@siteed/audio-studio 3.0.0

package/build/esm/AudioAnalysis/audioFeaturesWasm.js

@@ -0,0 +1,126 @@
+let modulePromise = null;
+function getModule() {
+    if (!modulePromise) {
+        modulePromise = (async () => {
+            // Same WASM module as mel spectrogram (now includes audio features)
+            // @ts-expect-error -- prebuilt Emscripten JS glue has no .d.ts
+            const mod = await import('../../prebuilt/wasm/mel-spectrogram.js');
+            const factory = mod.default ?? mod;
+            return factory();
+        })().catch((err) => {
+            modulePromise = null;
+            throw err;
+        });
+    }
+    return modulePromise;
+}
+// --- Struct layout for CAudioFeaturesResult (wasm32) ---
+// Offset  0: float spectralCentroid  (4 bytes)
+// Offset  4: float spectralFlatness  (4 bytes)
+// Offset  8: float spectralRolloff   (4 bytes)
+// Offset 12: float spectralBandwidth (4 bytes)
+// Offset 16: float* mfcc             (4 bytes pointer)
+// Offset 20: int mfccCount           (4 bytes)
+// Offset 24: float* chromagram       (4 bytes pointer)
+// Offset 28: int chromagramCount     (4 bytes)
+const STRUCT_SIZE = 32;
+function readResult(Module, ptr) {
+    const spectralCentroid = Module.getValue(ptr, 'float');
+    const spectralFlatness = Module.getValue(ptr + 4, 'float');
+    const spectralRolloff = Module.getValue(ptr + 8, 'float');
+    const spectralBandwidth = Module.getValue(ptr + 12, 'float');
+    const mfccPtr = Module.getValue(ptr + 16, 'i32');
+    const mfccCount = Module.getValue(ptr + 20, 'i32');
+    const chromaPtr = Module.getValue(ptr + 24, 'i32');
+    const chromaCount = Module.getValue(ptr + 28, 'i32');
+    const mfcc = [];
+    if (mfccPtr && mfccCount > 0) {
+        const offset = mfccPtr >> 2;
+        for (let i = 0; i < mfccCount; i++) {
+            mfcc.push(Module.HEAPF32[offset + i]);
+        }
+    }
+    const chromagram = [];
+    if (chromaPtr && chromaCount > 0) {
+        const offset = chromaPtr >> 2;
+        for (let i = 0; i < chromaCount; i++) {
+            chromagram.push(Module.HEAPF32[offset + i]);
+        }
+    }
+    return {
+        spectralCentroid,
+        spectralFlatness,
+        spectralRolloff,
+        spectralBandwidth,
+        mfcc,
+        chromagram,
+    };
+}
+// --- Streaming (per-frame) API ---
+let streamingModule = null;
+let streamingFramePtr = 0;
+let streamingFrameCapacity = 0;
+let streamingResultPtr = 0;
+/**
+ * Initialise the WASM streaming audio features processor.
+ * Call once before computeAudioFeaturesFrameWasm().
+ */
+export async function initAudioFeaturesWasm(sampleRate, fftLength = 1024, nMfcc = 13, nMelFilters = 26, computeMfcc = true, computeChroma = true) {
+    const Module = await getModule();
+    streamingModule = Module;
+    Module._audio_features_init(sampleRate, fftLength, nMfcc, nMelFilters, computeMfcc ? 1 : 0, computeChroma ? 1 : 0);
+    // Pre-allocate result struct on WASM heap
+    if (streamingResultPtr)
+        Module._free(streamingResultPtr);
+    streamingResultPtr = Module._malloc(STRUCT_SIZE);
+    // Zero-initialize to prevent freeing garbage pointers on first use
+    Module.HEAPU8.fill(0, streamingResultPtr, streamingResultPtr + STRUCT_SIZE);
+    // Frame input buffer allocated on demand
+    streamingFrameCapacity = 0;
+    streamingFramePtr = 0;
+}
+/**
+ * Compute audio features for a single frame via WASM C++.
+ * Returns null if not initialised or on error.
+ */
+export function computeAudioFeaturesFrameWasm(samples) {
+    if (!streamingModule || !streamingResultPtr)
+        return null;
+    const Module = streamingModule;
+    // (Re-)allocate frame input buffer if needed
+    if (samples.length > streamingFrameCapacity) {
+        if (streamingFramePtr)
+            Module._free(streamingFramePtr);
+        streamingFramePtr = Module._malloc(samples.length * 4);
+        streamingFrameCapacity = samples.length;
+    }
+    // Copy samples to WASM heap
+    Module.HEAPF32.set(samples, streamingFramePtr >> 2);
+    const ok = Module._audio_features_compute_frame(streamingFramePtr, samples.length, streamingResultPtr);
+    if (!ok)
+        return null;
+    const result = readResult(Module, streamingResultPtr);
+    // Free internal arrays (mfcc, chromagram) allocated by C
+    Module._audio_features_free_arrays(streamingResultPtr);
+    return result;
+}
+// --- Batch API ---
+/**
+ * Compute audio features for a buffer of samples via WASM C++.
+ * Lazy-loads the WASM module on first call.
+ */
+export async function computeAudioFeaturesWasm(audioData, sampleRate, fftLength = 1024, nMfcc = 13, nMelFilters = 26, computeMfcc = true, computeChroma = true) {
+    const Module = await getModule();
+    const numSamples = audioData.length;
+    const inputPtr = Module._malloc(numSamples * 4);
+    Module.HEAPF32.set(audioData, inputPtr >> 2);
+    const resultPtr = Module._audio_features_compute(inputPtr, numSamples, sampleRate, fftLength, nMfcc, nMelFilters, computeMfcc ? 1 : 0, computeChroma ? 1 : 0);
+    Module._free(inputPtr);
+    if (resultPtr === 0) {
+        throw new Error('audio_features_compute returned null');
+    }
+    const result = readResult(Module, resultPtr);
+    Module._audio_features_free(resultPtr);
+    return result;
+}
+//# sourceMappingURL=audioFeaturesWasm.js.map

