@siteed/expo-audio-stream 1.4.0 → 1.5.1

package/build/utils/convertPCMToFloat32.d.ts

@@ -1,13 +0,0 @@
-import { ConsoleLike } from '../ExpoAudioStream.types';
-export declare const WAV_HEADER_SIZE = 44;
-export declare const convertPCMToFloat32: ({ bitDepth, buffer, skipWavHeader, logger, }: {
-    buffer: ArrayBuffer;
-    bitDepth: number;
-    skipWavHeader?: boolean;
-    logger?: ConsoleLike;
-}) => Promise<{
-    pcmValues: Float32Array;
-    min: number;
-    max: number;
-}>;
-//# sourceMappingURL=convertPCMToFloat32.d.ts.map

package/build/utils/convertPCMToFloat32.d.ts.map

@@ -1 +0,0 @@
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package/build/utils/convertPCMToFloat32.js

@@ -1,52 +0,0 @@
-import { getWavFileInfo } from './getWavFileInfo';
-export const WAV_HEADER_SIZE = 44;
-const convertSample = (dataView, offset, bitDepth) => {
-    switch (bitDepth) {
-        case 8:
-            return (dataView.getUint8(offset) - 128) / 128;
-        case 16:
-            return dataView.getInt16(offset, true) / 32768;
-        case 24:
-            return (((dataView.getUint8(offset) |
-                (dataView.getUint8(offset + 1) << 8) |
-                (dataView.getUint8(offset + 2) << 16)) /
-                8388608) *
-                2 -
-                1);
-        case 32:
-            return dataView.getFloat32(offset, true);
-        default:
-            throw new Error(`Unsupported bit depth: ${bitDepth}`);
-    }
-};
-export const convertPCMToFloat32 = async ({ bitDepth, buffer, skipWavHeader = false, logger, }) => {
-    try {
-        logger?.debug(`Converting PCM to Float32: bitDepth: ${bitDepth}, buffer.byteLength: ${buffer.byteLength}`);
-        const dataView = new DataView(buffer);
-        let headerOffset = 0;
-        if (skipWavHeader) {
-            const wavFileInfo = await getWavFileInfo(buffer);
-            headerOffset = wavFileInfo.dataChunkOffset;
-        }
-        const dataLength = buffer.byteLength - headerOffset;
-        const sampleLength = Math.floor(dataLength / (bitDepth / 8));
-        const float32Array = new Float32Array(sampleLength);
-        let min = Infinity;
-        let max = -Infinity;
-        for (let i = 0; i < sampleLength; i++) {
-            const offset = headerOffset + i * (bitDepth / 8);
-            const value = convertSample(dataView, offset, bitDepth);
-            if (value < min)
-                min = value;
-            if (value > max)
-                max = value;
-            float32Array[i] = value;
-        }
-        return { pcmValues: float32Array, min, max };
-    }
-    catch (error) {
-        logger?.error(`Error converting PCM to Float32`, error);
-        return { pcmValues: new Float32Array(), min: 0, max: 0 };
-    }
-};
-//# sourceMappingURL=convertPCMToFloat32.js.map

package/build/utils/convertPCMToFloat32.js.map

@@ -1 +0,0 @@
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package/build/utils/encodingToBitDepth.d.ts

@@ -1,5 +0,0 @@
-import { BitDepth, EncodingType } from '../ExpoAudioStream.types';
-export declare const encodingToBitDepth: ({ encoding, }: {
-    encoding: EncodingType;
-}) => BitDepth;
-//# sourceMappingURL=encodingToBitDepth.d.ts.map

package/build/utils/encodingToBitDepth.d.ts.map

@@ -1 +0,0 @@
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package/build/utils/encodingToBitDepth.js

@@ -1,13 +0,0 @@
-export const encodingToBitDepth = ({ encoding, }) => {
-    switch (encoding) {
-        case 'pcm_32bit':
-            return 32;
-        case 'pcm_16bit':
-            return 16;
-        case 'pcm_8bit':
-            return 8;
-        default:
-            throw new Error(`Unsupported encoding type: ${encoding}`);
-    }
-};
-//# sourceMappingURL=encodingToBitDepth.js.map

package/build/utils/encodingToBitDepth.js.map

@@ -1 +0,0 @@
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package/build/utils/getWavFileInfo.d.ts

