@drop-ai/core 0.2.0 → 0.3.1

package/dist/audio/ExportAnalyzer.js

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+/**
+ * Export Analyzer — post-export analysis report.
+ *
+ * Computes: Peak, RMS, LUFS, True Peak, Dynamic Range, DC offset.
+ */
+import { measureLUFS } from './LufsNormalizer';
+import { measureTruePeakDbTP } from '../processing/TruePeakLimiter';
+/**
+ * Analyze an AudioBuffer and produce a comprehensive report.
+ */
+export function analyzeExport(buffer) {
+    const numChannels = buffer.numberOfChannels;
+    const length = buffer.length;
+    const sampleRate = buffer.sampleRate;
+    const peakLevels = [];
+    const rmsLevels = [];
+    const truePeakLevels = [];
+    const dcOffset = [];
+    let totalClipped = 0;
+    let hasClipping = false;
+    for (let ch = 0; ch < numChannels; ch++) {
+        const data = buffer.getChannelData(ch);
+        let peak = 0;
+        let sumSq = 0;
+        let sum = 0;
+        let clipped = 0;
+        for (let i = 0; i < length; i++) {
+            const abs = Math.abs(data[i]);
+            if (abs > peak)
+                peak = abs;
+            sumSq += data[i] * data[i];
+            sum += data[i];
+            if (abs >= 1.0) {
+                clipped++;
+            }
+        }
+        const rms = Math.sqrt(sumSq / length);
+        const dc = sum / length;
+        peakLevels.push(peak > 0 ? 20 * Math.log10(peak) : -Infinity);
+        rmsLevels.push(rms > 0 ? 20 * Math.log10(rms) : -Infinity);
+        truePeakLevels.push(measureTruePeakDbTP(data));
+        dcOffset.push(dc);
+        totalClipped += clipped;
+        if (clipped > 0)
+            hasClipping = true;
+    }
+    // LUFS measurement
+    const lufsResult = measureLUFS(buffer);
+    // Short-term max
+    const shortTermMax = lufsResult.shortTerm.length > 0
+        ? Math.max(...lufsResult.shortTerm.filter(l => isFinite(l)))
+        : -Infinity;
+    // Dynamic range (overall peak - overall RMS)
+    const overallPeak = Math.max(...peakLevels);
+    const overallRms = Math.max(...rmsLevels);
+    const dynamicRange = isFinite(overallPeak) && isFinite(overallRms)
+        ? overallPeak - overallRms
+        : 0;
+    return {
+        duration: length / sampleRate,
+        sampleRate,
+        channels: numChannels,
+        peakLevels,
+        rmsLevels,
+        truePeakLevels,
+        integratedLufs: lufsResult.integrated,
+        shortTermMax,
+        loudnessRange: lufsResult.range,
+        dynamicRange,
+        dcOffset,
+        clipping: hasClipping,
+        clippedSamples: totalClipped,
+    };
+}
+//# sourceMappingURL=ExportAnalyzer.js.map

package/dist/audio/ExportAnalyzer.js.map

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package/dist/audio/LufsNormalizer.d.ts

@@ -0,0 +1,36 @@
+/**
+ * LUFS Normalizer — ITU-R BS.1770-4 compliant gated loudness measurement.
+ *
+ * Implements:
+ *   1. K-weighting filter (high shelf + high pass)
+ *   2. 400ms block mean square computation
+ *   3. Absolute gate at -70 LUFS
+ *   4. Relative gate at -10 dB below ungated mean
+ *   5. Integrated loudness from gated blocks
+ */
+export interface LufsResult {
+    /** Integrated loudness in LUFS */
+    integrated: number;
+    /** Short-term loudness (3s window) values */
+    shortTerm: number[];
+    /** Momentary loudness (400ms window) values */
+    momentary: number[];
+    /** True peak in dBFS (per channel) */
+    truePeaks: number[];
+    /** Loudness range (LRA) in LU */
+    range: number;
+}
+/**
+ * Measure integrated LUFS of an AudioBuffer using the BS.1770-4 gated algorithm.
+ */
+export declare function measureLUFS(buffer: AudioBuffer): LufsResult;
+/**
+ * Normalize an AudioBuffer to a target LUFS level.
+ * Modifies the buffer in-place.
+ *
+ * @param buffer Audio buffer to normalize
+ * @param targetLufs Target integrated loudness (default: -14 LUFS)
+ * @returns The gain applied in dB, or 0 if silence
+ */
+export declare function normalizeLUFS(buffer: AudioBuffer, targetLufs?: number): number;
+//# sourceMappingURL=LufsNormalizer.d.ts.map

package/dist/audio/LufsNormalizer.d.ts.map

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package/dist/audio/LufsNormalizer.js