package/build/esm/AudioAnalysis/audioFeaturesWasm.js.map

@@ -0,0 +1 @@
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type { AudioFeaturesWasmModule } from './audio-features-wasm'\n\nexport interface AudioFeaturesWasmResult {\n    spectralCentroid: number\n    spectralFlatness: number\n    spectralRolloff: number\n    spectralBandwidth: number\n    mfcc: number[]\n    chromagram: number[]\n}\n\nlet modulePromise: Promise<AudioFeaturesWasmModule> | null = null\n\nfunction getModule(): Promise<AudioFeaturesWasmModule> {\n    if (!modulePromise) {\n        modulePromise = (async () => {\n            // Same WASM module as mel spectrogram (now includes audio features)\n            // @ts-expect-error -- prebuilt Emscripten JS glue has no .d.ts\n            const mod = await import('../../prebuilt/wasm/mel-spectrogram.js')\n            const factory = mod.default ?? mod\n            return factory() as Promise<AudioFeaturesWasmModule>\n        })().catch((err) => {\n            modulePromise = null\n            throw err\n        })\n    }\n    return modulePromise\n}\n\n// --- Struct layout for CAudioFeaturesResult (wasm32) ---\n// Offset  0: float spectralCentroid  (4 bytes)\n// Offset  4: float spectralFlatness  (4 bytes)\n// Offset  8: float spectralRolloff   (4 bytes)\n// Offset 12: float spectralBandwidth (4 bytes)\n// Offset 16: float* mfcc             (4 bytes pointer)\n// Offset 20: int mfccCount           (4 bytes)\n// Offset 24: float* chromagram       (4 bytes pointer)\n// Offset 28: int chromagramCount     (4 bytes)\nconst STRUCT_SIZE = 32\n\nfunction readResult(\n    Module: AudioFeaturesWasmModule,\n    ptr: number\n): AudioFeaturesWasmResult {\n    const spectralCentroid = Module.getValue(ptr, 'float')\n    const spectralFlatness = Module.getValue(ptr + 4, 'float')\n    const spectralRolloff = Module.getValue(ptr + 8, 'float')\n    const spectralBandwidth = Module.getValue(ptr + 12, 'float')\n\n    const mfccPtr = Module.getValue(ptr + 16, 'i32')\n    const mfccCount = Module.getValue(ptr + 20, 'i32')\n    const chromaPtr = Module.getValue(ptr + 24, 'i32')\n    const chromaCount = Module.getValue(ptr + 28, 'i32')\n\n    const mfcc: number[] = []\n    if (mfccPtr && mfccCount > 0) {\n        const offset = mfccPtr >> 2\n        for (let i = 0; i < mfccCount; i++) {\n            mfcc.push(Module.HEAPF32[offset + i])\n        }\n    }\n\n    const chromagram: number[] = []\n    if (chromaPtr && chromaCount > 0) {\n        const offset = chromaPtr >> 2\n        for (let i = 0; i < chromaCount; i++) {\n            chromagram.push(Module.HEAPF32[offset + i])\n        }\n    }\n\n    return {\n        spectralCentroid,\n        spectralFlatness,\n        spectralRolloff,\n        spectralBandwidth,\n        mfcc,\n        chromagram,\n    }\n}\n\n// --- Streaming (per-frame) API ---\n\nlet streamingModule: AudioFeaturesWasmModule | null = null\nlet streamingFramePtr = 0\nlet streamingFrameCapacity = 0\nlet streamingResultPtr = 0\n\n/**\n * Initialise the WASM streaming audio features processor.\n * Call once before computeAudioFeaturesFrameWasm().\n */\nexport async function initAudioFeaturesWasm(\n    sampleRate: number,\n    fftLength = 1024,\n    nMfcc = 13,\n    nMelFilters = 26,\n    computeMfcc = true,\n    computeChroma = true\n): Promise<void> {\n    const Module = await getModule()\n    streamingModule = Module\n\n    Module._audio_features_init(\n        sampleRate,\n        fftLength,\n        nMfcc,\n        nMelFilters,\n        computeMfcc ? 1 : 0,\n        computeChroma ? 1 : 0\n    )\n\n    // Pre-allocate result struct on WASM heap\n    if (streamingResultPtr) Module._free(streamingResultPtr)\n    streamingResultPtr = Module._malloc(STRUCT_SIZE)\n    // Zero-initialize to prevent freeing garbage pointers on first use\n    Module.HEAPU8.fill(0, streamingResultPtr, streamingResultPtr + STRUCT_SIZE)\n\n    // Frame input buffer allocated on demand\n    streamingFrameCapacity = 0\n    streamingFramePtr = 0\n}\n\n/**\n * Compute audio features for a single frame via WASM C++.\n * Returns null if not initialised or on error.\n */\nexport function computeAudioFeaturesFrameWasm(\n    samples: Float32Array\n): AudioFeaturesWasmResult | null {\n    if (!streamingModule || !streamingResultPtr) return null\n    const Module = streamingModule\n\n    // (Re-)allocate frame input buffer if needed\n    if (samples.length > streamingFrameCapacity) {\n        if (streamingFramePtr) Module._free(streamingFramePtr)\n        streamingFramePtr = Module._malloc(samples.length * 4)\n        streamingFrameCapacity = samples.length\n    }\n\n    // Copy samples to WASM heap\n    Module.HEAPF32.set(samples, streamingFramePtr >> 2)\n\n    const ok = Module._audio_features_compute_frame(\n        streamingFramePtr,\n        samples.length,\n        streamingResultPtr\n    )\n    if (!ok) return null\n\n    const result = readResult(Module, streamingResultPtr)\n\n    // Free internal arrays (mfcc, chromagram) allocated by C\n    Module._audio_features_free_arrays(streamingResultPtr)\n\n    return result\n}\n\n// --- Batch API ---\n\n/**\n * Compute audio features for a buffer of samples via WASM C++.\n * Lazy-loads the WASM module on first call.\n */\nexport async function computeAudioFeaturesWasm(\n    audioData: Float32Array,\n    sampleRate: number,\n    fftLength = 1024,\n    nMfcc = 13,\n    nMelFilters = 26,\n    computeMfcc = true,\n    computeChroma = true\n): Promise<AudioFeaturesWasmResult> {\n    const Module = await getModule()\n\n    const numSamples = audioData.length\n    const inputPtr = Module._malloc(numSamples * 4)\n    Module.HEAPF32.set(audioData, inputPtr >> 2)\n\n    const resultPtr = Module._audio_features_compute(\n        inputPtr,\n        numSamples,\n        sampleRate,\n        fftLength,\n        nMfcc,\n        nMelFilters,\n        computeMfcc ? 1 : 0,\n        computeChroma ? 1 : 0\n    )\n\n    Module._free(inputPtr)\n\n    if (resultPtr === 0) {\n        throw new Error('audio_features_compute returned null')\n    }\n\n    const result = readResult(Module, resultPtr)\n    Module._audio_features_free(resultPtr)\n\n    return result\n}\n"]}

