference: Section 3.3.1 (parallel formant structure)
+ *
+ * Input parameters:
+ *   - formants: Array of {F, B, A} objects defining each resonator
+ *
+ * Note: The number of formants is fixed at creation time. Calling update()
+ * with a different array length will throw an error. To change the number
+ * of formants, destroy this node and create a new one.
+ */
+export declare class FormantBank implements Node<FormantBankParams> {
+    private resonators;
+    private inputGain;
+    private outputGain;
+    in: AudioNode;
+    out: AudioNode;
+    private constructor();
+    /**
+     * Creates a new FormantBank node with the specified formants.
+     *
+     * All formant resonators are created in parallel for efficiency.
+     */
+    static create(ctx: AudioContext, params: FormantBankParams): Promise<FormantBank>;
+    /**
+     * Updates all formant parameters.
+     *
+     * The formants array must have the same length as the original.
+     * Each resonator receives its corresponding parameters.
+     *
+     * @throws Error if the formants array length doesn't match
+     */
+    update(params: FormantBankParams): void;
+    /**
+     * Returns the number of formant resonators in this bank.
+     */
+    get count(): number;
+    /**
+     * Returns the array of formant resonators for direct AudioParam access.
+     *
+     * Example usage:
+     *   formantBank.formants[0].F.setValueAtTime(800, ctx.currentTime);
+     *   formantBank.formants[1].B.linearRampToValueAtTime(100, ctx.currentTime + 0.1);
+     */
+    get formants(): readonly FormantResonator[];
+    destroy(): void;
+}
+//# sourceMappingURL=formant-bank.d.ts.map

package/dist/nodes/formant-bank.d.ts.map

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package/dist/nodes/formant-resonator.d.ts

@@ -0,0 +1,67 @@
+import type { Node } from "./types";
+export type FormantResonatorParams = {
+    /** Formant centre frequency in Hz */
+    F: number;
+    /** Formant bandwidth in Hz */
+    B: number;
+    /** Formant amplitude (linear gain) */
+    A: number;
+};
+/**
+ * Formant Resonator (Ri)
+ *
+ * A single formant filter representing one resonance of the vocal tract.
+ * Implemented as a 2-pole 2-zero digital resonator (bandpass filter) with
+ * controllable centre frequency (F), bandwidth (B), and amplitude (A).
+ *
+ * The transfer function (Section 3.3.1) is:
+ *
+ * R(z) = A * (1 - R) * (1 - R*z^{-2}) / (1 - 2R*cos(θ)*z^{-1} + R²*z^{-2})
+ *
+ * Where:
+ *   R = e^(-π * B * Ts)  (pole radius, Ts = sampling period)
+ *   θ = 2π * F * Ts      (pole angle)
+ *
+ * The vocal tract model uses six formant resonators (R1-R6) in parallel:
+ *   - F1-F3: Primary formants that determine vowel identity
+ *   - F4-F6: Higher formants that contribute to voice timbre
+ *
+ * Grouping F3, F4, and F5 closely together can produce the "singer's formant"
+ * characteristic of trained classical voices.
+ *
+ * Paper reference: Section 3.3.1 (filter structure), Section 4.3.1 (vowel table),
+ *                  Section 4.3.6 (bandwidths), Section 4.3.7 (amplitudes)
+ *
+ * Input parameters:
+ *   - F (formant centre frequency in Hz) — interpolated from vowel table,
+ *     scaled by αS (vocal tract size) and K (larynx position factor)
+ *   - B (formant bandwidth in Hz) — interpolated from vowel table
+ *   - A (formant amplitude, linear) — interpolated from vowel table,
+ *     with correction when harmonics coincide with formant frequencies
+ */
+export declare class FormantResonator implements Node<FormantResonatorParams> {
+    private ctx;
+    private workletNode;
+    in: AudioNode;
+    out: AudioNode;
+    private constructor();
+    /** Formant centre frequency AudioParam (a-rate, 100-8000 Hz) */
+    get F(): AudioParam;
+    /** Formant bandwidth AudioParam (a-rate, 20-1000 Hz) */
+    get B(): AudioParam;
+    /** Formant amplitude AudioParam (a-rate, 0-10) */
+    get A(): AudioParam;
+    /**
+     * Creates a new FormantResonator node.
+     *
+     * The AudioWorklet module is registered automatically on first use.
+     */
+    static create(ctx: AudioContext, params: FormantResonatorParams): Promise<FormantResonator>;
+    /**
+     * Updates the formant resonator parameters.
+     * Sets AudioParams via setTargetAtTime for smooth transitions.
+     */
+    update(params: FormantResonatorParams): void;
+    destroy(): void;
+}
+//# sourceMappingURL=formant-resonator.d.ts.map

package/dist/nodes/formant-resonator.d.ts.map

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package/dist/nodes/gain.d.ts

@@ -0,0 +1,16 @@
+import type { Node } from "./types";
+export type GainParams = {
+    gain: number;
+};
+export declare class Gain implements Node<GainParams> {
+    private ctx;
+    private gainNode;
+    in: AudioNode;
+    out: AudioNode;
+    private constructor();
+    static create(ctx: AudioContext, params: GainParams): Promise<Gain>;
+    update(params: GainParams): void;
+    destroy(): void;
+    get gain(): AudioParam;
+}
+//# sourceMappingURL=gain.d.ts.map

package/dist/nodes/gain.d.ts.map

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package/dist/nodes/glottal-flow-derivative.d.ts

