@drop-ai/core 0.2.0 → 0.3.0

package/dist/audio/SampleRateConverter.js

@@ -0,0 +1,84 @@
+/**
+ * Sample Rate Converter
+ * Ardour reference: libs/ardour/resampled_source.cc
+ *
+ * Uses OfflineAudioContext for high-quality resampling.
+ * Falls back to linear interpolation in non-browser environments.
+ */
+/**
+ * Resample an AudioBuffer to a target sample rate.
+ * Uses OfflineAudioContext (native resampler) when available.
+ *
+ * @param buffer Source audio buffer
+ * @param targetSampleRate Desired sample rate
+ * @returns Resampled AudioBuffer
+ */
+export async function resampleAudio(buffer, targetSampleRate) {
+    if (buffer.sampleRate === targetSampleRate) {
+        return buffer; // No conversion needed
+    }
+    // Use OfflineAudioContext for native high-quality resampling
+    if (typeof OfflineAudioContext !== 'undefined') {
+        return resampleWithOfflineContext(buffer, targetSampleRate);
+    }
+    // Fallback: linear interpolation
+    return resampleLinear(buffer, targetSampleRate);
+}
+/**
+ * Native resampling using OfflineAudioContext.
+ */
+async function resampleWithOfflineContext(buffer, targetSampleRate) {
+    const ratio = targetSampleRate / buffer.sampleRate;
+    const newLength = Math.ceil(buffer.length * ratio);
+    const offlineCtx = new OfflineAudioContext(buffer.numberOfChannels, newLength, targetSampleRate);
+    const source = offlineCtx.createBufferSource();
+    source.buffer = buffer;
+    source.connect(offlineCtx.destination);
+    source.start(0);
+    return await offlineCtx.startRendering();
+}
+/**
+ * Linear interpolation resampling (fallback for non-browser environments).
+ */
+function resampleLinear(buffer, targetSampleRate) {
+    const ratio = buffer.sampleRate / targetSampleRate;
+    const newLength = Math.ceil(buffer.length / ratio);
+    const numChannels = buffer.numberOfChannels;
+    // Create output channel data
+    const outputChannels = [];
+    for (let ch = 0; ch < numChannels; ch++) {
+        const input = buffer.getChannelData(ch);
+        const output = new Float32Array(newLength);
+        for (let i = 0; i < newLength; i++) {
+            const srcPos = i * ratio;
+            const srcIdx = Math.floor(srcPos);
+            const frac = srcPos - srcIdx;
+            if (srcIdx + 1 < input.length) {
+                output[i] = input[srcIdx] * (1 - frac) + input[srcIdx + 1] * frac;
+            }
+            else if (srcIdx < input.length) {
+                output[i] = input[srcIdx];
+            }
+        }
+        outputChannels.push(output);
+    }
+    // Return as a mock AudioBuffer-like object (for non-browser env)
+    return {
+        numberOfChannels: numChannels,
+        length: newLength,
+        sampleRate: targetSampleRate,
+        duration: newLength / targetSampleRate,
+        getChannelData(channel) {
+            return outputChannels[channel];
+        },
+        copyFromChannel() { },
+        copyToChannel() { },
+    };
+}
+/**
+ * Check if sample rate conversion is needed.
+ */
+export function needsResampling(sourceSampleRate, sessionSampleRate) {
+    return sourceSampleRate !== sessionSampleRate;
+}
+//# sourceMappingURL=SampleRateConverter.js.map

