@siteed/expo-audio-stream 1.0.3 → 1.0.5

package/build/utils/convertPCMToFloat32.d.ts

@@ -3,9 +3,9 @@ export declare const convertPCMToFloat32: ({ bitDepth, buffer, skipWavHeader, }:
     buffer: ArrayBuffer;
     bitDepth: number;
     skipWavHeader?: boolean;
-}) => {
+}) => Promise<{
     pcmValues: Float32Array;
     min: number;
     max: number;
-};
+}>;
 //# sourceMappingURL=convertPCMToFloat32.d.ts.map

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

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

@@ -1,41 +1,54 @@
+import { getWavFileInfo } from './getWavFileInfo';
+import { getLogger } from '../logger';
 export const WAV_HEADER_SIZE = 44;
-export const convertPCMToFloat32 = ({ bitDepth, buffer, skipWavHeader = false, }) => {
-    const dataView = new DataView(buffer);
-    const headerOffset = skipWavHeader ? WAV_HEADER_SIZE : 0;
-    const dataLength = buffer.byteLength - headerOffset;
-    const sampleLength = dataLength / (bitDepth / 8);
-    const float32Array = new Float32Array(sampleLength);
-    let min = Infinity;
-    let max = -Infinity;
-    for (let i = 0; i < sampleLength; i++) {
-        let value = 0;
-        const offset = headerOffset + i * (bitDepth / 8);
-        switch (bitDepth) {
-            case 8:
-                value = dataView.getUint8(offset) / 128;
-                break;
-            case 16:
-                value = dataView.getInt16(offset, true) / 32768;
-                break;
-            case 24:
-                value =
-                    (dataView.getUint8(offset) +
-                        (dataView.getUint8(offset + 1) << 8) +
-                        (dataView.getUint8(offset + 2) << 16)) /
-                        8388608;
-                break;
-            case 32:
-                value = dataView.getFloat32(offset, true);
-                break;
-            default:
-                throw new Error(`Unsupported bit depth: ${bitDepth}`);
+const logger = getLogger('convertPCMToFloat32');
+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, }) => {
+    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;
         }
-        if (value < min)
-            min = value;
-        if (value > max)
-            max = value;
-        float32Array[i] = value;
+        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 };
     }
-    return { pcmValues: float32Array, min, max };
 };
 //# sourceMappingURL=convertPCMToFloat32.js.map

package/build/utils/convertPCMToFloat32.js.map

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

@@ -1,4 +1,4 @@
-import { BitDepth, EncodingType } from "../ExpoAudioStream.types";
+import { BitDepth, EncodingType } from '../ExpoAudioStream.types';
 export declare const encodingToBitDepth: ({ encoding, }: {
     encoding: EncodingType;
 }) => BitDepth;

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

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

@@ -1,10 +1,10 @@
 export const encodingToBitDepth = ({ encoding, }) => {
     switch (encoding) {
-        case "pcm_32bit":
+        case 'pcm_32bit':
             return 32;
-        case "pcm_16bit":
+        case 'pcm_16bit':
             return 16;
-        case "pcm_8bit":
+        case 'pcm_8bit':
             return 8;
         default:
             throw new Error(`Unsupported encoding type: ${encoding}`);

package/build/utils/encodingToBitDepth.js.map

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

@@ -1,4 +1,4 @@
-import { BitDepth, SampleRate } from "../ExpoAudioStream.types";
+import { BitDepth, SampleRate } from '../ExpoAudioStream.types';
 /**
  * Interface representing the metadata of a WAV file.
  */
@@ -14,11 +14,12 @@ export interface WavFileInfo {
     creationDateTime?: string;
     comments?: string;
     compressionType?: string;
+    dataChunkOffset: number;
 }
 /**
- * Extracts metadata from a WAV file.
+ * Extracts metadata from a WAV buffer.
  *
- * @param arrayBuffer - The array buffer containing the WAV file data.
+ * @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>;

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

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package/build/utils/getWavFileInfo.js