@@ -1,26 +0,0 @@
-import { BitDepth, SampleRate } from '../ExpoAudioStream.types';
-/**
- * Interface representing the metadata of a WAV file.
- */
-export interface WavFileInfo {
-    sampleRate: SampleRate;
-    numChannels: number;
-    bitDepth: BitDepth;
-    size: number;
-    durationMs: number;
-    audioFormatDescription: string;
-    byteRate: number;
-    blockAlign: number;
-    creationDateTime?: string;
-    comments?: string;
-    compressionType?: string;
-    dataChunkOffset: number;
-}
-/**
- * Extracts metadata from a WAV buffer.
- *
- * @param arrayBuffer - The array buffer containing the WAV data.
- * @returns A promise that resolves to the extracted metadata.
- */
-export declare const getWavFileInfo: (arrayBuffer: ArrayBuffer) => Promise<WavFileInfo>;
-//# sourceMappingURL=getWavFileInfo.d.ts.map

package/build/utils/getWavFileInfo.d.ts.map

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package/build/utils/getWavFileInfo.js

@@ -1,92 +0,0 @@
-// packages/expo-audio-stream/src/utils/getWavFileInfo.ts
-import { DATA_CHUNK_ID, DEFAULT_BIT_DEPTH, DEFAULT_SAMPLE_RATE, FMT_CHUNK_ID, INFO_CHUNK_ID, RIFF_HEADER, WAVE_HEADER, } from '../constants';
-// Audio format descriptions
-const AUDIO_FORMATS = {
-    1: 'PCM',
-    3: 'IEEE float',
-    6: '8-bit ITU-T G.711 A-law',
-    7: '8-bit ITU-T G.711 µ-law',
-    65534: 'WAVE_FORMAT_EXTENSIBLE',
-};
-/**
- * Extracts metadata from a WAV buffer.
- *
- * @param arrayBuffer - The array buffer containing the WAV data.
- * @returns A promise that resolves to the extracted metadata.
- */
-export const getWavFileInfo = async (arrayBuffer) => {
-    const view = new DataView(arrayBuffer);
-    // Check if the file is a valid RIFF/WAVE file
-    const riffHeader = view.getUint32(0, false);
-    const waveHeader = view.getUint32(8, false);
-    if (riffHeader !== RIFF_HEADER || waveHeader !== WAVE_HEADER) {
-        throw new Error('Invalid WAV file');
-    }
-    // Initialize variables for the metadata
-    let fmtChunkOffset = 12;
-    let sampleRate = DEFAULT_SAMPLE_RATE;
-    let numChannels = 0;
-    let bitDepth = DEFAULT_BIT_DEPTH;
-    let dataChunkSize = 0;
-    let audioFormat = 0;
-    let byteRate = 0;
-    let blockAlign = 0;
-    let creationDateTime = '';
-    let comments = '';
-    let dataChunkOffset = 0;
-    // Parse chunks to find the "fmt " and "data" chunks
-    while (fmtChunkOffset < view.byteLength) {
-        const chunkId = view.getUint32(fmtChunkOffset, false);
-        const chunkSize = view.getUint32(fmtChunkOffset + 4, true);
-        if (chunkId === FMT_CHUNK_ID) {
-            // "fmt "
-            audioFormat = view.getUint16(fmtChunkOffset + 8, true);
-            if (!AUDIO_FORMATS[audioFormat]) {
-                throw new Error('Unsupported WAV file format');
-            }
-            numChannels = view.getUint16(fmtChunkOffset + 10, true);
-            sampleRate = view.getUint32(fmtChunkOffset + 12, true);
-            byteRate = view.getUint32(fmtChunkOffset + 16, true);
-            blockAlign = view.getUint16(fmtChunkOffset + 20, true);
-            bitDepth = view.getUint16(fmtChunkOffset + 22, true);
-        }
-        else if (chunkId === DATA_CHUNK_ID) {
-            // "data"
-            dataChunkSize = chunkSize;
-            dataChunkOffset = fmtChunkOffset + 8; // Position after chunk header
-            break;
-        }
-        else if (chunkId === INFO_CHUNK_ID) {
-            // "INFO"
-            // Read INFO chunk (assuming it contains a text-based creation date/time and comments)
-            const infoStart = fmtChunkOffset + 8;
-            const infoText = new TextDecoder().decode(new Uint8Array(arrayBuffer.slice(infoStart, infoStart + chunkSize)));
-            const infoParts = infoText.split('\0');
-            creationDateTime = infoParts[0];
-            comments = infoParts[1];
-        }
-        fmtChunkOffset += 8 + chunkSize;
-    }
-    if (!sampleRate || !numChannels || !bitDepth || !dataChunkSize) {
-        throw new Error('Incomplete WAV file information');
-    }
-    // Calculate duration
-    const bytesPerSample = bitDepth / 8;
-    const numSamples = dataChunkSize / (numChannels * bytesPerSample);
-    const durationMs = (numSamples / sampleRate) * 1000;
-    return {
-        sampleRate,
-        numChannels,
-        bitDepth,
-        size: arrayBuffer.byteLength,
-        durationMs,
-        audioFormatDescription: AUDIO_FORMATS[audioFormat],
-        byteRate,
-        blockAlign,
-        creationDateTime: creationDateTime || undefined,
-        comments: comments || undefined,
-        compressionType: audioFormat === 1 ? 'None' : AUDIO_FORMATS[audioFormat],
-        dataChunkOffset,
-    };
-};
-//# sourceMappingURL=getWavFileInfo.js.map