package/dist/audio/SampleRateConverter.js.map

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

@@ -0,0 +1,29 @@
+/**
+ * Silence Padding & Trimming
+ * Ardour reference: libs/ardour/export_format_specification.h
+ *   (silence_beginning, silence_end fields)
+ *
+ * Adds or removes silence at the start/end of audio buffers.
+ */
+/**
+ * Add silence padding to start and/or end of audio data.
+ *
+ * @param channelData Array of channel data (modified in place via return)
+ * @param startFrames Number of silence frames to prepend
+ * @param endFrames Number of silence frames to append
+ * @returns New channel data arrays with padding
+ */
+export declare function addSilencePadding(channelData: Float32Array[], startFrames: number, endFrames: number): Float32Array[];
+/**
+ * Trim silence from start and/or end of audio data.
+ *
+ * @param channelData Array of channel data
+ * @param threshold Amplitude threshold below which is considered silence (default: 0.001)
+ * @returns Trimmed channel data and the number of frames trimmed from each end
+ */
+export declare function trimSilence(channelData: Float32Array[], threshold?: number): {
+    data: Float32Array[];
+    trimmedStart: number;
+    trimmedEnd: number;
+};
+//# sourceMappingURL=SilencePadding.d.ts.map

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

@@ -0,0 +1 @@
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package/dist/audio/SilencePadding.js

@@ -0,0 +1,86 @@
+/**
+ * Silence Padding & Trimming
+ * Ardour reference: libs/ardour/export_format_specification.h
+ *   (silence_beginning, silence_end fields)
+ *
+ * Adds or removes silence at the start/end of audio buffers.
+ */
+/**
+ * Add silence padding to start and/or end of audio data.
+ *
+ * @param channelData Array of channel data (modified in place via return)
+ * @param startFrames Number of silence frames to prepend
+ * @param endFrames Number of silence frames to append
+ * @returns New channel data arrays with padding
+ */
+export function addSilencePadding(channelData, startFrames, endFrames) {
+    if (startFrames <= 0 && endFrames <= 0) {
+        return channelData;
+    }
+    const originalLength = channelData[0].length;
+    const newLength = startFrames + originalLength + endFrames;
+    return channelData.map(data => {
+        const result = new Float32Array(newLength);
+        // Copy original data after start padding
+        result.set(data, startFrames);
+        return result;
+    });
+}
+/**
+ * Trim silence from start and/or end of audio data.
+ *
+ * @param channelData Array of channel data
+ * @param threshold Amplitude threshold below which is considered silence (default: 0.001)
+ * @returns Trimmed channel data and the number of frames trimmed from each end
+ */
+export function trimSilence(channelData, threshold = 0.001) {
+    if (channelData.length === 0 || channelData[0].length === 0) {
+        return { data: channelData, trimmedStart: 0, trimmedEnd: 0 };
+    }
+    const length = channelData[0].length;
+    // Find first non-silent sample (across all channels)
+    let startIdx = 0;
+    for (let i = 0; i < length; i++) {
+        let isSilent = true;
+        for (const data of channelData) {
+            if (Math.abs(data[i]) > threshold) {
+                isSilent = false;
+                break;
+            }
+        }
+        if (!isSilent) {
+            startIdx = i;
+            break;
+        }
+        if (i === length - 1) {
+            // Entire buffer is silence
+            return {
+                data: channelData.map(() => new Float32Array(0)),
+                trimmedStart: length,
+                trimmedEnd: 0,
+            };
+        }
+    }
+    // Find last non-silent sample
+    let endIdx = length - 1;
+    for (let i = length - 1; i >= startIdx; i--) {
+        let isSilent = true;
+        for (const data of channelData) {
+            if (Math.abs(data[i]) > threshold) {
+                isSilent = false;
+                break;
+            }
+        }
+        if (!isSilent) {
+            endIdx = i;
+            break;
+        }
+    }
+    const trimmedLength = endIdx - startIdx + 1;
+    return {
+        data: channelData.map(data => data.slice(startIdx, endIdx + 1)),
+        trimmedStart: startIdx,
+        trimmedEnd: length - endIdx - 1,
+    };
+}
+//# sourceMappingURL=SilencePadding.js.map