package/build/esm/AudioAnalysis/extractAudioAnalysis.js

@@ -0,0 +1,205 @@
+import AudioStudioModule from '../AudioStudioModule';
+import { isWeb } from '../constants';
+import { processAudioBuffer } from '../utils/audioProcessing';
+import { cleanNativeOptions } from '../utils/cleanNativeOptions';
+import { convertPCMToFloat32 } from '../utils/convertPCMToFloat32';
+import crc32 from '../utils/crc32';
+import { getWavFileInfo } from '../utils/getWavFileInfo';
+import { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web';
+import { wasmGlueJs } from '../workers/wasmGlueString.web';
+function calculateCRC32ForDataPoint(data) {
+    // Convert float array to byte array for CRC32
+    const byteArray = new Uint8Array(data.length * 4);
+    const dataView = new DataView(byteArray.buffer);
+    for (let i = 0; i < data.length; i++) {
+        dataView.setFloat32(i * 4, data[i], true);
+    }
+    return crc32.buf(byteArray);
+}
+/**
+ * Extracts detailed audio analysis from the specified audio file or buffer.
+ * Supports either time-based or byte-based ranges for flexibility in analysis.
+ *
+ * @param props - The options for extraction, including file URI, ranges, and decoding settings.
+ * @returns A promise that resolves to the audio analysis data.
+ * @throws {Error} If both time and byte ranges are provided or if required parameters are missing.
+ */
+export async function extractAudioAnalysis(props) {
+    const { fileUri, arrayBuffer, decodingOptions, logger, segmentDurationMs = 100, features, } = props;
+    if (isWeb) {
+        try {
+            // Create AudioContext here
+            const audioContext = new (window.AudioContext ||
+                window.webkitAudioContext)({
+                sampleRate: decodingOptions?.targetSampleRate ?? 16000,
+            });
+            try {
+                const processedBuffer = await processAudioBuffer({
+                    arrayBuffer,
+                    fileUri,
+                    targetSampleRate: decodingOptions?.targetSampleRate ?? 16000,
+                    targetChannels: decodingOptions?.targetChannels ?? 1,
+                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,
+                    startTimeMs: 'startTimeMs' in props ? props.startTimeMs : undefined,
+                    endTimeMs: 'endTimeMs' in props ? props.endTimeMs : undefined,
+                    position: 'position' in props ? props.position : undefined,
+                    length: 'length' in props ? props.length : undefined,
+                    audioContext, // Pass the context we created
+                    logger,
+                });
+                const channelData = processedBuffer.buffer.getChannelData(0);
+                // Create worker blob: WASM glue (defines createMelSpectrogramModule) + worker code
+                const blob = new Blob([wasmGlueJs, '\n', InlineFeaturesExtractor], { type: 'application/javascript' });
+                const workerUrl = URL.createObjectURL(blob);
+                const worker = new Worker(workerUrl);
+                return new Promise((resolve, reject) => {
+                    worker.onmessage = (event) => {
+                        if (event.data.error) {
+                            reject(new Error(event.data.error));
+                            return;
+                        }
+                        const result = event.data.result;
+                        // Calculate CRC32 after worker completes if requested
+                        if (features?.crc32) {
+                            const samplesPerSegment = Math.floor((processedBuffer.sampleRate *
+                                segmentDurationMs) /
+                                1000);
+                            result.dataPoints = result.dataPoints.map((point, index) => {
+                                const startSample = index * samplesPerSegment;
+                                const segmentData = channelData.slice(startSample, startSample + samplesPerSegment);
+                                return {
+                                    ...point,
+                                    features: {
+                                        ...point.features,
+                                        crc32: calculateCRC32ForDataPoint(segmentData),
+                                    },
+                                };
+                            });
+                        }
+                        URL.revokeObjectURL(workerUrl);
+                        worker.terminate();
+                        resolve(result);
+                    };
+                    worker.onerror = (error) => {
+                        URL.revokeObjectURL(workerUrl);
+                        worker.terminate();
+                        reject(error);
+                    };
+                    worker.postMessage({
+                        channelData,
+                        sampleRate: processedBuffer.sampleRate,
+                        segmentDurationMs,
+                        bitDepth: decodingOptions?.targetBitDepth ?? 32,
+                        numberOfChannels: processedBuffer.channels,
+                        fullAudioDurationMs: processedBuffer.durationMs,
+                        // enableLogging: !!logger,
+                        features,
+                    });
+                });
+            }
+            finally {
+                await audioContext.close();
+            }
+        }
+        catch (error) {
+            logger?.error('Failed to process audio:', error);
+            throw error;
+        }
+    }
+    else {
+        // Strip non-serializable fields — logger and arrayBuffer cause
+        // "Cannot convert '[object Object]' to a Kotlin type" on Android.
+        const { logger: _logger, arrayBuffer: _arrayBuffer, ...nativeOptions } = props;
+        // Clean undefined values to avoid Android Kotlin bridge crash
+        return await AudioStudioModule.extractAudioAnalysis(cleanNativeOptions(nativeOptions));
+    }
+}
+/**
+ * Analyzes WAV files without decoding, preserving original PCM values.
+ * Use this function when you need to ensure the analysis matches other software by avoiding any transformations.
+ *
+ * @param props - The options for WAV analysis, including file URI and range.
+ * @returns A promise that resolves to the audio analysis data.
+ */
+export const extractRawWavAnalysis = async ({ fileUri, segmentDurationMs = 100, // Default to 100ms
+arrayBuffer, bitDepth, durationMs, sampleRate, numberOfChannels, features, logger, position = 0, length, }) => {
+    if (isWeb) {
+        if (!arrayBuffer && !fileUri) {
+            throw new Error('Either arrayBuffer or fileUri must be provided');
+        }
+        if (!arrayBuffer) {
+            logger?.log(`fetching fileUri`, fileUri);
+            const response = await fetch(fileUri);
+            if (!response.ok) {
+                throw new Error(`Failed to fetch fileUri: ${response.statusText}`);
+            }
+            arrayBuffer = await response.arrayBuffer();
+            logger?.log(`fetched fileUri`, arrayBuffer.byteLength, arrayBuffer);
+        }
+        // Create a new copy of the ArrayBuffer to avoid detachment issues
+        const bufferCopy = arrayBuffer.slice(0);
+        logger?.log(`extractAudioAnalysis bitDepth=${bitDepth} len=${bufferCopy.byteLength}`, bufferCopy.slice(0, 100));
+        let actualBitDepth = bitDepth;
+        if (!actualBitDepth) {
+            logger?.log(`extractAudioAnalysis bitDepth not provided -- getting wav file info`);
+            const fileInfo = await getWavFileInfo(bufferCopy);
+            actualBitDepth = fileInfo.bitDepth;
+        }
+        logger?.log(`extractAudioAnalysis actualBitDepth=${actualBitDepth}`);
+        const { pcmValues: channelData, min, max, } = await convertPCMToFloat32({
+            buffer: arrayBuffer,
+            bitDepth: actualBitDepth,
+        });
+        logger?.log(`extractAudioAnalysis convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`);
+        // Apply position and length constraints to channelData if specified
+        const startIndex = position;
+        const endIndex = length ? startIndex + length : channelData.length;
+        const constrainedChannelData = channelData.slice(startIndex, endIndex);
+        return new Promise((resolve, reject) => {
+            const blob = new Blob([wasmGlueJs, '\n', InlineFeaturesExtractor], {
+                type: 'application/javascript',
+            });
+            const url = URL.createObjectURL(blob);
+            const worker = new Worker(url);
+            worker.onmessage = (event) => {
+                URL.revokeObjectURL(url);
+                worker.terminate();
+                resolve(event.data.result);
+            };
+            worker.onerror = (error) => {
+                URL.revokeObjectURL(url);
+                worker.terminate();
+                reject(error);
+            };
+            worker.postMessage({
+                command: 'process',
+                channelData: constrainedChannelData,
+                sampleRate,
+                segmentDurationMs,
+                logger,
+                bitDepth,
+                fullAudioDurationMs: durationMs,
+                numberOfChannels,
+            });
+        });
+    }
+    else {
+        if (!fileUri) {
+            throw new Error('fileUri is required');
+        }
+        logger?.log(`extractAudioAnalysis`, {
+            fileUri,
+            segmentDurationMs,
+        });
+        const res = await AudioStudioModule.extractAudioAnalysis({
+            fileUri,
+            segmentDurationMs,
+            features,
+            position,
+            length,
+        });
+        logger?.log(`extractAudioAnalysis`, res);
+        return res;
+    }
+};
+//# sourceMappingURL=extractAudioAnalysis.js.map