@@ -0,0 +1,119 @@
+import type { Node } from "./types";
+import { PulseTrain } from "./pulse-train";
+import { GlottalFormant } from "./glottal-formant";
+import { SpectralTilt } from "./spectral-tilt";
+import { NoiseSource } from "./noise-source";
+export type GlottalFlowDerivativeParams = {
+    /** Fundamental frequency in Hz (derived from P, P₀) */
+    f0: number;
+    /** Glottal formant centre frequency in Hz, computed as f0 / (2 * Oq) */
+    Fg: number;
+    /** Glottal formant bandwidth in Hz, computed from f0, Oq, and αm */
+    Bg: number;
+    /** Source amplitude, derived from E and Oq */
+    Ag: number;
+    /** First stage spectral tilt attenuation in dB at 3 kHz, derived from E and M */
+    Tl1: number;
+    /** Second stage spectral tilt attenuation in dB at 3 kHz, derived from E and M */
+    Tl2: number;
+    /** Noise amplitude, derived from B (breathiness) */
+    An: number;
+    /** Jitter depth: maximum f₀ perturbation as a fraction (0-0.3 for ±30%), derived from R */
+    jitterDepth: number;
+    /** Shimmer depth: maximum amplitude perturbation as a fraction (0-1 for ±100%), derived from R */
+    shimmerDepth: number;
+};
+/**
+ * Glottal Flow Derivative (GFD)
+ *
+ * The glottal flow derivative model represents the sound source produced by
+ * the vibrating vocal folds. It consists of a voiced component (periodic pulses
+ * shaped by the glottal formant and spectral tilt) and an unvoiced component
+ * (filtered noise for breathiness).
+ *
+ * The noise is modulated by the glottal flow derivative, so aspiration noise
+ * follows the glottal cycle - loudest during the open phase, quiet when closed.
+ *
+ * Signal flow:
+ *
+ *   PulseTrain(f0) → GlottalFormant(Fg, Bg, Ag) → SpectralTilt(Tl1, Tl2) ──┬────→ Output
+ *                                                                          │        ↑
+ *                                                                          │ (mod)  │
+ *                                                                          ↓        │
+ *   NoiseSource(An) ─────────────────────────────────────────────────→ [Multiply] ──┘
+ *
+ * Paper reference: Section 3.2 (overall structure)
+ *
+ * Input parameters:
+ *   - f0 (fundamental frequency) — controls the pulse train period
+ *   - Fg (glottal formant centre frequency) — computed as f0 / (2·Oq)
+ *   - Bg (glottal formant bandwidth) — computed from f0, Oq, and αm
+ *   - Ag (source amplitude) — derived from E, Oq, and R (shimmer)
+ *   - Tl1 (first tilt stage attenuation in dB at 3 kHz) — derived from E and M
+ *   - Tl2 (second tilt stage attenuation in dB at 3 kHz) — derived from E and M
+ *   - An (noise amplitude) — derived from B (breathiness)
+ */
+export declare class GlottalFlowDerivative implements Node<GlottalFlowDerivativeParams> {
+    private pulseTrain;
+    private glottalFormant;
+    private spectralTilt;
+    private noiseSource;
+    private noiseModulator;
+    private outputGain;
+    in: AudioNode | null;
+    out: AudioNode;
+    private constructor();
+    /**
+     * Creates a new GlottalFlowDerivative node.
+     *
+     * Sets up the voiced path (pulse train → glottal formant → spectral tilt)
+     * and the noise modulation path where noise is multiplied by the GFD signal.
+     */
+    static create(ctx: AudioContext, params: GlottalFlowDerivativeParams): Promise<GlottalFlowDerivative>;
+    /**
+     * Updates all glottal flow derivative parameters.
+     * Each sub-node receives its relevant parameters.
+     */
+    update(params: GlottalFlowDerivativeParams): void;
+    /**
+     * Returns the pulse train node for direct AudioParam access.
+     *
+     * Example usage:
+     *   gfd.pulseTrainNode.f0.setValueAtTime(440, ctx.currentTime);
+     *   gfd.pulseTrainNode.jitterDepth.linearRampToValueAtTime(0.1, ctx.currentTime + 0.5);
+     */
+    get pulseTrainNode(): PulseTrain;
+    /**
+     * Returns the glottal formant node for direct AudioParam access.
+     *
+     * Example usage:
+     *   gfd.glottalFormantNode.Fg.setValueAtTime(220, ctx.currentTime);
+     *   gfd.glottalFormantNode.Ag.setTargetAtTime(0.5, ctx.currentTime, 0.1);
+     */
+    get glottalFormantNode(): GlottalFormant;
+    /**
+     * Returns the spectral tilt node for direct AudioParam access.
+     *
+     * Example usage:
+     *   gfd.spectralTiltNode.Tl1.setValueAtTime(10, ctx.currentTime);
+     */
+    get spectralTiltNode(): SpectralTilt;
+    /**
+     * Returns the noise source node for direct AudioParam access.
+     *
+     * Example usage:
+     *   gfd.noiseSourceNode.An.setValueAtTime(0.3, ctx.currentTime);
+     */
+    get noiseSourceNode(): NoiseSource;
+    /**
+     * Starts the voiced excitation (pulse train).
+     */
+    start(): void;
+    /**
+     * Stops the voiced excitation (pulse train).
+     * Note: Noise source continues to produce output based on An parameter.
+     */
+    stop(): void;
+    destroy(): void;
+}
+//# sourceMappingURL=glottal-flow-derivative.d.ts.map

package/dist/nodes/glottal-flow-derivative.d.ts.map

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