package/build/utils/getWavFileInfo.js.map

@@ -1 +0,0 @@
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The array buffer containing the WAV data.\n * @returns A promise that resolves to the extracted metadata.\n */\nexport const getWavFileInfo = async (\n    arrayBuffer: ArrayBuffer\n): Promise<WavFileInfo> => {\n    const view = new DataView(arrayBuffer)\n\n    // Check if the file is a valid RIFF/WAVE file\n    const riffHeader = view.getUint32(0, false)\n    const waveHeader = view.getUint32(8, false)\n    if (riffHeader !== RIFF_HEADER || waveHeader !== WAVE_HEADER) {\n        throw new Error('Invalid WAV file')\n    }\n\n    // Initialize variables for the metadata\n    let fmtChunkOffset = 12\n    let sampleRate: SampleRate = DEFAULT_SAMPLE_RATE\n    let numChannels = 0\n    let bitDepth: BitDepth = DEFAULT_BIT_DEPTH\n    let dataChunkSize = 0\n    let audioFormat = 0\n    let byteRate = 0\n    let blockAlign = 0\n    let creationDateTime = ''\n    let comments = ''\n    let dataChunkOffset = 0\n\n    // Parse chunks to find the \"fmt \" and \"data\" chunks\n    while (fmtChunkOffset < view.byteLength) {\n        const chunkId = view.getUint32(fmtChunkOffset, false)\n        const chunkSize = view.getUint32(fmtChunkOffset + 4, true)\n        if (chunkId === FMT_CHUNK_ID) {\n            // \"fmt \"\n            audioFormat = view.getUint16(fmtChunkOffset + 8, true)\n            if (!AUDIO_FORMATS[audioFormat]) {\n                throw new Error('Unsupported WAV file format')\n            }\n            numChannels = view.getUint16(fmtChunkOffset + 10, true)\n            sampleRate = view.getUint32(fmtChunkOffset + 12, true) as SampleRate\n            byteRate = view.getUint32(fmtChunkOffset + 16, true)\n            blockAlign = view.getUint16(fmtChunkOffset + 20, true)\n            bitDepth = view.getUint16(fmtChunkOffset + 22, true) as BitDepth\n        } else if (chunkId === DATA_CHUNK_ID) {\n            // \"data\"\n            dataChunkSize = chunkSize\n            dataChunkOffset = fmtChunkOffset + 8 // Position after chunk header\n            break\n        } else if (chunkId === INFO_CHUNK_ID) {\n            // \"INFO\"\n            // Read INFO chunk (assuming it contains a text-based creation date/time and comments)\n            const infoStart = fmtChunkOffset + 8\n            const infoText = new TextDecoder().decode(\n                new Uint8Array(\n                    arrayBuffer.slice(infoStart, infoStart + chunkSize)\n                )\n            )\n            const infoParts = infoText.split('\\0')\n            creationDateTime = infoParts[0]\n            comments = infoParts[1]\n        }\n        fmtChunkOffset += 8 + chunkSize\n    }\n\n    if (!sampleRate || !numChannels || !bitDepth || !dataChunkSize) {\n        throw new Error('Incomplete WAV file information')\n    }\n\n    // Calculate duration\n    const bytesPerSample = bitDepth / 8\n    const numSamples = dataChunkSize / (numChannels * bytesPerSample)\n    const durationMs = (numSamples / sampleRate) * 1000\n\n    return {\n        sampleRate,\n        numChannels,\n        bitDepth,\n        size: arrayBuffer.byteLength,\n        durationMs,\n        audioFormatDescription: AUDIO_FORMATS[audioFormat],\n        byteRate,\n        blockAlign,\n        creationDateTime: creationDateTime || undefined,\n        comments: comments || undefined,\n        compressionType:\n            audioFormat === 1 ? 'None' : AUDIO_FORMATS[audioFormat],\n        dataChunkOffset,\n    }\n}\n"]}