@@ -0,0 +1,216 @@
+/**
+ * LUFS Normalizer — ITU-R BS.1770-4 compliant gated loudness measurement.
+ *
+ * Implements:
+ *   1. K-weighting filter (high shelf + high pass)
+ *   2. 400ms block mean square computation
+ *   3. Absolute gate at -70 LUFS
+ *   4. Relative gate at -10 dB below ungated mean
+ *   5. Integrated loudness from gated blocks
+ */
+/**
+ * Pre-filter (high shelf ~+4 dB at ~1681 Hz) for K-weighting.
+ * ITU-R BS.1770-4 Table 1, Stage 1.
+ */
+function highShelfCoeffs(sampleRate) {
+    const f0 = 1681.974450955533;
+    const G = 3.999843853973347;
+    const Q = 0.7071752369554196;
+    const K = Math.tan((Math.PI * f0) / sampleRate);
+    const Vh = Math.pow(10, G / 20);
+    const Vb = Math.pow(Vh, 0.4996667741545416);
+    const a0 = 1 + K / Q + K * K;
+    return {
+        b0: (Vh + Vb * K / Q + K * K) / a0,
+        b1: (2 * (K * K - Vh)) / a0,
+        b2: (Vh - Vb * K / Q + K * K) / a0,
+        a1: (2 * (K * K - 1)) / a0,
+        a2: (1 - K / Q + K * K) / a0,
+    };
+}
+/**
+ * High-pass filter (~38 Hz) for K-weighting.
+ * ITU-R BS.1770-4 Table 1, Stage 2.
+ */
+function highPassCoeffs(sampleRate) {
+    const f0 = 38.13547087602444;
+    const Q = 0.5003270373238773;
+    const K = Math.tan((Math.PI * f0) / sampleRate);
+    const a0 = 1 + K / Q + K * K;
+    return {
+        b0: 1 / a0,
+        b1: -2 / a0,
+        b2: 1 / a0,
+        a1: (2 * (K * K - 1)) / a0,
+        a2: (1 - K / Q + K * K) / a0,
+    };
+}
+/**
+ * Apply a biquad filter to samples, returning a new array.
+ */
+function applyBiquad(samples, coeffs) {
+    const out = new Float32Array(samples.length);
+    let x1 = 0, x2 = 0, y1 = 0, y2 = 0;
+    for (let i = 0; i < samples.length; i++) {
+        const x0 = samples[i];
+        const y0 = coeffs.b0 * x0 + coeffs.b1 * x1 + coeffs.b2 * x2
+            - coeffs.a1 * y1 - coeffs.a2 * y2;
+        out[i] = y0;
+        x2 = x1;
+        x1 = x0;
+        y2 = y1;
+        y1 = y0;
+    }
+    return out;
+}
+/**
+ * Channel weight per ITU-R BS.1770-4:
+ * L, R, C = 1.0; Ls, Rs (surround) = 1.41 (~+1.5 dB)
+ * LFE = 0 (excluded)
+ */
+function getChannelWeight(channelIndex, totalChannels) {
+    if (totalChannels <= 2)
+        return 1.0;
+    // 5.1: L=0, R=1, C=2, LFE=3, Ls=4, Rs=5
+    if (totalChannels >= 6 && channelIndex === 3)
+        return 0; // LFE
+    if (totalChannels >= 6 && (channelIndex === 4 || channelIndex === 5))
+        return 1.41;
+    return 1.0;
+}
+/**
+ * Measure integrated LUFS of an AudioBuffer using the BS.1770-4 gated algorithm.
+ */
+export function measureLUFS(buffer) {
+    const sampleRate = buffer.sampleRate;
+    const numChannels = buffer.numberOfChannels;
+    // 400ms block size
+    const blockSize = Math.round(sampleRate * 0.4);
+    // 75% overlap → step = 100ms
+    const stepSize = Math.round(sampleRate * 0.1);
+    // Pre-compute K-weighting filter coefficients
+    const shelf = highShelfCoeffs(sampleRate);
+    const hp = highPassCoeffs(sampleRate);
+    // K-weight all channels
+    const kWeighted = [];
+    for (let ch = 0; ch < numChannels; ch++) {
+        const raw = buffer.getChannelData(ch);
+        const stage1 = applyBiquad(raw, shelf);
+        const stage2 = applyBiquad(stage1, hp);
+        kWeighted.push(stage2);
+    }
+    // Compute mean square per block, per channel, then sum with channel weights
+    const blockLoudnesses = [];
+    const numBlocks = Math.floor((buffer.length - blockSize) / stepSize) + 1;
+    for (let b = 0; b < numBlocks; b++) {
+        const start = b * stepSize;
+        const end = start + blockSize;
+        if (end > buffer.length)
+            break;
+        let sumWeightedMeanSquare = 0;
+        for (let ch = 0; ch < numChannels; ch++) {
+            const weight = getChannelWeight(ch, numChannels);
+            if (weight === 0)
+                continue;
+            const data = kWeighted[ch];
+            let sumSq = 0;
+            for (let i = start; i < end; i++) {
+                sumSq += data[i] * data[i];
+            }
+            sumWeightedMeanSquare += weight * (sumSq / blockSize);
+        }
+        // LUFS for this block
