audio-ui/lib/audio/effects/advanced.ts

/**
 * Advanced effects (Pitch Shifter, Time Stretcher, Distortion, Bitcrusher)
 */

import { getAudioContext } from '../context';

export interface PitchShifterParameters {
  semitones: number;   // -12 to +12 - pitch shift in semitones
  cents: number;       // -100 to +100 - fine tuning in cents
  mix: number;         // 0-1 - dry/wet mix
}

export interface TimeStretchParameters {
  rate: number;        // 0.5-2.0 - playback rate (0.5 = half speed, 2 = double speed)
  preservePitch: boolean; // whether to preserve pitch
  mix: number;         // 0-1 - dry/wet mix
}

export interface DistortionParameters {
  drive: number;       // 0-1 - amount of distortion
  tone: number;        // 0-1 - pre-distortion tone control
  output: number;      // 0-1 - output level
  type: 'soft' | 'hard' | 'tube'; // distortion type
  mix: number;         // 0-1 - dry/wet mix
}

export interface BitcrusherParameters {
  bitDepth: number;    // 1-16 - bit depth
  sampleRate: number;  // 100-48000 - sample rate reduction
  mix: number;         // 0-1 - dry/wet mix
}

/**
 * Apply pitch shifting to audio buffer
 * Uses simple time-domain pitch shifting (overlap-add)
 */
export async function applyPitchShift(
  buffer: AudioBuffer,
  params: PitchShifterParameters
): Promise<AudioBuffer> {
  const audioContext = getAudioContext();
  const channels = buffer.numberOfChannels;
  const sampleRate = buffer.sampleRate;

  // Calculate pitch shift ratio
  const totalCents = params.semitones * 100 + params.cents;
  const pitchRatio = Math.pow(2, totalCents / 1200);

  // For pitch shifting, we change the playback rate then resample
  const newLength = Math.floor(buffer.length / pitchRatio);
  const outputBuffer = audioContext.createBuffer(channels, newLength, sampleRate);

  // Simple linear interpolation resampling
  for (let channel = 0; channel < channels; channel++) {
    const inputData = buffer.getChannelData(channel);
    const outputData = outputBuffer.getChannelData(channel);

    for (let i = 0; i < newLength; i++) {
      const srcIndex = i * pitchRatio;
      const srcIndexInt = Math.floor(srcIndex);
      const srcIndexFrac = srcIndex - srcIndexInt;

      if (srcIndexInt < buffer.length - 1) {
        const sample1 = inputData[srcIndexInt];
        const sample2 = inputData[srcIndexInt + 1];
        const interpolated = sample1 + (sample2 - sample1) * srcIndexFrac;

        // Mix dry/wet
        const dry = i < buffer.length ? inputData[i] : 0;
        outputData[i] = dry * (1 - params.mix) + interpolated * params.mix;
      } else if (srcIndexInt < buffer.length) {
        const dry = i < buffer.length ? inputData[i] : 0;
        outputData[i] = dry * (1 - params.mix) + inputData[srcIndexInt] * params.mix;
      }
    }
  }

  return outputBuffer;
}

/**
 * Apply time stretching to audio buffer
 * Changes duration without affecting pitch (basic implementation)
 */
export async function applyTimeStretch(
  buffer: AudioBuffer,
  params: TimeStretchParameters
): Promise<AudioBuffer> {
  const audioContext = getAudioContext();
  const channels = buffer.numberOfChannels;
  const sampleRate = buffer.sampleRate;

  if (params.preservePitch) {
    // Time stretch with pitch preservation (overlap-add)
    const newLength = Math.floor(buffer.length / params.rate);
    const outputBuffer = audioContext.createBuffer(channels, newLength, sampleRate);

    const windowSize = 2048;
    const hopSize = Math.floor(windowSize / 4);

    for (let channel = 0; channel < channels; channel++) {
      const inputData = buffer.getChannelData(channel);
      const outputData = outputBuffer.getChannelData(channel);

      let readPos = 0;
      let writePos = 0;

      while (writePos < newLength) {
        // Simple overlap-add
        for (let i = 0; i < windowSize && writePos + i < newLength; i++) {
          const readIndex = Math.floor(readPos + i);
          if (readIndex < buffer.length) {
            // Hanning window
            const window = 0.5 * (1 - Math.cos((2 * Math.PI * i) / windowSize));
            outputData[writePos + i] += inputData[readIndex] * window;
          }
        }