package/dist/audio/SilencePadding.js.map

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

@@ -10,18 +10,10 @@
  * configurable memory limit and evicting least-recently-used entries when
  * the limit is exceeded.
  *
- * This class has **no** external dependencies (no audio framework imports)
- * and can be unit-tested in isolation.
- *
- * @example
- * ```ts
- * const cache = new SourceCache(256); // 256 MB limit
- *
- * cache.set('kick.wav', decodedBuffer);
- *
- * const buf = cache.get('kick.wav');  // cache hit
- * console.log(cache.getCacheHitRate()); // 1.0
- * ```
+ * Uses JavaScript `Map` insertion-order semantics for O(1) LRU eviction:
+ * on every access the entry is deleted and re-inserted so it moves to the
+ * tail (most-recently-used), while eviction always removes from the head
+ * (least-recently-used).
  *
  * @module SourceCache
  */
@@ -30,7 +22,11 @@
  * {@link AudioBuffer} instances, bounded by a configurable memory limit.
  */
 export declare class SourceCache {
-    /** Map of source id to cache entry. */
+    /**
+     * Map of source id to cache entry.
+     * Iteration order = insertion order (oldest first).
+     * On access we delete + re-insert to move the entry to the tail.
+     */
     private cache;
     /** Maximum allowed memory usage in bytes. */
     private maxMemoryBytes;
@@ -48,8 +44,8 @@ export declare class SourceCache {
     /**
      * Retrieve a cached {@link AudioBuffer} by source id.
      *
-     * On a hit the entry's `lastAccess` timestamp is refreshed so it is less
-     * likely to be evicted.
+     * On a hit the entry is moved to the tail of the Map (most-recently-used)
+     * so it is less likely to be evicted.
      *
      * @param sourceId - Unique identifier for the audio source.
      * @returns The cached buffer, or `null` on a miss.
@@ -89,7 +85,8 @@ export declare class SourceCache {
      */
     evict(): void;
     /**
-     * Evict the single least-recently-used entry.
+     * Evict the single least-recently-used entry — O(1) using Map
+     * iteration order (head = oldest).
      */
     private evictOne;
     /**

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

@@ -1 +1 @@
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+{"version":3,"file":"SourceCache.d.ts","sourceRoot":"","sources":["../../src/audio/SourceCache.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;;;;;GAkBG;AAYH;;;GAGG;AACH,qBAAa,WAAW;IACpB;;;;OAIG;IACH,OAAO,CAAC,KAAK,CAAsC;IAEnD,6CAA6C;IAC7C,OAAO,CAAC,cAAc,CAAS;IAE/B,+CAA+C;IAC/C,OAAO,CAAC,kBAAkB,CAAK;IAI/B,0CAA0C;IAC1C,OAAO,CAAC,IAAI,CAAK;IACjB,8BAA8B;IAC9B,OAAO,CAAC,MAAM,CAAK;IAEnB;;;OAGG;gBACS,WAAW,GAAE,MAAY;IAQrC;;;;;;;;OAQG;IACH,GAAG,CAAC,QAAQ,EAAE,MAAM,GAAG,WAAW,GAAG,IAAI;IAazC;;;;;;;;;;OAUG;IACH,GAAG,CAAC,QAAQ,EAAE,MAAM,EAAE,MAAM,EAAE,WAAW,GAAG,IAAI;IAsBhD;;;;;OAKG;IACH,GAAG,CAAC,QAAQ,EAAE,MAAM,GAAG,OAAO;IAI9B;;;;OAIG;IACH,MAAM,CAAC,QAAQ,EAAE,MAAM,GAAG,IAAI;IAY9B;;;;;;OAMG;IACH,KAAK,IAAI,IAAI;IASb;;;OAGG;IACH,OAAO,CAAC,QAAQ;IAWhB;;OAEG;IACH,cAAc,IAAI,MAAM;IAIxB;;OAEG;IACH,gBAAgB,IAAI,MAAM;IAI1B;;;;OAIG;IACH,eAAe,IAAI,MAAM;IAKzB;;OAEG;IACH,IAAI,IAAI,IAAI,MAAM,CAEjB;IAED;;OAEG;IACH,KAAK,IAAI,IAAI;IAWb;;;;OAIG;IACH,OAAO,CAAC,kBAAkB;CAG7B"}