+        const blockLufs = sumWeightedMeanSquare > 0
+            ? -0.691 + 10 * Math.log10(sumWeightedMeanSquare)
+            : -Infinity;
+        blockLoudnesses.push(blockLufs);
+    }
+    // === Gated measurement ===
+    // Step 1: Absolute gate at -70 LUFS
+    const ABSOLUTE_GATE = -70;
+    const aboveAbsoluteGate = blockLoudnesses.filter(l => l > ABSOLUTE_GATE);
+    if (aboveAbsoluteGate.length === 0) {
+        return { integrated: -Infinity, shortTerm: [], momentary: blockLoudnesses, truePeaks: [], range: 0 };
+    }
+    // Step 2: Compute mean of blocks above absolute gate
+    const ungatedMeanPower = aboveAbsoluteGate.reduce((sum, l) => {
+        return sum + Math.pow(10, l / 10);
+    }, 0) / aboveAbsoluteGate.length;
+    const ungatedMeanLufs = -0.691 + 10 * Math.log10(ungatedMeanPower);
+    // Step 3: Relative gate at -10 dB below ungated mean
+    const relativeGate = ungatedMeanLufs - 10;
+    // Step 4: Compute mean of blocks above both gates
+    const aboveBothGates = blockLoudnesses.filter(l => l > ABSOLUTE_GATE && l > relativeGate);
+    let integrated = -Infinity;
+    if (aboveBothGates.length > 0) {
+        const gatedMeanPower = aboveBothGates.reduce((sum, l) => {
+            return sum + Math.pow(10, l / 10);
+        }, 0) / aboveBothGates.length;
+        integrated = -0.691 + 10 * Math.log10(gatedMeanPower);
+    }
+    // Short-term loudness (3s window)
+    const shortTermBlockSize = Math.round(sampleRate * 3);
+    const shortTermStep = Math.round(sampleRate * 1);
+    const shortTerm = [];
+    for (let start = 0; start + shortTermBlockSize <= buffer.length; start += shortTermStep) {
+        let sumWeightedMS = 0;
+        for (let ch = 0; ch < numChannels; ch++) {
+            const weight = getChannelWeight(ch, numChannels);
+            if (weight === 0)
+                continue;
+            const data = kWeighted[ch];
+            let sumSq = 0;
+            for (let i = start; i < start + shortTermBlockSize; i++) {
+                sumSq += data[i] * data[i];
+            }
+            sumWeightedMS += weight * (sumSq / shortTermBlockSize);
+        }
+        shortTerm.push(sumWeightedMS > 0 ? -0.691 + 10 * Math.log10(sumWeightedMS) : -Infinity);
+    }
+    // True peak detection (simple — for full spec, 4x oversample in TruePeakLimiter)
+    const truePeaks = [];
+    for (let ch = 0; ch < numChannels; ch++) {
+        const data = buffer.getChannelData(ch);
+        let peak = 0;
+        for (let i = 0; i < data.length; i++) {
+            const abs = Math.abs(data[i]);
+            if (abs > peak)
+                peak = abs;
+        }
+        truePeaks.push(peak > 0 ? 20 * Math.log10(peak) : -Infinity);
+    }
+    // LRA (Loudness Range) — simplified: difference between 10th and 95th percentile of short-term
+    let range = 0;
+    const validShortTerm = shortTerm.filter(l => l > ABSOLUTE_GATE && l > relativeGate).sort((a, b) => a - b);
+    if (validShortTerm.length >= 2) {
+        const p10 = validShortTerm[Math.floor(validShortTerm.length * 0.1)];
+        const p95 = validShortTerm[Math.floor(validShortTerm.length * 0.95)];
+        range = p95 - p10;
+    }
+    return { integrated, shortTerm, momentary: blockLoudnesses, truePeaks, range };
+}
+/**
+ * Normalize an AudioBuffer to a target LUFS level.
+ * Modifies the buffer in-place.
+ *
+ * @param buffer Audio buffer to normalize
+ * @param targetLufs Target integrated loudness (default: -14 LUFS)
+ * @returns The gain applied in dB, or 0 if silence
+ */
+export function normalizeLUFS(buffer, targetLufs = -14) {
+    const result = measureLUFS(buffer);
+    if (result.integrated === -Infinity || !isFinite(result.integrated)) {
+        return 0; // silence — no gain to apply
+    }
+    const gainDb = targetLufs - result.integrated;
+    const gainLinear = Math.pow(10, gainDb / 20);
+    // Apply gain to all channels
+    for (let ch = 0; ch < buffer.numberOfChannels; ch++) {
+        const data = buffer.getChannelData(ch);
+        for (let i = 0; i < data.length; i++) {
+            data[i] *= gainLinear;
+        }
+    }
+    return gainDb;
+}
+//# sourceMappingURL=LufsNormalizer.js.map