        readPos += hopSize * params.rate;
        writePos += hopSize;
      }

      // Normalize
      let maxVal = 0;
      for (let i = 0; i < newLength; i++) {
        maxVal = Math.max(maxVal, Math.abs(outputData[i]));
      }
      if (maxVal > 0) {
        for (let i = 0; i < newLength; i++) {
          outputData[i] /= maxVal;
        }
      }
    }

    return outputBuffer;
  } else {
    // Simple speed change (changes pitch)
    const newLength = Math.floor(buffer.length / params.rate);
    const outputBuffer = audioContext.createBuffer(channels, newLength, sampleRate);

    for (let channel = 0; channel < channels; channel++) {
      const inputData = buffer.getChannelData(channel);
      const outputData = outputBuffer.getChannelData(channel);

      for (let i = 0; i < newLength; i++) {
        const srcIndex = i * params.rate;
        const srcIndexInt = Math.floor(srcIndex);
        const srcIndexFrac = srcIndex - srcIndexInt;

        if (srcIndexInt < buffer.length - 1) {
          const sample1 = inputData[srcIndexInt];
          const sample2 = inputData[srcIndexInt + 1];
          outputData[i] = sample1 + (sample2 - sample1) * srcIndexFrac;
        } else if (srcIndexInt < buffer.length) {
          outputData[i] = inputData[srcIndexInt];
        }
      }
    }

    return outputBuffer;
  }
}

/**
 * Apply distortion/overdrive effect
 */
export async function applyDistortion(
  buffer: AudioBuffer,
  params: DistortionParameters
): Promise<AudioBuffer> {
  const audioContext = getAudioContext();
  const channels = buffer.numberOfChannels;
  const length = buffer.length;
  const sampleRate = buffer.sampleRate;

  const outputBuffer = audioContext.createBuffer(channels, length, sampleRate);

  // Distortion function based on type
  const distort = (sample: number, drive: number, type: string): number => {
    const x = sample * (1 + drive * 10);

    switch (type) {
      case 'soft':
        // Soft clipping (tanh)
        return Math.tanh(x);

      case 'hard':
        // Hard clipping
        return Math.max(-1, Math.min(1, x));

      case 'tube':
        // Tube-like distortion (asymmetric)
        if (x > 0) {
          return 1 - Math.exp(-x);
        } else {
          return -1 + Math.exp(x);
        }

      default:
        return x;
    }
  };

  for (let channel = 0; channel < channels; channel++) {
    const inputData = buffer.getChannelData(channel);
    const outputData = outputBuffer.getChannelData(channel);

    // Simple low-pass filter for tone control
    let filterState = 0;
    const filterCutoff = params.tone;

    for (let i = 0; i < length; i++) {
      let sample = inputData[i];

      // Pre-distortion tone filter
      filterState = filterState * (1 - filterCutoff) + sample * filterCutoff;
      sample = filterState;

      // Apply distortion
      const distorted = distort(sample, params.drive, params.type);

      // Output level
      const processed = distorted * params.output;

      // Mix dry/wet
      outputData[i] = inputData[i] * (1 - params.mix) + processed * params.mix;
    }
  }

  return outputBuffer;
}

/**
 * Apply bitcrusher effect
 */
export async function applyBitcrusher(
  buffer: AudioBuffer,
  params: BitcrusherParameters
): Promise<AudioBuffer> {
  const audioContext = getAudioContext();
  const channels = buffer.numberOfChannels;
  const length = buffer.length;
  const sampleRate = buffer.sampleRate;

  const outputBuffer = audioContext.createBuffer(channels, length, sampleRate);

  // Calculate bit depth quantization step
  const bitLevels = Math.pow(2, params.bitDepth);
  const step = 2 / bitLevels;

  // Calculate sample rate reduction ratio
  const srRatio = sampleRate / params.sampleRate;

  for (let channel = 0; channel < channels; channel++) {
    const inputData = buffer.getChannelData(channel);
    const outputData = outputBuffer.getChannelData(channel);

    let holdSample = 0;
    let holdCounter = 0;

    for (let i = 0; i < length; i++) {
      // Sample rate reduction (sample and hold)
      if (holdCounter <= 0) {
        let sample = inputData[i];

        // Bit depth reduction
        sample = Math.floor(sample / step) * step;

        holdSample = sample;
        holdCounter = srRatio;
      }

      holdCounter--;

      // Mix dry/wet
      outputData[i] = inputData[i] * (1 - params.mix) + holdSample * params.mix;
    }
  }

  return outputBuffer;
}