@@ -1,17 +1,17 @@
 // 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";
+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",
+    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 file.
+ * Extracts metadata from a WAV buffer.
  *
- * @param arrayBuffer - The array buffer containing the WAV file data.
+ * @param arrayBuffer - The array buffer containing the WAV data.
  * @returns A promise that resolves to the extracted metadata.
  */
 export const getWavFileInfo = async (arrayBuffer) => {
@@ -20,7 +20,7 @@ export const getWavFileInfo = async (arrayBuffer) => {
     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");
+        throw new Error('Invalid WAV file');
     }
     // Initialize variables for the metadata
     let fmtChunkOffset = 12;
@@ -31,8 +31,9 @@ export const getWavFileInfo = async (arrayBuffer) => {
     let audioFormat = 0;
     let byteRate = 0;
     let blockAlign = 0;
-    let creationDateTime = "";
-    let comments = "";
+    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);
@@ -41,7 +42,7 @@ export const getWavFileInfo = async (arrayBuffer) => {
             // "fmt "
             audioFormat = view.getUint16(fmtChunkOffset + 8, true);
             if (!AUDIO_FORMATS[audioFormat]) {
-                throw new Error("Unsupported WAV file format");
+                throw new Error('Unsupported WAV file format');
             }
             numChannels = view.getUint16(fmtChunkOffset + 10, true);
             sampleRate = view.getUint32(fmtChunkOffset + 12, true);
@@ -52,6 +53,7 @@ export const getWavFileInfo = async (arrayBuffer) => {
         else if (chunkId === DATA_CHUNK_ID) {
             // "data"
             dataChunkSize = chunkSize;
+            dataChunkOffset = fmtChunkOffset + 8; // Position after chunk header
             break;
         }
         else if (chunkId === INFO_CHUNK_ID) {
@@ -59,14 +61,14 @@ export const getWavFileInfo = async (arrayBuffer) => {
             // 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");
+            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");
+        throw new Error('Incomplete WAV file information');
     }
     // Calculate duration
     const bytesPerSample = bitDepth / 8;
@@ -83,7 +85,8 @@ export const getWavFileInfo = async (arrayBuffer) => {
         blockAlign,
         creationDateTime: creationDateTime || undefined,
         comments: comments || undefined,
-        compressionType: audioFormat === 1 ? "None" : AUDIO_FORMATS[audioFormat],
+        compressionType: audioFormat === 1 ? 'None' : AUDIO_FORMATS[audioFormat],
+        dataChunkOffset,
     };
 };
 //# sourceMappingURL=getWavFileInfo.js.map

package/build/utils/getWavFileInfo.js.map

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// in bytes\n  durationMs: number; // in ms\n  audioFormatDescription: string; // Description of the audio format\n  byteRate: number; // Average bytes per second\n  blockAlign: number; // Number of bytes for one sample including all channels\n  creationDateTime?: string; // Optional creation date and time\n  comments?: string; // Optional comments or tags\n  compressionType?: string; // Optional compression type\n}\n\n/**\n * Extracts metadata from a WAV file.\n *\n * @param arrayBuffer - The array buffer containing the WAV file 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\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      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(arrayBuffer.slice(infoStart, infoStart + chunkSize)),\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: audioFormat === 1 ? \"None\" : AUDIO_FORMATS[audioFormat],\n  };\n};\n"]}
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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.map

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+{"version":3,"file":"writeWavHeader.d.ts","sourceRoot":"","sources":["../../src/utils/writeWavHeader.ts"],"names":[],"mappings":"AACA,UAAU,gBAAgB;IACtB,MAAM,EAAE,WAAW,CAAA;IACnB,UAAU,EAAE,MAAM,CAAA;IAClB,WAAW,EAAE,MAAM,CAAA;IACnB,QAAQ,EAAE,MAAM,CAAA;CACnB;AAED,eAAO,MAAM,cAAc,mDAKxB,gBAAgB,KAAG,WA0CrB,CAAA"}

package/build/utils/writeWavHeader.js

@@ -14,10 +14,10 @@ export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) =
     const existingHeader = view.getUint32(0, false) === 0x52494646; // "RIFF" in ASCII
     if (!existingHeader) {
         // Write the WAV header
-        writeString(view, 0, "RIFF"); // ChunkID
+        writeString(view, 0, 'RIFF'); // ChunkID
         view.setUint32(4, 36 + numSamples * blockAlign, true); // ChunkSize
-        writeString(view, 8, "WAVE"); // Format
-        writeString(view, 12, "fmt "); // Subchunk1ID
+        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
@@ -25,7 +25,7 @@ export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) =
         view.setUint32(28, byteRate, true); // ByteRate
         view.setUint16(32, blockAlign, true); // BlockAlign
         view.setUint16(34, bitDepth, true); // BitsPerSample
-        writeString(view, 36, "data"); // Subchunk2ID
+        writeString(view, 36, 'data'); // Subchunk2ID
         view.setUint32(40, numSamples * blockAlign, true); // Subchunk2Size
     }
     else {

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

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package/build/workers/InlineFeaturesExtractor.web.js.map