package/dist/audio/LufsNormalizer.js.map

@@ -0,0 +1 @@
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package/dist/audio/MeterUtils.d.ts

@@ -1,8 +1,6 @@
 /**
  * Utility functions for audio metering.
  * Operates on raw Float32Array time-domain data from AnalyserNode.
- *
- * Ardour reference: libs/ardour/meter.cc
  */
 /**
  * Convert a linear amplitude value to decibels (dBFS).

package/dist/audio/MeterUtils.d.ts.map

@@ -1 +1 @@
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package/dist/audio/MeterUtils.js

@@ -1,8 +1,6 @@
 /**
  * Utility functions for audio metering.
  * Operates on raw Float32Array time-domain data from AnalyserNode.
- *
- * Ardour reference: libs/ardour/meter.cc
  */
 /**
  * Convert a linear amplitude value to decibels (dBFS).

package/dist/audio/MeterUtils.js.map

@@ -1 +1 @@
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package/dist/audio/OfflineExporter.d.ts

@@ -3,7 +3,6 @@ import { ExportConfig } from '../domain/ExportConfig';
 import { ExportStatus } from '../domain/ExportStatus';
 /**
  * Offline Audio Exporter
- * Ardour 참조: libs/ardour/export_handler.cc
  *
  * Web Audio의 OfflineAudioContext를 활용하여 실시간보다 빠르게 렌더링합니다.
  */
@@ -19,13 +18,8 @@ export declare class OfflineExporter {
     /**
      * Normalize audio buffer using peak normalization.
      *
-     * NOTE: Despite previously being labeled as "LUFS", this method performs simple
-     * peak normalization (not ITU-R BS.1770 loudness measurement). The target value
-     * is interpreted as a peak level in dBFS, not as an integrated loudness value.
-     * True LUFS normalization would require K-weighted filtering and gated measurement.
-     *
      * @param buffer Audio data (modified in-place)
-     * @param targetPeakDb Target peak level in dBFS (default: -14 dBFS)
+     * @param targetPeakDb Target peak level in dBFS (default: -1 dBFS)
      */
     private static normalizeBuffer;
     /**

