Merge pull request #4 from entrepeneur4lyf/fix/export-44100-resample-dither

Fix 44.1k export resampling and add 16-bit TPDF dithering

Author: entrepeneur4lyf

web/app.js

@@ -40,6 +40,7 @@ import {
   // Renderer
   renderOffline,
   renderToAudioBuffer,
+  resampleAudioBuffer,
   // Waveform
   initWaveSurfer,
   destroyWaveSurfer,
@@ -1418,12 +1419,24 @@ async function processAudio() {
           throw new Error('Cancelled');
         }
 
-        updateProgress(85, 'Encoding WAV...');
+        let exportBuffer = result.audioBuffer;
+        if (exportBuffer.sampleRate !== parsedSampleRate) {
+          updateProgress(83, `Resampling to ${parsedSampleRate / 1000}kHz...`);
+        let exportBuffer = result.audioBuffer;
+        if (exportBuffer.sampleRate !== parsedSampleRate) {
+          updateProgress(86, `Resampling to ${parsedSampleRate / 1000}kHz...`);
+          exportBuffer = await resampleAudioBuffer(exportBuffer, parsedSampleRate);
+        }
+        if (processingCancelled) {
+          throw new Error('Cancelled');
+        }
+
+        updateProgress(88, 'Encoding WAV...');
         // Yield so the UI can repaint before encoding begins.
         await new Promise(resolve => setTimeout(resolve, 0));
 
-        outputData = await encodeWAVAsync(result.audioBuffer, parsedSampleRate, parsedBitDepth, {
-          onProgress: (p) => updateProgress(Math.round(85 + p * 10), 'Encoding WAV...'),
+        outputData = await encodeWAVAsync(exportBuffer, parsedSampleRate, parsedBitDepth, {
+          onProgress: (p) => updateProgress(Math.round(88 + p * 10), 'Encoding WAV...'),
           shouldCancel: () => processingCancelled
         });
       } catch (workerErr) {

web/ui/encoder.js

@@ -9,6 +9,20 @@ import { applyFinalFilters } from '../lib/dsp/final-filters.js';
 // WAV Encoding
 // ============================================================================
 
+const DITHER_BUFFER_SIZE = 256;
+const ditherBuffer = new Float32Array(DITHER_BUFFER_SIZE);
+for (let i = 0; i < DITHER_BUFFER_SIZE; i++) {
+  ditherBuffer[i] = (Math.random() + Math.random()) - 1;
+}
+let ditherIndex = 0;
+
+function triangularDither() {
+  // TPDF in integer (LSB) domain: [-1, 1]
+  const dither = ditherBuffer[ditherIndex];
+  ditherIndex = (ditherIndex + 1) % DITHER_BUFFER_SIZE;
+  return dither;
+}
+
 /**
  * Encode an AudioBuffer to WAV format (supports 16-bit and 24-bit)
  * @param {AudioBuffer} audioBuffer - Source audio buffer
@@ -18,8 +32,9 @@ import { applyFinalFilters } from '../lib/dsp/final-filters.js';
  */
 export function encodeWAV(audioBuffer, targetSampleRate, bitDepth) {
   const numChannels = audioBuffer.numberOfChannels;
+  const safeBitDepth = bitDepth === 24 ? 24 : 16;
   const sampleRate = targetSampleRate || audioBuffer.sampleRate;
-  const bytesPerSample = bitDepth / 8;
+  const bytesPerSample = safeBitDepth / 8;
 
   const channelData = [];
   for (let ch = 0; ch < numChannels; ch++) {
@@ -47,22 +62,27 @@ export function encodeWAV(audioBuffer, targetSampleRate, bitDepth) {
   view.setUint32(24, sampleRate, true);
   view.setUint32(28, sampleRate * numChannels * bytesPerSample, true);
   view.setUint16(32, numChannels * bytesPerSample, true);
-  view.setUint16(34, bitDepth, true);
+  view.setUint16(34, safeBitDepth, true);
   writeString(36, 'data');
   view.setUint32(40, dataSize, true);
 
   let offset = 44;
-  const maxVal = bitDepth === 16 ? 32767 : 8388607;
+  const maxVal = safeBitDepth === 16 ? 32767 : 8388607;
 
   for (let i = 0; i < numSamples; i++) {
     for (let ch = 0; ch < numChannels; ch++) {
       const sample = Math.max(-1, Math.min(1, channelData[ch][i]));
-      const intSample = Math.round(sample * maxVal);
-
-      if (bitDepth === 16) {
-        view.setInt16(offset, intSample, true);
+      const scaled = sample * maxVal;
+      // Dither is only needed for 16-bit export where quantization noise is audible.
+      const intSample = safeBitDepth === 16
+        ? Math.round(scaled + triangularDither())
+        : Math.round(scaled);
+
+      if (safeBitDepth === 16) {
+        const clampedSample = Math.max(-32768, Math.min(32767, intSample));
+        view.setInt16(offset, clampedSample, true);
         offset += 2;
-      } else if (bitDepth === 24) {
+      } else if (safeBitDepth === 24) {
         // Clamp to prevent overflow in bitwise operations
         const clampedSample = Math.max(-8388607, Math.min(8388607, intSample));
         view.setUint8(offset, clampedSample & 0xFF);
@@ -145,10 +165,15 @@ export async function encodeWAVAsync(audioBuffer, targetSampleRate, bitDepth, op
     for (let s = i; s < end; s++) {
       for (let ch = 0; ch < numChannels; ch++) {
         const sample = Math.max(-1, Math.min(1, channelData[ch][s]));
-        const intSample = Math.round(sample * maxVal);
+        const scaled = sample * maxVal;
+        // Dither is only needed for 16-bit export where quantization noise is audible.
+        const intSample = safeBitDepth === 16
+          ? Math.round(scaled + triangularDither())
+          : Math.round(scaled);
 
         if (safeBitDepth === 16) {
-          view.setInt16(offset, intSample, true);
+          const clampedSample = Math.max(-32768, Math.min(32767, intSample));
+          view.setInt16(offset, clampedSample, true);
           offset += 2;
         } else {
           const clampedSample = Math.max(-8388607, Math.min(8388607, intSample));

web/ui/renderer.js

@@ -162,6 +162,30 @@ function applyDSPChain(buffer, settings, onProgress = null, logPrefix = '[DSP]')
   return { buffer: renderedBuffer, measuredLufs };
 }
 
+/**
+ * Resample an AudioBuffer to a target sample rate.
+ * Uses OfflineAudioContext so output sample data and WAV header stay aligned.
+ * @param {AudioBuffer} sourceBuffer - Source audio buffer
+ * @param {number} targetSampleRate - Target sample rate in Hz
+ * @returns {Promise<AudioBuffer>} Resampled buffer (or original if unchanged)
+ */
+export async function resampleAudioBuffer(sourceBuffer, targetSampleRate) {
+  if (!sourceBuffer || !targetSampleRate || sourceBuffer.sampleRate === targetSampleRate) {
+    return sourceBuffer;
+  }
+
+  const numChannels = sourceBuffer.numberOfChannels;
+  const numSamples = Math.ceil(sourceBuffer.duration * targetSampleRate);
+  const offlineCtx = new OfflineAudioContext(numChannels, numSamples, targetSampleRate);
+  const source = offlineCtx.createBufferSource();
+
+  source.buffer = sourceBuffer;
+  source.connect(offlineCtx.destination);
+  source.start(0);
+
+  return offlineCtx.startRendering();
+}
+
 // ============================================================================
 // Offline Rendering (Export)
 // ============================================================================