@@ -1 +1 @@
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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 / (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        const absValue = Math.abs(value);\n        localMinAmplitude = Math.min(localMinAmplitude, absValue);\n        localMaxAmplitude = Math.max(localMaxAmplitude, absValue);\n\n        if (absValue !== 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, rms);\n      maxAmplitude = Math.max(maxAmplitude, rms);\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 (!isSpeech && start - lastSpeechEnd < SPEECH_INERTIA_DURATION) {\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 ? extractMFCC(segmentData, sampleRate) : [];\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 newData = {\n        id: uniqueIdCounter++, // Assign unique ID and increment the counter\n        amplitude: algorithm === \"peak\" ? localMaxAmplitude : 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      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`;\n"]}
+{"version":3,"file":"InlineFeaturesExtractor.web.js","sourceRoot":"","sources":["../../src/workers/InlineFeaturesExtractor.web.tsx"],"names":[],"mappings":"AAAA,MAAM,CAAC,MAAM,uBAAuB,GAAG;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;CA6StC,CAAA","sourcesContent":["export const InlineFeaturesExtractor = `\n\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\n  console.log(\"[AudioFeaturesExtractor] Worker received message\", 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 ((weightedSum / sum) * (sampleRate / 2)) / magnitudeSpectrum.length;\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 / (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        const absValue = Math.abs(value);\n        localMinAmplitude = Math.min(localMinAmplitude, absValue);\n        localMaxAmplitude = Math.max(localMaxAmplitude, absValue);\n\n        if (absValue !== 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, rms);\n      maxAmplitude = Math.max(maxAmplitude, rms);\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 (!isSpeech && start - lastSpeechEnd < SPEECH_INERTIA_DURATION) {\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 ? extractMFCC(segmentData, sampleRate) : [];\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 newData = {\n        id: uniqueIdCounter++, // Assign unique ID and increment the counter\n        amplitude: algorithm === \"peak\" ? localMaxAmplitude : 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      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`\n"]}

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

@@ -1 +1 @@
-{"version":3,"file":"inlineAudioWebWorker.web.d.ts","sourceRoot":"","sources":["../../src/workers/inlineAudioWebWorker.web.tsx"],"names":[],"mappings":"AAAA,eAAO,MAAM,oBAAoB,4/PAiPhC,CAAC"}
+{"version":3,"file":"inlineAudioWebWorker.web.d.ts","sourceRoot":"","sources":["../../src/workers/inlineAudioWebWorker.web.tsx"],"names":[],"mappings":"AAAA,eAAO,MAAM,oBAAoB,4/PAiPhC,CAAA"}