package/dist/audio/OfflineExporter.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"OfflineExporter.d.ts","sourceRoot":"","sources":["../../src/audio/OfflineExporter.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,UAAU,EAAE,OAAO,EAAE,MAAM,iBAAiB,CAAC;AACtD,OAAO,EAAE,YAAY,EAAgB,MAAM,wBAAwB,CAAC;AACpE,OAAO,EAAE,YAAY,EAAkB,MAAM,wBAAwB,CAAC;AAStE;;;;GAIG;AACH,qBAAa,eAAe;IACxB;;;;;;OAMG;WACiB,MAAM,CACtB,MAAM,EAAE,YAAY,EACpB,MAAM,EAAE,YAAY,EACpB,aAAa,EAAE,CAAC,QAAQ,EAAE,OAAO,EAAE,EAAE,UAAU,EAAE,UAAU,EAAE,QAAQ,EAAE,UAAU,KAAK,OAAO,CAAC,WAAW,CAAC,GAC3G,OAAO,CAAC,IAAI,CAAC;IAyFhB;;;;;OAKG;mBACkB,eAAe;IA0BpC;;OAEG;mBACkB,YAAY;IAkBjC;;OAEG;IACH,OAAO,CAAC,MAAM,CAAC,SAAS;IA0CxB;;;OAGG;WACiB,WAAW,CAC3B,MAAM,EAAE,YAAY,EACpB,MAAM,EAAE,YAAY,EACpB,QAAQ,EAAE,OAAO,EAAE,EACnB,UAAU,EAAE,GAAG,CAAC,OAAO,EAAE,MAAM,CAAC,EAChC,aAAa,EAAE,CAAC,GAAG,EAAE,OAAO,EAAE,EAAE,UAAU,EAAE,UAAU,EAAE,QAAQ,EAAE,UAAU,KAAK,OAAO,CAAC,WAAW,CAAC,GACtG,OAAO,CAAC,GAAG,CAAC,OAAO,EAAE;QAAE,IAAI,EAAE,IAAI,CAAC;QAAC,QAAQ,EAAE,MAAM,CAAA;KAAE,CAAC,CAAC;CA8D7D"}