package/build/esm/AudioAnalysis/extractAudioAnalysis.js.map

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packages/audio-studio/src/AudioAnalysis/extractAudioAnalysis.ts\n/**\n * This module provides functions for extracting and analyzing audio data.\n * - `extractAudioAnalysis`: For detailed analysis with customizable ranges and decoding options.\n * - `extractWavAudioAnalysis`: For analyzing WAV files without decoding, preserving original PCM values.\n * - `extractPreview`: For generating quick previews of audio waveforms, optimized for UI rendering.\n */\nimport { ConsoleLike } from '../AudioStudio.types'\nimport AudioStudioModule from '../AudioStudioModule'\nimport { isWeb } from '../constants'\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DataPoint,\n    DecodingConfig,\n} from './AudioAnalysis.types'\nimport { processAudioBuffer } from '../utils/audioProcessing'\nimport { cleanNativeOptions } from '../utils/cleanNativeOptions'\nimport { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'\nimport crc32 from '../utils/crc32'\nimport { getWavFileInfo, WavFileInfo } from '../utils/getWavFileInfo'\nimport { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web'\nimport { wasmGlueJs } from '../workers/wasmGlueString.web'\n\nfunction calculateCRC32ForDataPoint(data: Float32Array): number {\n    // Convert float array to byte array for CRC32\n    const byteArray = new Uint8Array(data.length * 4)\n    const dataView = new DataView(byteArray.buffer)\n\n    for (let i = 0; i < data.length; i++) {\n        dataView.setFloat32(i * 4, data[i], true)\n    }\n\n    return crc32.buf(byteArray)\n}\n\nexport interface ExtractWavAudioAnalysisProps {\n    fileUri?: string // should provide either fileUri or arrayBuffer\n    wavMetadata?: WavFileInfo\n    arrayBuffer?: ArrayBuffer\n    bitDepth?: number\n    durationMs?: number\n    sampleRate?: number\n    numberOfChannels?: number\n    position?: number // Optional number of bytes to skip. Default is 0\n    length?: number // Optional number of bytes to read.\n    segmentDurationMs?: number // Optional number of points per second. Use to reduce the number of points and compute the number of datapoints to return.\n    features?: AudioFeaturesOptions\n    featuresExtratorUrl?: string\n    logger?: ConsoleLike\n    decodingOptions?: DecodingConfig\n}\n\n// Define base options interface with common properties\ninterface BaseExtractOptions {\n    fileUri?: string\n    arrayBuffer?: ArrayBuffer\n    /**\n     * Duration of each analysis segment in milliseconds. Defaults to 100ms if not specified.\n     */\n    segmentDurationMs?: number\n    features?: AudioFeaturesOptions\n    decodingOptions?: DecodingConfig\n    logger?: ConsoleLike\n}\n\n// Time-based range options\ninterface TimeRangeOptions extends BaseExtractOptions {\n    startTimeMs?: number\n    endTimeMs?: number\n    position?: never\n    length?: never\n}\n\n// Byte-based range options\ninterface ByteRangeOptions extends BaseExtractOptions {\n    position?: number\n    length?: number\n    startTimeMs?: never\n    endTimeMs?: never\n}\n\n/**\n * Options for extracting audio analysis.\n * - For time-based analysis, provide `startTimeMs` and `endTimeMs`.\n * - For byte-based analysis, provide `position` and `length`.\n * - Do not mix time and byte ranges.\n */\nexport type ExtractAudioAnalysisProps = TimeRangeOptions | ByteRangeOptions\n\n/**\n * Extracts detailed audio analysis from the specified audio file or buffer.\n * Supports either time-based or byte-based ranges for flexibility in analysis.\n *\n * @param props - The options for extraction, including file URI, ranges, and decoding settings.\n * @returns A promise that resolves to the audio analysis data.\n * @throws {Error} If both time and byte ranges are provided or if required parameters are missing.\n */\nexport async function extractAudioAnalysis(\n    props: ExtractAudioAnalysisProps\n): Promise<AudioAnalysis> {\n    const {\n        fileUri,\n        arrayBuffer,\n        decodingOptions,\n        logger,\n        segmentDurationMs = 100,\n        features,\n    } = props\n\n    if (isWeb) {\n        try {\n            // Create AudioContext here\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)({\n                sampleRate: decodingOptions?.targetSampleRate ?? 16000,\n            })\n\n            try {\n                const processedBuffer = await processAudioBuffer({\n                    arrayBuffer,\n                    fileUri,\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate ?? 16000,\n                    targetChannels: decodingOptions?.targetChannels ?? 1,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                    startTimeMs:\n                        'startTimeMs' in props ? props.startTimeMs : undefined,\n                    endTimeMs:\n                        'endTimeMs' in props ? props.endTimeMs : undefined,\n                    position: 'position' in props ? props.position : undefined,\n                    length: 'length' in props ? props.length : undefined,\n                    audioContext, // Pass the context we created\n                    logger,\n                })\n\n                const channelData = processedBuffer.buffer.getChannelData(0)\n\n                // Create worker blob: WASM glue (defines createMelSpectrogramModule) + worker code\n                const blob = new Blob(\n                    [wasmGlueJs, '\\n', InlineFeaturesExtractor],\n                    { type: 'application/javascript' }\n                )\n                const workerUrl = URL.createObjectURL(blob)\n                const worker = new Worker(workerUrl)\n\n                return new Promise((resolve, reject) => {\n                    worker.onmessage = (event) => {\n                        if (event.data.error) {\n                            reject(new Error(event.data.error))\n                            return\n                        }\n\n                        const result: AudioAnalysis = event.data.result\n                        // Calculate CRC32 after worker completes if requested\n                        if (features?.crc32) {\n                            const samplesPerSegment = Math.floor(\n                                (processedBuffer.sampleRate *\n                                    segmentDurationMs) /\n                                    1000\n                            )\n\n                            result.dataPoints = result.dataPoints.map(\n                                (point: DataPoint, index: number) => {\n                                    