package/dist/audio/SourceCache.js

@@ -10,18 +10,10 @@
  * configurable memory limit and evicting least-recently-used entries when
  * the limit is exceeded.
  *
- * This class has **no** external dependencies (no audio framework imports)
- * and can be unit-tested in isolation.
- *
- * @example
- * ```ts
- * const cache = new SourceCache(256); // 256 MB limit
- *
- * cache.set('kick.wav', decodedBuffer);
- *
- * const buf = cache.get('kick.wav');  // cache hit
- * console.log(cache.getCacheHitRate()); // 1.0
- * ```
+ * Uses JavaScript `Map` insertion-order semantics for O(1) LRU eviction:
+ * on every access the entry is deleted and re-inserted so it moves to the
+ * tail (most-recently-used), while eviction always removes from the head
+ * (least-recently-used).
  *
  * @module SourceCache
  */
@@ -35,7 +27,11 @@ export class SourceCache {
      *   512 MB.
      */
     constructor(maxMemoryMB = 512) {
-        /** Map of source id to cache entry. */
+        /**
+         * Map of source id to cache entry.
+         * Iteration order = insertion order (oldest first).
+         * On access we delete + re-insert to move the entry to the tail.
+         */
         this.cache = new Map();
         /** Running total of cached memory in bytes. */
         this.currentMemoryBytes = 0;
@@ -52,8 +48,8 @@ export class SourceCache {
     /**
      * Retrieve a cached {@link AudioBuffer} by source id.
      *
-     * On a hit the entry's `lastAccess` timestamp is refreshed so it is less
-     * likely to be evicted.
+     * On a hit the entry is moved to the tail of the Map (most-recently-used)
+     * so it is less likely to be evicted.
      *
      * @param sourceId - Unique identifier for the audio source.
      * @returns The cached buffer, or `null` on a miss.
@@ -61,7 +57,9 @@ export class SourceCache {
     get(sourceId) {
         const entry = this.cache.get(sourceId);
         if (entry) {
-            entry.lastAccess = Date.now();
+            // Move to tail (most-recently-used) — O(1)
+            this.cache.delete(sourceId);
+            this.cache.set(sourceId, entry);
             this.hits++;
             return entry.buffer;
         }
@@ -85,16 +83,12 @@ export class SourceCache {
             this.remove(sourceId);
         }
         const sizeBytes = this.estimateBufferSize(buffer);
-        // Evict until there is room (or the cache is empty)
+        // Evict LRU entries until there is room (or the cache is empty)
         while (this.cache.size > 0 &&
             this.currentMemoryBytes + sizeBytes > this.maxMemoryBytes) {
             this.evictOne();
         }
-        const entry = {
-            buffer,
-            lastAccess: Date.now(),
-            sizeBytes,
-        };
+        const entry = { buffer, sizeBytes };
         this.cache.set(sourceId, entry);
         this.currentMemoryBytes += sizeBytes;
     }
@@ -136,19 +130,13 @@ export class SourceCache {
         }
     }
     /**
-     * Evict the single least-recently-used entry.
+     * Evict the single least-recently-used entry — O(1) using Map
+     * iteration order (head = oldest).
      */
     evictOne() {
-        let oldestId = null;
-        let oldestAccess = Infinity;
-        for (const [id, entry] of this.cache) {
-            if (entry.lastAccess < oldestAccess) {
-                oldestAccess = entry.lastAccess;
-                oldestId = id;
-            }
-        }
-        if (oldestId !== null) {
-            this.remove(oldestId);
+        const first = this.cache.keys().next();
+        if (!first.done) {
+            this.remove(first.value);
         }
     }
     // ---------------------------------------------------------------

package/dist/audio/SourceCache.js.map

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package/dist/audio/engine/Declick.d.ts