package/build/utils/writeWavHeader.d.ts

@@ -1,49 +0,0 @@
-/**
- * Options for creating a WAV header.
- */
-export interface WavHeaderOptions {
-    /** Optional buffer containing audio data. If provided, it will be combined with the header. */
-    buffer?: ArrayBuffer;
-    /** The sample rate of the audio in Hz (e.g., 44100). */
-    sampleRate: number;
-    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */
-    numChannels: number;
-    /** The bit depth of the audio (e.g., 16, 24, or 32). */
-    bitDepth: number;
-}
-/**
- * Writes or updates a WAV (RIFF) header based on the provided options.
- *
- * This function can be used in three ways:
- * 1. To create a standalone WAV header (when no buffer is provided).
- * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).
- * 3. To update an existing WAV header in the provided buffer.
- *
- * For streaming audio where the final size is unknown, this function sets the size fields
- * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the
- * `updateWavHeaderSize` function once the final size is known.
- *
- * @param options - The options for creating or updating the WAV header.
- * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
- *
- * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
- *
- * @example
- * // Create a standalone WAV header
- * const header = writeWavHeader({
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- *
- * @example
- * // Create a WAV header and combine it with audio data
- * const completeWav = writeWavHeader({
- *   buffer: audioData,
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- */
-export declare const writeWavHeader: ({ buffer, sampleRate, numChannels, bitDepth, }: WavHeaderOptions) => ArrayBuffer;
-//# sourceMappingURL=writeWavHeader.d.ts.map

package/build/utils/writeWavHeader.d.ts.map

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package/build/utils/writeWavHeader.js

@@ -1,91 +0,0 @@
-// packages/expo-audio-stream/src/utils/writeWavHeader.ts
-/**
- * Writes or updates a WAV (RIFF) header based on the provided options.
- *
- * This function can be used in three ways:
- * 1. To create a standalone WAV header (when no buffer is provided).
- * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).
- * 3. To update an existing WAV header in the provided buffer.
- *
- * For streaming audio where the final size is unknown, this function sets the size fields
- * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the
- * `updateWavHeaderSize` function once the final size is known.
- *
- * @param options - The options for creating or updating the WAV header.
- * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
- *
- * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
- *
- * @example
- * // Create a standalone WAV header
- * const header = writeWavHeader({
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- *
- * @example
- * // Create a WAV header and combine it with audio data
- * const completeWav = writeWavHeader({
- *   buffer: audioData,
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- */
-export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) => {
-    const bytesPerSample = bitDepth / 8;
-    const blockAlign = numChannels * bytesPerSample;
-    const byteRate = sampleRate * blockAlign;
-    // Function to write a string to the DataView
-    const writeString = (view, offset, string) => {
-        for (let i = 0; i < string.length; i++) {
-            view.setUint8(offset + i, string.charCodeAt(i));
-        }
-    };
-    // Function to write or update the header
-    const writeHeader = (view, dataSize = 0xffffffff) => {
-        writeString(view, 0, 'RIFF'); // ChunkID
-        view.setUint32(4, 36 + dataSize, true); // ChunkSize
-        writeString(view, 8, 'WAVE'); // Format
-        writeString(view, 12, 'fmt '); // Subchunk1ID
-        view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)
-        view.setUint16(20, bitDepth === 32 ? 3 : 1, true); // AudioFormat (3 for float, 1 for PCM)
-        view.setUint16(22, numChannels, true); // NumChannels
-        view.setUint32(24, sampleRate, true); // SampleRate
-        view.setUint32(28, byteRate, true); // ByteRate
-        view.setUint16(32, blockAlign, true); // BlockAlign
-        view.setUint16(34, bitDepth, true); // BitsPerSample
-        writeString(view, 36, 'data'); // Subchunk2ID
-        view.setUint32(40, dataSize, true); // Subchunk2Size
-    };
-    if (buffer) {
-        if (buffer.byteLength < 44) {
-            throw new Error('Buffer is too small to contain a valid WAV header');
-        }
-        const view = new DataView(buffer);
-        // Check if the buffer already has a WAV header by looking for "RIFF" at the start
-        const existingHeader = view.getUint32(0, false) === 0x52494646; // "RIFF" in ASCII
-        if (existingHeader) {
-            // Update the existing header
-            writeHeader(view, buffer.byteLength - 44);
-            return buffer;
-        }
-        else {
-            // Combine the new header with the existing buffer
-            const newBuffer = new ArrayBuffer(44 + buffer.byteLength);
-            const newView = new DataView(newBuffer);
-            writeHeader(newView, buffer.byteLength);
-            new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44);
-            return newBuffer;
-        }
-    }
-    else {
-        // Create a standalone header
-        const headerBuffer = new ArrayBuffer(44);
-        const view = new DataView(headerBuffer);
-        writeHeader(view);
-        return headerBuffer;
-    }
-};
-//# sourceMappingURL=writeWavHeader.js.map