const startSample =\n                                        index * samplesPerSegment\n                                    const segmentData = channelData.slice(\n                                        startSample,\n                                        startSample + samplesPerSegment\n                                    )\n\n                                    return {\n                                        ...point,\n                                        features: {\n                                            ...point.features,\n                                            crc32: calculateCRC32ForDataPoint(\n                                                segmentData\n                                            ),\n                                        },\n                                    }\n                                }\n                            )\n                        }\n\n                        URL.revokeObjectURL(workerUrl)\n                        worker.terminate()\n                        resolve(result)\n                    }\n\n                    worker.onerror = (error) => {\n                        URL.revokeObjectURL(workerUrl)\n                        worker.terminate()\n                        reject(error)\n                    }\n\n                    worker.postMessage({\n                        channelData,\n                        sampleRate: processedBuffer.sampleRate,\n                        segmentDurationMs,\n                        bitDepth: decodingOptions?.targetBitDepth ?? 32,\n                        numberOfChannels: processedBuffer.channels,\n                        fullAudioDurationMs: processedBuffer.durationMs,\n                        // enableLogging: !!logger,\n                        features,\n                    })\n                })\n            } finally {\n                await audioContext.close()\n            }\n        } catch (error) {\n            logger?.error('Failed to process audio:', error)\n            throw error\n        }\n    } else {\n        // Strip non-serializable fields — logger and arrayBuffer cause\n        // \"Cannot convert '[object Object]' to a Kotlin type\" on Android.\n        const {\n            logger: _logger,\n            arrayBuffer: _arrayBuffer,\n            ...nativeOptions\n        } = props\n        // Clean undefined values to avoid Android Kotlin bridge crash\n        return await AudioStudioModule.extractAudioAnalysis(\n            cleanNativeOptions(nativeOptions)\n        )\n    }\n}\n\n/**\n * Analyzes WAV files without decoding, preserving original PCM values.\n * Use this function when you need to ensure the analysis matches other software by avoiding any transformations.\n *\n * @param props - The options for WAV analysis, including file URI and range.\n * @returns A promise that resolves to the audio analysis data.\n */\nexport const extractRawWavAnalysis = async ({\n    fileUri,\n    segmentDurationMs = 100, // Default to 100ms\n    arrayBuffer,\n    bitDepth,\n    durationMs,\n    sampleRate,\n    numberOfChannels,\n    features,\n    logger,\n    position = 0,\n    length,\n}: ExtractWavAudioAnalysisProps): Promise<AudioAnalysis> => {\n    if (isWeb) {\n        if (!arrayBuffer && !fileUri) {\n            throw new Error('Either arrayBuffer or fileUri must be provided')\n        }\n\n        if (!arrayBuffer) {\n            logger?.log(`fetching fileUri`, fileUri)\n            const response = await fetch(fileUri!)\n\n            if (!response.ok) {\n                throw new Error(\n                    `Failed to fetch fileUri: ${response.statusText}`\n                )\n            }\n\n            arrayBuffer = await response.arrayBuffer()\n            logger?.log(`fetched fileUri`, arrayBuffer.byteLength, arrayBuffer)\n        }\n\n        // Create a new copy of the ArrayBuffer to avoid detachment issues\n        const bufferCopy = arrayBuffer.slice(0)\n        logger?.log(\n            `extractAudioAnalysis bitDepth=${bitDepth} len=${bufferCopy.byteLength}`,\n            bufferCopy.slice(0, 100)\n        )\n\n        let actualBitDepth = bitDepth\n        if (!actualBitDepth) {\n            logger?.log(\n                `extractAudioAnalysis bitDepth not provided -- getting wav file info`\n            )\n            const fileInfo = await getWavFileInfo(bufferCopy)\n            actualBitDepth = fileInfo.bitDepth\n        }\n        logger?.log(`extractAudioAnalysis actualBitDepth=${actualBitDepth}`)\n\n        const {\n            pcmValues: channelData,\n            min,\n            max,\n        } = await convertPCMToFloat32({\n            buffer: arrayBuffer,\n            bitDepth: actualBitDepth,\n        })\n        logger?.log(\n            `extractAudioAnalysis convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`\n        )\n\n        // Apply position and length constraints to channelData if specified\n        const startIndex = position\n        const endIndex = length ? startIndex + length : channelData.length\n        const constrainedChannelData = channelData.slice(startIndex, endIndex)\n\n        return new Promise((resolve, reject) => {\n            const blob = new Blob([wasmGlueJs, '\\n', InlineFeaturesExtractor], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            const worker = new Worker(url)\n\n            worker.onmessage = (event) => {\n                URL.revokeObjectURL(url)\n                worker.terminate()\n                resolve(event.data.result)\n            }\n\n            worker.onerror = (error) => {\n                URL.revokeObjectURL(url)\n                worker.terminate()\n                reject(error)\n            }\n\n            worker.postMessage({\n                command: 'process',\n                channelData: constrainedChannelData,\n                sampleRate,\n                segmentDurationMs,\n                logger,\n                bitDepth,\n                fullAudioDurationMs: durationMs,\n                numberOfChannels,\n            })\n        })\n    } else {\n        if (!fileUri) {\n            throw new Error('fileUri is required')\n        }\n        logger?.log(`extractAudioAnalysis`, {\n            fileUri,\n            segmentDurationMs,\n        })\n        const res = await AudioStudioModule.extractAudioAnalysis({\n            fileUri,\n            segmentDurationMs,\n            features,\n            position,\n            length,\n        })\n        logger?.log(`extractAudioAnalysis`, res)\n        return res\n    }\n}\n"]}