@@ -0,0 +1,98 @@
+/**
+ * Declick (DeclickAmp) - Provides smooth amplitude transitions
+ * to avoid clicks/pops at record start/stop and region boundaries.
+ * Inspired by Ardour's DeclickAmp.
+ */
+export declare class DeclickAmp {
+    private _targetGain;
+    private _currentGain;
+    private _sampleRate;
+    private _rampLength;
+    private _rampIncrement;
+    /**
+     * @param sampleRate - The audio sample rate (e.g. 44100, 48000)
+     * @param rampLengthMs - Duration of a full 0->1 or 1->0 ramp in ms. Defaults to ~1.5ms (~64 samples at 44100).
+     */
+    constructor(sampleRate: number, rampLengthMs?: number);
+    /**
+     * Set target gain (0 for fade out, 1 for fade in).
+     */
+    setTarget(gain: number): void;
+    /**
+     * Apply declick ramp to a buffer in-place.
+     * Uses linear ramping from _currentGain toward _targetGain.
+     * @returns true if ramp is still in progress after processing
+     */
+    applyToBuffer(buffer: Float32Array, offset: number, count: number): boolean;
+    /**
+     * Apply declick ramp to multi-channel buffers in-place.
+     * All channels share the same gain trajectory.
+     * @returns true if ramp is still in progress after processing
+     */
+    applyToBuffers(buffers: Float32Array[], offset: number, count: number): boolean;
+    /**
+     * Check if currently ramping (current gain has not yet reached target).
+     */
+    isRamping(): boolean;
+    /**
+     * Reset to initial state.
+     * @param initialGain - Starting gain value (default 0.0)
+     */
+    reset(initialGain?: number): void;
+    /**
+     * Get current gain value.
+     */
+    getCurrentGain(): number;
+}
+/**
+ * Declicker - Provides crossfade curves for loop boundaries.
+ * Used to create smooth transitions at loop points.
+ * Uses raised cosine (Hann) curves for perceptually smooth fades.
+ */
+export declare class Declicker {
+    private _fadeIn;
+    private _fadeOut;
+    private _length;
+    /**
+     * @param lengthSamples - Number of samples for the fade region
+     */
+    constructor(lengthSamples: number);
+    /**
+     * Pre-compute fade curves using a raised cosine (Hann window).
+     * fade_in[i]  = 0.5 * (1 - cos(pi * i / (N-1)))
+     * fade_out[i] = 0.5 * (1 + cos(pi * i / (N-1)))
+     * This ensures fade_in + fade_out = 1.0 at every sample (constant power crossfade).
+     */
+    computeCurves(): void;
+    /**
+     * Apply fade-in curve to the start of a buffer.
+     * @param buffer - The audio buffer to modify in-place
+     * @param offset - Optional offset into the buffer (default 0)
+     */
+    applyFadeIn(buffer: Float32Array, offset?: number): void;
+    /**
+     * Apply fade-out curve to the end of a buffer.
+     * @param offset - Optional: the sample index where the fade-out begins.
+     *                 Defaults to buffer.length - fadeLength.
+     */
+    applyFadeOut(buffer: Float32Array, offset?: number): void;
+    /**
+     * Apply crossfade between two buffers, writing result into output.
+     * output[i] = outgoing[i] * fadeOut[i] + incoming[i] * fadeIn[i]
+     * All three buffers must be at least _length samples long.
+     */
+    applyCrossfade(outgoing: Float32Array, incoming: Float32Array, output: Float32Array): void;
+    /**
+     * Get the pre-computed fade-in curve.
+     */
+    getFadeIn(): Float32Array;
+    /**
+     * Get the pre-computed fade-out curve.
+     */
+    getFadeOut(): Float32Array;
+    /**
+     * Get the length of the fade region in samples.
+     */
+    getLength(): number;
+}
+//# sourceMappingURL=Declick.d.ts.map

package/dist/audio/engine/Declick.d.ts.map

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