feat: build reproducible dataset editor

tests/unit/acoustics.test.js

@@ -6,9 +6,11 @@ import {
   prepareDelaySpectrum,
 } from "/packages/acoustics-core/fractional-delay.js";
 import { dbToLinear, distance3 } from "/packages/acoustics-core/math.js";
+import { renderSceneMix } from "/packages/acoustics-core/mixdown.js";
 import { whiteNoiseAt } from "/packages/acoustics-core/noise.js";
 import { sphericalAttenuation, synthesizeMicrophoneSignals } from "/packages/acoustics-core/propagation.js";
 import { getFrameCount } from "/packages/acoustics-core/stft.js";
+import { encodeMonoPcm16Wav } from "/packages/acoustics-core/wav.js";
 
 function assertNear(actual, expected, tolerance, label) {
   assert(Math.abs(actual - expected) <= tolerance, `${label}: expected ${expected} ± ${tolerance}, received ${actual}`);
@@ -302,3 +304,68 @@ test("CPU DAS가 event-loop 취소와 working-set 한도를 지킨다", async ()
   assert(stftStarted, "STFT started before abort");
   assert(abortError?.name === "AbortError", "async abort");
 });
+
+test("scene mix가 peak 정규화와 clipping 통계를 정확히 기록한다", () => {
+  const assetBuffers = new Map([
+    ["levels", { sampleRate: 4, channels: [Float32Array.of(2, -1.5, 1, -0.5)] }],
+  ]);
+  const scene = {
+    sampleRate: 4,
+    duration: 1,
+    seed: 1,
+    mix: { normalization: "peak", targetPeak: 0.5 },
+    sources: [{
+      id: "levels",
+      gainDb: 0,
+      startTime: 0,
+      duration: 1,
+      signal: { type: "asset", assetId: "levels" },
+    }],
+    ambient: [],
+  };
+
+  const normalized = renderSceneMix(scene, assetBuffers);
+  assertNear(normalized.preNormalizationPeak, 2, 1e-12, "pre-normalization peak");
+  assertNear(normalized.appliedScale, 0.25, 1e-12, "normalization scale");
+  assertNear(normalized.peak, 0.5, 1e-12, "normalized peak");
+  assertEqual(normalized.clippedSampleCount, 0, "normalized clipping count");
+  [0.5, -0.375, 0.25, -0.125].forEach((expected, index) => {
+    assertNear(normalized.samples[index], expected, 1e-12, `normalized sample ${index}`);
+  });
+
+  const unnormalized = renderSceneMix({ ...scene, mix: { normalization: "none", targetPeak: 0.5 } }, assetBuffers);
+  assertNear(unnormalized.preNormalizationPeak, 2, 1e-12, "unnormalized input peak");
+  assertNear(unnormalized.appliedScale, 1, 1e-12, "unnormalized scale");
+  assertNear(unnormalized.peak, 2, 1e-12, "unnormalized output peak");
+  assertEqual(unnormalized.clippedSampleCount, 2, "unnormalized clipping count");
+});
+
+test("PCM16 WAV가 표준 mono header를 쓰고 범위 밖 sample을 clamp한다", async () => {
+  const blob = encodeMonoPcm16Wav(Float32Array.of(-2, -1, -0.5, 0, 0.5, 1, 2), 8000);
+  const buffer = await blob.arrayBuffer();
+  const bytes = new Uint8Array(buffer);
+  const view = new DataView(buffer);
+  const ascii = (offset, length) => String.fromCharCode(...bytes.subarray(offset, offset + length));
+
+  assertEqual(blob.type, "audio/wav", "WAV MIME type");
+  assertEqual(buffer.byteLength, 58, "WAV byte length");
+  assertEqual(ascii(0, 4), "RIFF", "RIFF id");
+  assertEqual(view.getUint32(4, true), 50, "RIFF chunk size");
+  assertEqual(ascii(8, 4), "WAVE", "WAVE id");
+  assertEqual(ascii(12, 4), "fmt ", "fmt id");
+  assertEqual(view.getUint32(16, true), 16, "PCM fmt chunk size");
+  assertEqual(view.getUint16(20, true), 1, "PCM format");
+  assertEqual(view.getUint16(22, true), 1, "mono channel count");
+  assertEqual(view.getUint32(24, true), 8000, "sample rate");
+  assertEqual(view.getUint32(28, true), 16000, "byte rate");
+  assertEqual(view.getUint16(32, true), 2, "block align");
+  assertEqual(view.getUint16(34, true), 16, "bits per sample");
+  assertEqual(ascii(36, 4), "data", "data id");
+  assertEqual(view.getUint32(40, true), 14, "PCM data byte length");
+
+  const actualSamples = Array.from({ length: 7 }, (_, index) => view.getInt16(44 + index * 2, true));
+  const expectedSamples = [-32768, -32768, -16384, 0, 16383, 32767, 32767];
+  expectedSamples.forEach((expected, index) => {
+    assertEqual(actualSamples[index], expected, `PCM sample ${index}`);
+  });
+});