package/build/esm/AudioAnalysis/extractAudioData.js

@@ -0,0 +1,14 @@
+import AudioStudioModule from '../AudioStudioModule';
+import { isWeb } from '../constants';
+import { cleanNativeOptions } from '../utils/cleanNativeOptions';
+export const extractAudioData = async (props) => {
+    if (isWeb) {
+        // Web implementation handles logger natively in AudioStudioModule.ts
+        return await AudioStudioModule.extractAudioData(props);
+    }
+    // Native: only pass serializable fields — logger causes crash on Android
+    const { logger: _logger, ...nativeOptions } = props;
+    // Clean undefined values to avoid Android Kotlin bridge crash
+    return await AudioStudioModule.extractAudioData(cleanNativeOptions(nativeOptions));
+};
+//# sourceMappingURL=extractAudioData.js.map

package/build/esm/AudioAnalysis/extractAudioData.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"extractAudioData.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/extractAudioData.ts"],"names":[],"mappings":"AACA,OAAO,iBAAiB,MAAM,sBAAsB,CAAA;AACpD,OAAO,EAAE,KAAK,EAAE,MAAM,cAAc,CAAA;AACpC,OAAO,EAAE,kBAAkB,EAAE,MAAM,6BAA6B,CAAA;AAEhE,MAAM,CAAC,MAAM,gBAAgB,GAAG,KAAK,EAAE,KAA8B,EAAE,EAAE;IACrE,IAAI,KAAK,EAAE,CAAC;QACR,qEAAqE;QACrE,OAAO,MAAM,iBAAiB,CAAC,gBAAgB,CAAC,KAAK,CAAC,CAAA;IAC1D,CAAC;IACD,yEAAyE;IACzE,MAAM,EAAE,MAAM,EAAE,OAAO,EAAE,GAAG,aAAa,EAAE,GAAG,KAAK,CAAA;IACnD,8DAA8D;IAC9D,OAAO,MAAM,iBAAiB,CAAC,gBAAgB,CAC3C,kBAAkB,CAAC,aAAa,CAAC,CACpC,CAAA;AACL,CAAC,CAAA","sourcesContent":["import { ExtractAudioDataOptions } from '../AudioStudio.types'\nimport AudioStudioModule from '../AudioStudioModule'\nimport { isWeb } from '../constants'\nimport { cleanNativeOptions } from '../utils/cleanNativeOptions'\n\nexport const extractAudioData = async (props: ExtractAudioDataOptions) => {\n    if (isWeb) {\n        // Web implementation handles logger natively in AudioStudioModule.ts\n        return await AudioStudioModule.extractAudioData(props)\n    }\n    // Native: only pass serializable fields — logger causes crash on Android\n    const { logger: _logger, ...nativeOptions } = props\n    // Clean undefined values to avoid Android Kotlin bridge crash\n    return await AudioStudioModule.extractAudioData(\n        cleanNativeOptions(nativeOptions)\n    )\n}\n"]}