package/dist/audio/OfflineExporter.js

@@ -3,10 +3,11 @@ import { AudioBufferToWav } from '../utils/AudioBufferToWav';
 import { OggEncoder } from '../utils/OggEncoder';
 import { FlacEncoder } from '../utils/FlacEncoder';
 import { DitherProcessor, DitherType } from '../utils/DitherProcessor';
+import { normalizeLUFS } from './LufsNormalizer';
+import { applyTruePeakLimiter } from '../processing/TruePeakLimiter';
 import * as lamejs from 'lamejs';
 /**
  * Offline Audio Exporter
- * Ardour 참조: libs/ardour/export_handler.cc
  *
  * Web Audio의 OfflineAudioContext를 활용하여 실시간보다 빠르게 렌더링합니다.
  */
@@ -42,7 +43,22 @@ export class OfflineExporter {
             // Normalize Phase (if enabled)
             if (config.normalize) {
                 status.setProgress(ExportProgress.NORMALIZING);
-                await this.normalizeBuffer(renderedBuffer, config.targetPeakDb);
+                if (config.normalizeMode === 'lufs') {
+                    // LUFS normalization (ITU-R BS.1770-4)
+                    const gainDb = normalizeLUFS(renderedBuffer, config.targetLufs);
+                    console.log(`[OfflineExporter] LUFS normalization applied: ${gainDb.toFixed(2)} dB gain`);
+                }
+                else {
+                    // Peak normalization (legacy)
+                    await this.normalizeBuffer(renderedBuffer, config.targetPeakDb);
+                }
+            }
+            // True Peak Limiter (after normalization, before encoding)
+            if (config.truePeakLimit) {
+                const limited = applyTruePeakLimiter(renderedBuffer, config.truePeakCeiling);
+                if (limited) {
+                    console.log(`[OfflineExporter] True peak limiter applied (ceiling: ${config.truePeakCeiling} dBTP)`);
+                }
             }
             // Dithering Phase (before encoding, when reducing bit depth)
             if (config.ditherType !== DitherType.NONE) {
@@ -72,16 +88,11 @@ export class OfflineExporter {
     /**
      * Normalize audio buffer using peak normalization.
      *
-     * NOTE: Despite previously being labeled as "LUFS", this method performs simple
-     * peak normalization (not ITU-R BS.1770 loudness measurement). The target value
-     * is interpreted as a peak level in dBFS, not as an integrated loudness value.
-     * True LUFS normalization would require K-weighted filtering and gated measurement.
-     *
      * @param buffer Audio data (modified in-place)
-     * @param targetPeakDb Target peak level in dBFS (default: -14 dBFS)
+     * @param targetPeakDb Target peak level in dBFS (default: -1 dBFS)
      */
     static async normalizeBuffer(buffer, targetPeakDb) {
-        const target = targetPeakDb !== null && targetPeakDb !== void 0 ? targetPeakDb : -14; // Default to -14 dBFS
+        const target = targetPeakDb !== null && targetPeakDb !== void 0 ? targetPeakDb : -1;
         let maxPeak = 0;
         for (let ch = 0; ch < buffer.numberOfChannels; ch++) {
             const data = buffer.getChannelData(ch);
@@ -113,7 +124,7 @@ export class OfflineExporter {
             case 'mp3':
                 return this.encodeMP3(buffer, config);
             case 'ogg':
-                return OggEncoder.encode(buffer, (_a = config.quality) !== null && _a !== void 0 ? _a : 0.5);
+                return await OggEncoder.encode(buffer, (_a = config.quality) !== null && _a !== void 0 ? _a : 0.5);
             case 'flac':
                 return FlacEncoder.encode(buffer, config.sampleFormat === 'int24' ? 24 : 16);
             default:
@@ -180,7 +191,16 @@ export class OfflineExporter {
             const renderedBuffer = await getTrackAudio([trackId], config.startFrame, config.endFrame);
             // Normalize if enabled
             if (config.normalize) {
-                await this.normalizeBuffer(renderedBuffer, config.targetPeakDb);
+                if (config.normalizeMode === 'lufs') {
+                    normalizeLUFS(renderedBuffer, config.targetLufs);
+                }
+                else {
+                    await this.normalizeBuffer(renderedBuffer, config.targetPeakDb);
+                }
+            }
+            // True Peak Limiter
+            if (config.truePeakLimit) {
+                applyTruePeakLimiter(renderedBuffer, config.truePeakCeiling);
             }
             // Encode
             const blob = await this.encodeToBlob(renderedBuffer, config);

package/dist/audio/OfflineExporter.js.map

@@ -1 +1 @@
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package/dist/audio/ProcessingGraph.d.ts

@@ -2,7 +2,6 @@
  * ProcessingGraph — a directed acyclic graph (DAG) for modeling signal-processing
  * node execution order.
  *
- * Inspired by Ardour's internal graph scheduler (`libs/ardour/graph.cc`).
  * The graph is a standalone utility with no audio-framework dependencies,
  * making it suitable for unit testing and headless use.
  *