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If provided, it will be combined with the header. */\n    buffer?: ArrayBuffer\n    /** The sample rate of the audio in Hz (e.g., 44100). */\n    sampleRate: number\n    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */\n    numChannels: number\n    /** The bit depth of the audio (e.g., 16, 24, or 32). */\n    bitDepth: number\n}\n\n/**\n * Writes or updates a WAV (RIFF) header based on the provided options.\n *\n * This function can be used in three ways:\n * 1. To create a standalone WAV header (when no buffer is provided).\n * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).\n * 3. To update an existing WAV header in the provided buffer.\n *\n * For streaming audio where the final size is unknown, this function sets the size fields\n * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the\n * `updateWavHeaderSize` function once the final size is known.\n *\n * @param options - The options for creating or updating the WAV header.\n * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.\n *\n * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.\n *\n * @example\n * // Create a standalone WAV header\n * const header = writeWavHeader({\n *   sampleRate: 44100,\n *   numChannels: 2,\n *   bitDepth: 16\n * });\n *\n * @example\n * // Create a WAV header and combine it with audio data\n * const completeWav = writeWavHeader({\n *   buffer: audioData,\n *   sampleRate: 44100,\n *   numChannels: 2,\n *   bitDepth: 16\n * });\n */\nexport const writeWavHeader = ({\n    buffer,\n    sampleRate,\n    numChannels,\n    bitDepth,\n}: WavHeaderOptions): ArrayBuffer => {\n    const bytesPerSample = bitDepth / 8\n    const blockAlign = numChannels * bytesPerSample\n    const byteRate = sampleRate * blockAlign\n\n    // Function to write a string to the DataView\n    const writeString = (view: DataView, offset: number, string: string) => {\n        for (let i = 0; i < string.length; i++) {\n            view.setUint8(offset + i, string.charCodeAt(i))\n        }\n    }\n\n    // Function to write or update the header\n    const writeHeader = (view: DataView, dataSize: number = 0xffffffff) => {\n        writeString(view, 0, 'RIFF') // ChunkID\n        view.setUint32(4, 36 + dataSize, true) // ChunkSize\n        writeString(view, 8, 'WAVE') // Format\n        writeString(view, 12, 'fmt ') // Subchunk1ID\n        view.setUint32(16, 16, true) // Subchunk1Size (16 for PCM)\n        view.setUint16(20, bitDepth === 32 ? 3 : 1, true) // AudioFormat (3 for float, 1 for PCM)\n        view.setUint16(22, numChannels, true) // NumChannels\n        view.setUint32(24, sampleRate, true) // SampleRate\n        view.setUint32(28, byteRate, true) // ByteRate\n        view.setUint16(32, blockAlign, true) // BlockAlign\n        view.setUint16(34, bitDepth, true) // BitsPerSample\n        writeString(view, 36, 'data') // Subchunk2ID\n        view.setUint32(40, dataSize, true) // Subchunk2Size\n    }\n\n    if (buffer) {\n        if (buffer.byteLength < 44) {\n            throw new Error('Buffer is too small to contain a valid WAV header')\n        }\n\n        const view = new DataView(buffer)\n\n        // Check if the buffer already has a WAV header by looking for \"RIFF\" at the start\n        const existingHeader = view.getUint32(0, false) === 0x52494646 // \"RIFF\" in ASCII\n\n        if (existingHeader) {\n            // Update the existing header\n            writeHeader(view, buffer.byteLength - 44)\n            return buffer\n        } else {\n            // Combine the new header with the existing buffer\n            const newBuffer = new ArrayBuffer(44 + buffer.byteLength)\n            const newView = new DataView(newBuffer)\n            writeHeader(newView, buffer.byteLength)\n            new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44)\n            return newBuffer\n        }\n    } else {\n        // Create a standalone header\n        const headerBuffer = new ArrayBuffer(44)\n        const view = new DataView(headerBuffer)\n        writeHeader(view)\n        return headerBuffer\n    }\n}\n"]}