package/build/esm/AudioAnalysis/extractMelSpectrogram.js

@@ -0,0 +1,86 @@
+/**
+ * @experimental This feature is experimental and currently only available on Android.
+ * The API may change in future versions.
+ */
+import { AudioStudioModule } from '..';
+import { isWeb } from '../constants';
+import { computeMelSpectrogramWasm } from './melSpectrogramWasm';
+import { processAudioBuffer, } from '../utils/audioProcessing';
+import { cleanNativeOptions } from '../utils/cleanNativeOptions';
+/**
+ * Maximum duration in milliseconds that extractMelSpectrogram will process in a single call.
+ * The C++ core requires the entire trimmed range as a contiguous float array in memory,
+ * so this bound prevents OOM on all platforms. Callers needing longer ranges can iterate
+ * in windows of this size using startTimeMs/endTimeMs.
+ */
+export const MAX_DURATION_MS = 30_000;
+/**
+ * Extracts a mel spectrogram from audio data
+ *
+ * @experimental This feature is experimental.
+ * Uses shared C++ implementation on all platforms (native on iOS/Android, WASM on web).
+ */
+export async function extractMelSpectrogram(options) {
+    let { fileUri, arrayBuffer, windowSizeMs, hopLengthMs, nMels, fMin = 0, fMax, windowType = 'hann', normalize = false, logScale = true, decodingOptions, startTimeMs, endTimeMs, logger, } = options;
+    // Apply max duration guard
+    if (startTimeMs == null && endTimeMs == null) {
+        startTimeMs = 0;
+        endTimeMs = MAX_DURATION_MS;
+        logger?.warn?.(`extractMelSpectrogram: no time range specified, defaulting to 0–${MAX_DURATION_MS}ms`);
+    }
+    else {
+        const start = startTimeMs ?? 0;
+        const end = endTimeMs ?? start + MAX_DURATION_MS;
+        if (end - start > MAX_DURATION_MS) {
+            endTimeMs = start + MAX_DURATION_MS;
+            logger?.warn?.(`extractMelSpectrogram: requested range ${end - start}ms exceeds max ${MAX_DURATION_MS}ms, clamping endTimeMs to ${endTimeMs}`);
+        }
+    }
+    if (isWeb) {
+        // Create audio context
+        const audioContext = new (window.AudioContext ||
+            window.webkitAudioContext)();
+        try {
+            // Process audio data using the existing utility
+            const processedAudio = await processAudioBuffer({
+                arrayBuffer,
+                fileUri,
+                targetSampleRate: decodingOptions?.targetSampleRate || 16000,
+                targetChannels: decodingOptions?.targetChannels || 1,
+                normalizeAudio: decodingOptions?.normalizeAudio ?? false,
+                startTimeMs,
+                endTimeMs,
+                audioContext,
+                logger: options.logger,
+            });
+            // Calculate window and hop size in samples
+            const sampleRate = processedAudio.sampleRate;
+            const windowSize = Math.floor((windowSizeMs * sampleRate) / 1000);
+            const hopLength = Math.floor((hopLengthMs * sampleRate) / 1000);
+            const maxFreq = fMax || sampleRate / 2;
+            // Extract the mel spectrogram via WASM (same C++ as native)
+            const spectrogram = await computeMelSpectrogramWasm(processedAudio.channelData, sampleRate, nMels, windowSize, hopLength, fMin, maxFreq, windowType, normalize, logScale);
+            const timeSteps = spectrogram.length;
+            return {
+                spectrogram,
+                sampleRate,
+                nMels,
+                timeSteps,
+                durationMs: processedAudio.durationMs,
+            };
+        }
+        catch (error) {
+            logger?.error('Error extracting mel spectrogram:', error);
+            throw error;
+        }
+        finally {
+            // Close the audio context
+            await audioContext.close();
+        }
+    }
+    // Strip logger/arrayBuffer (non-serializable) then clean undefined values
+    // to avoid Android "Cannot convert '[object Object]' to Kotlin type" crash
+    const { logger: _logger, arrayBuffer: _arrayBuffer, ...nativeOptions } = options;
+    return AudioStudioModule.extractMelSpectrogram(cleanNativeOptions(nativeOptions));
+}
+//# sourceMappingURL=extractMelSpectrogram.js.map

package/build/esm/AudioAnalysis/extractMelSpectrogram.js.map

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Callers needing longer ranges can iterate\n * in windows of this size using startTimeMs/endTimeMs.\n */\nexport const MAX_DURATION_MS = 30_000\n\n/**\n * Extracts a mel spectrogram from audio data\n *\n * @experimental This feature is experimental.\n * Uses shared C++ implementation on all platforms (native on iOS/Android, WASM on web).\n */\nexport async function extractMelSpectrogram(\n    options: ExtractMelSpectrogramOptions\n): Promise<MelSpectrogram> {\n    let {\n        fileUri,\n        arrayBuffer,\n        windowSizeMs,\n        hopLengthMs,\n        nMels,\n        fMin = 0,\n        fMax,\n        windowType = 'hann',\n        normalize = false,\n        logScale = true,\n        decodingOptions,\n        startTimeMs,\n        endTimeMs,\n        logger,\n    } = options\n\n    // Apply max duration guard\n    if (startTimeMs == null && endTimeMs == null) {\n        startTimeMs = 0\n        endTimeMs = MAX_DURATION_MS\n        logger?.warn?.(\n            `extractMelSpectrogram: no time range specified, defaulting to 0–${MAX_DURATION_MS}ms`\n        )\n    } else {\n        const start = startTimeMs ?? 0\n        const end = endTimeMs ?? start + MAX_DURATION_MS\n        if (end - start > MAX_DURATION_MS) {\n            endTimeMs = start + MAX_DURATION_MS\n            logger?.warn?.(\n                `extractMelSpectrogram: requested range ${end - start}ms exceeds max ${MAX_DURATION_MS}ms, clamping endTimeMs to ${endTimeMs}`\n            )\n        }\n    }\n\n    if (isWeb) {\n        // Create audio context\n        const audioContext = new (window.AudioContext ||\n            (window as any).webkitAudioContext)()\n\n        try {\n            // Process audio data using the existing utility\n            const processedAudio: ProcessedAudioData = await processAudioBuffer(\n                {\n                    arrayBuffer,\n                    fileUri,\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate || 16000,\n                    targetChannels: decodingOptions?.targetChannels || 1,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                    startTimeMs,\n                    endTimeMs,\n                    audioContext,\n                    logger: options.logger,\n                }\n            )\n\n            // Calculate window and hop size in samples\n            const sampleRate = processedAudio.sampleRate\n            const windowSize = Math.floor((windowSizeMs * sampleRate) / 1000)\n            const hopLength = Math.floor((hopLengthMs * sampleRate) / 1000)\n            const maxFreq = fMax || sampleRate / 2\n\n            // Extract the mel spectrogram via WASM (same C++ as native)\n            const spectrogram = await computeMelSpectrogramWasm(\n                processedAudio.channelData,\n                sampleRate,\n                nMels,\n                windowSize,\n                hopLength,\n                fMin,\n                maxFreq,\n                windowType,\n                normalize,\n                logScale\n            )\n\n            const timeSteps = spectrogram.length\n\n            return {\n                spectrogram,\n                sampleRate,\n                nMels,\n                timeSteps,\n                durationMs: processedAudio.durationMs,\n            }\n        } catch (error) {\n            logger?.error('Error extracting mel spectrogram:', error)\n            throw error\n        } finally {\n            // Close the audio context\n            await audioContext.close()\n        }\n    }\n    // Strip logger/arrayBuffer (non-serializable) then clean undefined values\n    // to avoid Android \"Cannot convert '[object Object]' to Kotlin type\" crash\n    const {\n        logger: _logger,\n        arrayBuffer: _arrayBuffer,\n        ...nativeOptions\n    } = options\n    return AudioStudioModule.extractMelSpectrogram(\n        cleanNativeOptions(nativeOptions)\n    )\n}\n"]}