package/dist/audio/ProcessingGraph.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"ProcessingGraph.d.ts","sourceRoot":"","sources":["../../src/audio/ProcessingGraph.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;;GAeG;AAEH;;;;;GAKG;AACH,MAAM,WAAW,SAAS;IACtB,uCAAuC;IACvC,EAAE,EAAE,MAAM,CAAC;IACX,mEAAmE;IACnE,OAAO,CAAC,IAAI,IAAI,CAAC;CACpB;AAED;;;;;;;;;;;;;;;;;;;;;GAqBG;AACH,qBAAa,eAAe;IACxB,wCAAwC;IACxC,OAAO,CAAC,KAAK,CAAqC;IAElD;;;OAGG;IACH,OAAO,CAAC,KAAK,CAAuC;IAMpD;;;;;;;OAOG;IACH,OAAO,CAAC,IAAI,EAAE,SAAS,GAAG,IAAI;IAO9B;;;;;OAKG;IACH,UAAU,CAAC,MAAM,EAAE,MAAM,GAAG,IAAI;IAShC;;;;OAIG;IACH,OAAO,CAAC,MAAM,EAAE,MAAM,GAAG,SAAS,GAAG,SAAS;IAI9C;;OAEG;IACH,IAAI,SAAS,IAAI,MAAM,CAEtB;IAMD;;;;;;;;OAQG;IACH,OAAO,CAAC,IAAI,EAAE,MAAM,EAAE,EAAE,EAAE,MAAM,GAAG,IAAI;IAUvC;;;;;OAKG;IACH,UAAU,CAAC,IAAI,EAAE,MAAM,EAAE,EAAE,EAAE,MAAM,GAAG,IAAI;IAW1C;;;;;;;OAOG;IACH,eAAe,IAAI,SAAS,EAAE;IAoE9B;;;;;;;;;;;OAWG;IACH,iBAAiB,IAAI,SAAS,EAAE,EAAE;IA2ElC;;;;OAIG;IACH,YAAY,IAAI,OAAO;IA6DvB;;;;;;;OAOG;IACH,eAAe,CAAC,MAAM,EAAE,MAAM,GAAG,MAAM,EAAE;IAKzC;;;;;;OAMG;IACH,aAAa,CAAC,MAAM,EAAE,MAAM,GAAG,MAAM,EAAE;IAcvC;;OAEG;IACH,KAAK,IAAI,IAAI;CAIhB"}
+{"version":3,"file":"ProcessingGraph.d.ts","sourceRoot":"","sources":["../../src/audio/ProcessingGraph.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;GAcG;AAEH;;;;;GAKG;AACH,MAAM,WAAW,SAAS;IACtB,uCAAuC;IACvC,EAAE,EAAE,MAAM,CAAC;IACX,mEAAmE;IACnE,OAAO,CAAC,IAAI,IAAI,CAAC;CACpB;AAED;;;;;;;;;;;;;;;;;;;;;GAqBG;AACH,qBAAa,eAAe;IACxB,wCAAwC;IACxC,OAAO,CAAC,KAAK,CAAqC;IAElD;;;OAGG;IACH,OAAO,CAAC,KAAK,CAAuC;IAMpD;;;;;;;OAOG;IACH,OAAO,CAAC,IAAI,EAAE,SAAS,GAAG,IAAI;IAO9B;;;;;OAKG;IACH,UAAU,CAAC,MAAM,EAAE,MAAM,GAAG,IAAI;IAShC;;;;OAIG;IACH,OAAO,CAAC,MAAM,EAAE,MAAM,GAAG,SAAS,GAAG,SAAS;IAI9C;;OAEG;IACH,IAAI,SAAS,IAAI,MAAM,CAEtB;IAMD;;;;;;;;OAQG;IACH,OAAO,CAAC,IAAI,EAAE,MAAM,EAAE,EAAE,EAAE,MAAM,GAAG,IAAI;IAUvC;;;;;OAKG;IACH,UAAU,CAAC,IAAI,EAAE,MAAM,EAAE,EAAE,EAAE,MAAM,GAAG,IAAI;IAW1C;;;;;;;OAOG;IACH,eAAe,IAAI,SAAS,EAAE;IAoE9B;;;;;;;;;;;OAWG;IACH,iBAAiB,IAAI,SAAS,EAAE,EAAE;IA2ElC;;;;OAIG;IACH,YAAY,IAAI,OAAO;IA6DvB;;;;;;;OAOG;IACH,eAAe,CAAC,MAAM,EAAE,MAAM,GAAG,MAAM,EAAE;IAKzC;;;;;;OAMG;IACH,aAAa,CAAC,MAAM,EAAE,MAAM,GAAG,MAAM,EAAE;IAcvC;;OAEG;IACH,KAAK,IAAI,IAAI;CAIhB"}

package/dist/audio/ProcessingGraph.js

@@ -2,7 +2,6 @@
  * ProcessingGraph — a directed acyclic graph (DAG) for modeling signal-processing
  * node execution order.
  *
- * Inspired by Ardour's internal graph scheduler (`libs/ardour/graph.cc`).
  * The graph is a standalone utility with no audio-framework dependencies,
  * making it suitable for unit testing and headless use.
  *

package/dist/audio/ProcessingGraph.js.map

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