package/build/workers/InlineFeaturesExtractor.web.d.ts

@@ -1,2 +0,0 @@
-export declare const InlineFeaturesExtractor = "\n// Unique ID counter\nlet uniqueIdCounter = 0\n\nself.onmessage = function (event) {\n    const {\n        channelData, // this is only the newly recorded data when live recording.\n        sampleRate,\n        pointsPerSecond,\n        algorithm,\n        bitDepth,\n        fullAudioDurationMs,\n        numberOfChannels,\n        features: _features,\n    } = event.data\n    const features = _features || {}\n\n    const SILENCE_THRESHOLD = 0.01\n    const MIN_SILENCE_DURATION = 1.5 * sampleRate // 1.5 seconds of silence\n    const SPEECH_INERTIA_DURATION = 0.1 * sampleRate // Speech inertia duration in samples\n    const RMS_THRESHOLD = 0.01\n    const ZCR_THRESHOLD = 0.1\n\n    // Placeholder functions for feature extraction\n    const extractMFCC = (segmentData, sampleRate) => {\n        // Implement MFCC extraction logic here\n        return []\n    }\n\n    const extractSpectralCentroid = (segmentData, sampleRate) => {\n        const magnitudeSpectrum = segmentData.map((v) => v * v)\n        const sum = magnitudeSpectrum.reduce((a, b) => a + b, 0)\n        if (sum === 0) return 0\n\n        const weightedSum = magnitudeSpectrum.reduce(\n            (acc, value, index) => acc + index * value,\n            0\n        )\n        return (\n            ((weightedSum / sum) * (sampleRate / 2)) / magnitudeSpectrum.length\n        )\n    }\n\n    const extractSpectralFlatness = (segmentData) => {\n        const magnitudeSpectrum = segmentData.map((v) => Math.abs(v))\n        const geometricMean = Math.exp(\n            magnitudeSpectrum\n                .map((v) => Math.log(v + Number.MIN_VALUE))\n                .reduce((a, b) => a + b) / magnitudeSpectrum.length\n        )\n        const arithmeticMean =\n            magnitudeSpectrum.reduce((a, b) => a + b) / magnitudeSpectrum.length\n        return arithmeticMean === 0 ? 0 : geometricMean / arithmeticMean\n    }\n\n    const extractSpectralRollOff = (segmentData, sampleRate) => {\n        const magnitudeSpectrum = segmentData.map((v) => Math.abs(v))\n        const totalEnergy = magnitudeSpectrum.reduce((a, b) => a + b, 0)\n        const rollOffThreshold = totalEnergy * 0.85\n        let cumulativeEnergy = 0\n\n        for (let i = 0; i < magnitudeSpectrum.length; i++) {\n            cumulativeEnergy += magnitudeSpectrum[i]\n            if (cumulativeEnergy >= rollOffThreshold) {\n                return (i / magnitudeSpectrum.length) * (sampleRate / 2)\n            }\n        }\n\n        return 0\n    }\n\n    const extractSpectralBandwidth = (segmentData, sampleRate) => {\n        const centroid = extractSpectralCentroid(segmentData, sampleRate)\n        const magnitudeSpectrum = segmentData.map((v) => Math.abs(v))\n        const sum = magnitudeSpectrum.reduce((a, b) => a + b, 0)\n        if (sum === 0) return 0\n\n        const weightedSum = magnitudeSpectrum.reduce(\n            (acc, value, index) => acc + value * Math.pow(index - centroid, 2),\n            0\n        )\n        return Math.sqrt(weightedSum / sum)\n    }\n\n    const extractChromagram = (segmentData, sampleRate) => {\n        return [] // TODO implement\n    }\n\n    const extractHNR = (segmentData) => {\n        const frameSize = segmentData.length\n        const autocorrelation = new Float32Array(frameSize)\n\n        // Compute the autocorrelation of the segment data\n        for (let i = 0; i < frameSize; i++) {\n            let sum = 0\n            for (let j = 0; j < frameSize - i; j++) {\n                sum += segmentData[j] * segmentData[j + i]\n            }\n            autocorrelation[i] = sum\n        }\n\n        // Find the maximum autocorrelation value (excluding the zero lag)\n        const maxAutocorrelation = Math.max(...autocorrelation.subarray(1))\n\n        // Compute the HNR\n        return autocorrelation[0] !