package/build/esm/AudioAnalysis/extractPreview.js

@@ -0,0 +1,25 @@
+import { extractAudioAnalysis } from './extractAudioAnalysis';
+/**
+ * Generates a simplified preview of the audio waveform for quick visualization.
+ * Ideal for UI rendering with a specified number of points.
+ *
+ * @param options - The options for the preview, including file URI and time range.
+ * @returns A promise that resolves to the audio preview data.
+ */
+export async function extractPreview({ fileUri, numberOfPoints = 100, startTimeMs = 0, endTimeMs = 30000, // First 30 seconds
+decodingOptions, logger, }) {
+    const durationMs = endTimeMs - startTimeMs;
+    const segmentDurationMs = Math.floor(durationMs / numberOfPoints);
+    // Call extractAudioAnalysis with calculated parameters
+    const analysis = await extractAudioAnalysis({
+        fileUri,
+        startTimeMs,
+        endTimeMs,
+        logger,
+        segmentDurationMs,
+        decodingOptions,
+    });
+    // Transform the result into AudioPreview format
+    return analysis;
+}
+//# sourceMappingURL=extractPreview.js.map

package/build/esm/AudioAnalysis/extractPreview.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"extractPreview.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/extractPreview.ts"],"names":[],"mappings":"AACA,OAAO,EAAE,oBAAoB,EAAE,MAAM,wBAAwB,CAAA;AAE7D;;;;;;GAMG;AACH,MAAM,CAAC,KAAK,UAAU,cAAc,CAAC,EACjC,OAAO,EACP,cAAc,GAAG,GAAG,EACpB,WAAW,GAAG,CAAC,EACf,SAAS,GAAG,KAAK,EAAE,mBAAmB;AACtC,eAAe,EACf,MAAM,GACO;IACb,MAAM,UAAU,GAAG,SAAS,GAAG,WAAW,CAAA;IAC1C,MAAM,iBAAiB,GAAG,IAAI,CAAC,KAAK,CAAC,UAAU,GAAG,cAAc,CAAC,CAAA;IAEjE,uDAAuD;IACvD,MAAM,QAAQ,GAAG,MAAM,oBAAoB,CAAC;QACxC,OAAO;QACP,WAAW;QACX,SAAS;QACT,MAAM;QACN,iBAAiB;QACjB,eAAe;KAClB,CAAC,CAAA;IAEF,gDAAgD;IAChD,OAAO,QAAQ,CAAA;AACnB,CAAC","sourcesContent":["import { PreviewOptions, AudioAnalysis } from './AudioAnalysis.types'\nimport { extractAudioAnalysis } from './extractAudioAnalysis'\n\n/**\n * Generates a simplified preview of the audio waveform for quick visualization.\n * Ideal for UI rendering with a specified number of points.\n *\n * @param options - The options for the preview, including file URI and time range.\n * @returns A promise that resolves to the audio preview data.\n */\nexport async function extractPreview({\n    fileUri,\n    numberOfPoints = 100,\n    startTimeMs = 0,\n    endTimeMs = 30000, // First 30 seconds\n    decodingOptions,\n    logger,\n}: PreviewOptions): Promise<AudioAnalysis> {\n    const durationMs = endTimeMs - startTimeMs\n    const segmentDurationMs = Math.floor(durationMs / numberOfPoints)\n\n    // Call extractAudioAnalysis with calculated parameters\n    const analysis = await extractAudioAnalysis({\n        fileUri,\n        startTimeMs,\n        endTimeMs,\n        logger,\n        segmentDurationMs,\n        decodingOptions,\n    })\n\n    // Transform the result into AudioPreview format\n    return analysis\n}\n"]}

package/build/esm/AudioAnalysis/extractWaveform.js

@@ -0,0 +1,11 @@
+import AudioStudioModule from '../AudioStudioModule';
+export const extractWaveform = async ({ fileUri, numberOfSamples, offset = 0, length, }) => {
+    const res = await AudioStudioModule.extractAudioAnalysis({
+        fileUri,
+        numberOfSamples,
+        offset,
+        length,
+    });
+    return res;
+};
+//# sourceMappingURL=extractWaveform.js.map

package/build/esm/AudioAnalysis/extractWaveform.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"extractWaveform.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/extractWaveform.ts"],"names":[],"mappings":"AAAA,OAAO,iBAAiB,MAAM,sBAAsB,CAAA;AAQpD,MAAM,CAAC,MAAM,eAAe,GAAG,KAAK,EAAE,EAClC,OAAO,EACP,eAAe,EACf,MAAM,GAAG,CAAC,EACV,MAAM,GACa,EAAoB,EAAE;IACzC,MAAM,GAAG,GAAG,MAAM,iBAAiB,CAAC,oBAAoB,CAAC;QACrD,OAAO;QACP,eAAe;QACf,MAAM;QACN,MAAM;KACT,CAAC,CAAA;IACF,OAAO,GAAG,CAAA;AACd,CAAC,CAAA","sourcesContent":["import AudioStudioModule from '../AudioStudioModule'\n\nexport interface ExtractWaveformProps {\n    fileUri: string\n    numberOfSamples: number\n    offset?: number\n    length?: number\n}\nexport const extractWaveform = async ({\n    fileUri,\n    numberOfSamples,\n    offset = 0,\n    length,\n}: ExtractWaveformProps): Promise<unknown> => {\n    const res = await AudioStudioModule.extractAudioAnalysis({\n        fileUri,\n        numberOfSamples,\n        offset,\n        length,\n    })\n    return res\n}\n"]}