== 0\n            ? 10 *\n                  Math.log10(\n                      maxAutocorrelation /\n                          (autocorrelation[0] - maxAutocorrelation)\n                  )\n            : 0\n    }\n\n    const extractWaveform = (\n        channelData, // Float32Array\n        sampleRate, // number\n        pointsPerSecond, // number\n        algorithm // string\n    ) => {\n        const totalSamples = channelData.length\n        const segmentDuration = totalSamples / sampleRate\n        const totalPoints = Math.max(\n            Math.ceil(segmentDuration * pointsPerSecond),\n            1\n        )\n        const pointInterval = Math.ceil(totalSamples / totalPoints)\n        const dataPoints = []\n        let minAmplitude = Infinity\n        let maxAmplitude = -Infinity\n        let silenceStart = null\n        let lastSpeechEnd = -Infinity\n        let isSpeech = false\n\n        const expectedPoints = segmentDuration * pointsPerSecond\n        const samplesPerPoint = Math.ceil(channelData.length / expectedPoints)\n\n        for (let i = 0; i < expectedPoints; i++) {\n            const start = i * samplesPerPoint\n            const end = Math.min(start + samplesPerPoint, totalSamples)\n\n            let sumSquares = 0\n            let zeroCrossings = 0\n            let prevValue = channelData[start]\n            let localMinAmplitude = Infinity\n            let localMaxAmplitude = -Infinity\n            let hasNonZeroValue = false\n\n            // compute values for the segment\n            for (let j = start; j < end; j++) {\n                const value = channelData[j]\n                sumSquares += value * value\n                if (j > start && value * prevValue < 0) {\n                    zeroCrossings++\n                }\n                prevValue = value\n\n                // We need to keep absolute value otherwise we cannot visualize properly\n                const absValue = Math.abs(value)\n                localMinAmplitude = Math.min(localMinAmplitude, absValue)\n                localMaxAmplitude = Math.max(localMaxAmplitude, absValue)\n\n                if (value !== 0) {\n                    hasNonZeroValue = true\n                }\n            }\n\n            // Post-processing checks\n            if (!hasNonZeroValue) {\n                // All values are zero\n                localMinAmplitude = 0\n                localMaxAmplitude = 0\n            }\n\n            const rms = Math.sqrt(sumSquares / (end - start))\n            minAmplitude = Math.min(minAmplitude, localMinAmplitude)\n            maxAmplitude = Math.max(maxAmplitude, localMaxAmplitude)\n\n            const energy = sumSquares\n            const zcr = zeroCrossings / (end - start)\n\n            const silent = rms < SILENCE_THRESHOLD\n            const dB = 20 * Math.log10(rms)\n\n            if (silent) {\n                if (silenceStart === null) {\n                    silenceStart = start\n                } else if (start - silenceStart > MIN_SILENCE_DURATION) {\n                    // Silence detected for longer than the threshold, set amplitude to 0\n                    localMaxAmplitude = 0\n                    localMinAmplitude = 0\n                    isSpeech = false\n                }\n            } else {\n                silenceStart = null\n                if (\n                    !isSpeech &&\n                    start - lastSpeechEnd < SPEECH_INERTIA_DURATION\n                ) {\n                    isSpeech = true\n                }\n                lastSpeechEnd = end\n            }\n\n            const activeSpeech =\n                (rms > RMS_THRESHOLD && zcr > ZCR_THRESHOLD) ||\n                (isSpeech && start - lastSpeechEnd < SPEECH_INERTIA_DURATION)\n\n            if (activeSpeech) {\n                isSpeech = true\n                lastSpeechEnd = end\n            } else {\n                isSpeech = false\n            }\n\n            const bytesPerSample = bitDepth / 8\n            const startPosition = start * bytesPerSample * numberOfChannels // Calculate start position in bytes\n            const endPosition = end * bytesPerSample * numberOfChannels // Calculate end position in bytes\n\n            // Compute features\n            const segmentData = channelData.slice(start, end)\n            const mfcc = features.mfcc\n                ? extractMFCC(segmentData, sampleRate)\n                : []\n            const spectralCentroid = features.spectralCentroid\n                ? extractSpectralCentroid(segmentData, sampleRate)\n                : 0\n            const spectralFlatness = features.spectralFlatness\n                ? extractSpectralFlatness(segmentData)\n                : 0\n            const spectralRollOff = features.spectralRollOff\n                ? extractSpectralRollOff(segmentData, sampleRate)\n                : 0\n            const spectralBandwidth = features.spectralBandwidth\n                ? extractSpectralBandwidth(segmentData, sampleRate)\n                : 0\n            const chromagram = features.chromagram\n                ? extractChromagram(segmentData, sampleRate)\n                : []\n            const hnr = features.hnr ? extractHNR(segmentData) : 0\n\n            const peakAmp = Math.max(Math.abs(localMaxAmplitude), Math.abs(localMinAmplitude))\n            const newData = {\n                id: uniqueIdCounter++, // Assign unique ID and increment the counter\n                amplitude: algorithm === 'peak' ? peakAmp : rms,\n                activeSpeech,\n                dB,\n                silent,\n                features: {\n                    energy,\n                    rms,\n                    minAmplitude: localMinAmplitude,\n                    maxAmplitude: localMaxAmplitude,\n                    zcr,\n                    mfcc: [], // Placeholder for MFCC features\n                    spectralCentroid, // Computed spectral centroid\n                    spectralFlatness, // Computed spectral flatness\n                    spectralRollOff, // Computed spectral roll-off\n                    spectralBandwidth, // Computed spectral bandwidth\n                    chromagram, // Computed chromagram\n                    hnr, // Computed HNR\n                },\n                startTime: start / sampleRate,\n                endTime: end / sampleRate,\n                startPosition,\n                endPosition,\n                samples: end - start,\n                speaker: 0, // Assuming speaker detection is to be handled later\n            }\n\n            dataPoints.push(newData)\n        }\n\n        return {\n            pointsPerSecond,\n            amplitudeAlgorithm: algorithm,\n            durationMs: fullAudioDurationMs,\n            bitDepth,\n            samples: totalSamples,\n            numberOfChannels,\n            sampleRate,\n            dataPoints,\n            amplitudeRange: {\n                min: minAmplitude,\n                max: maxAmplitude,\n            },\n            speakerChanges: [], // Placeholder for future speaker detection logic\n        }\n    }\n\n    try {\n        const result = extractWaveform(\n            channelData,\n            sampleRate,\n            pointsPerSecond,\n            algorithm\n        )\n        self.postMessage({\n            command: 'features',\n            result,\n        })\n    } catch (error) {\n        console.error('[AudioFeaturesExtractor] Error in processing', error)\n        self.postMessage({ error: error.message })\n    } finally {\n        // Do not close the worker so it can be re-used for subsequent messages\n        // self.close();\n    }\n}\n";
-//# sourceMappingURL=InlineFeaturesExtractor.web.d.ts.map

package/build/workers/InlineFeaturesExtractor.web.d.ts.map

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