package/build/workers/inlineAudioWebWorker.web.js.map

@@ -1 +1 @@
-{"version":3,"file":"inlineAudioWebWorker.web.js","sourceRoot":"","sources":["../../src/workers/inlineAudioWebWorker.web.tsx"],"names":[],"mappings":"AAAA,MAAM,CAAC,MAAM,oBAAoB,GAAG;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;CAiPnC,CAAC","sourcesContent":["export const InlineAudioWebWorker = `\nconst DEFAULT_BIT_DEPTH = 32;\nconst DEFAULT_SAMPLE_RATE = 44100;\n\nclass RecorderProcessor extends AudioWorkletProcessor {\n  constructor() {\n    super();\n    this.recordedBuffers = []; // Float32Array\n    this.newRecBuffer = []; // Float32Array\n    this.resampledBuffer = []; // Float32Array\n    this.exportIntervalSamples = 0;\n    this.samplesSinceLastExport = 0;\n    this.recordSampleRate = DEFAULT_SAMPLE_RATE; // To be overwritten\n    this.exportSampleRate = DEFAULT_SAMPLE_RATE; // To be overwritten\n    this.recordBitDepth = DEFAULT_BIT_DEPTH; // Default to 32-bit depth\n    this.exportBitDepth = DEFAULT_BIT_DEPTH; // To be overwritten\n    this.numberOfChannels = 1; // Default to 1 channel (mono)\n    this.isRecording = true;\n    this.port.onmessage = this.handleMessage.bind(this);\n  }\n\n  handleMessage(event) {\n    switch (event.data.command) {\n      case \"init\":\n        this.recordSampleRate = event.data.recordSampleRate;\n        this.exportSampleRate =\n          event.data.exportSampleRate || event.data.recordSampleRate;\n        this.exportIntervalSamples =\n          this.recordSampleRate * (event.data.interval / 1000);\n        if (event.data.numberOfChannels) {\n          this.numberOfChannels = event.data.numberOfChannels;\n        }\n        if (event.data.recordBitDepth) {\n          this.recordBitDepth = event.data.recordBitDepth;\n        }\n        this.exportBitDepth =\n          event.data.exportBitDepth || this.recordBitDepth || DEFAULT_BIT_DEPTH;\n        break;\n      case \"stop\":\n        this.isRecording = false;\n        this.getAllRecordedData()\n          .then((fullRecordedData) => {\n            this.port.postMessage({\n              command: \"recordedData\",\n              recordedData: fullRecordedData,\n              bitDepth: this.exportBitDepth,\n              sampleRate: this.exportSampleRate,\n            });\n            return fullRecordedData;\n          })\n          .catch((error) => {\n            console.error(\"Error extracting recorded data:\", error);\n          });\n        break;\n    }\n  }\n\n  process(inputs, outputs, parameters) {\n    if (!this.isRecording) return true;\n    const input = inputs[0];\n    if (input.length > 0) {\n      const newBuffer = new Float32Array(input[0]);\n      this.newRecBuffer.push(newBuffer);\n      this.recordedBuffers.push(newBuffer);\n      this.samplesSinceLastExport += newBuffer.length;\n\n      if (this.samplesSinceLastExport >= this.exportIntervalSamples) {\n        this.exportNewData();\n        this.samplesSinceLastExport = 0;\n      }\n    }\n    return true;\n  }\n\n  mergeBuffers(bufferArray, recLength) {\n    const result = new Float32Array(recLength);\n    let offset = 0;\n    for (let i = 0; i < bufferArray.length; i++) {\n      result.set(bufferArray[i], offset);\n      offset += bufferArray[i].length;\n    }\n    return result;\n  }\n\n  floatTo16BitPCM(input) {\n    const output = new Int16Array(input.length);\n    for (let i = 0; i < input.length; i++) {\n      const s = Math.max(-1, Math.min(1, input[i]));\n      output[i] = s < 0 ? s * 0x8000 : s * 0x7fff;\n    }\n    console.debug(\n      \"Float to 16-bit PCM conversion complete. Output byte length:\",\n      output.byteLength,\n    );\n    return output;\n  }\n\n  floatTo32BitPCM(input) {\n    const output = new Int32Array(input.length);\n    for (let i = 0; i < input.length; i++) {\n      const s = Math.max(-1, Math.min(1, input[i]));\n      output[i] = s < 0 ? s * 0x80000000 : s * 0x7fffffff;\n    }\n    console.debug(\n      \"Float to 32-bit PCM conversion complete. Output byte length:\",\n      output.byteLength,\n    );\n    return output;\n  }\n\n  resample(samples, targetSampleRate) {\n    if (this.recordSampleRate === targetSampleRate) {\n      return samples;\n    }\n    const resampledBuffer = new Float32Array(\n      (samples.length * targetSampleRate) / this.recordSampleRate,\n    );\n    const ratio = this.recordSampleRate / targetSampleRate;\n    let offset = 0;\n    for (let i = 0; i < resampledBuffer.length; i++) {\n      const nextOffset = Math.floor((i + 1) * ratio);\n      let accum = 0;\n      let count = 0;\n      for (let j = offset; j < nextOffset && j < samples.length; j++) {\n        accum += samples[j];\n        count++;\n      }\n      resampledBuffer[i] = accum / count;\n      offset = nextOffset;\n    }\n    return resampledBuffer;\n  }\n\n  async resampleBuffer(buffer, targetSampleRate) {\n    if (typeof OfflineAudioContext === \"undefined\") {\n      return this.resample(buffer, targetSampleRate);\n    }\n\n    if (this.recordSampleRate === targetSampleRate) {\n      return buffer;\n    }\n    const offlineContext = new OfflineAudioContext(\n      this.numberOfChannels,\n      buffer.length,\n      this.recordSampleRate,\n    );\n    const sourceBuffer = offlineContext.createBuffer(\n      this.numberOfChannels,\n      buffer.length,\n      this.recordSampleRate,\n    );\n    sourceBuffer.copyToChannel(buffer, 0);\n\n    const bufferSource = offlineContext.createBufferSource();\n    bufferSource.buffer = sourceBuffer;\n    bufferSource.connect(offlineContext.destination);\n    bufferSource.start();\n\n    const renderedBuffer = await offlineContext.startRendering();\n\n    const resampledBuffer = new Float32Array(renderedBuffer.length);\n    renderedBuffer.copyFromChannel(resampledBuffer, 0);\n\n    return resampledBuffer;\n  }\n\n  async exportNewData() {\n    // Calculate the total length of the new recorded buffers\n    const length = this.newRecBuffer.reduce(\n      (acc, buffer) => acc + buffer.length,\n      0,\n    );\n\n    // Merge all new recorded buffers into a single buffer\n    const mergedBuffer = this.mergeBuffers(this.newRecBuffer, length);\n\n    const resampledBuffer = await this.resampleBuffer(\n      mergedBuffer,\n      this.exportSampleRate,\n    );\n\n    let finalBuffer = resampledBuffer; // Float32Array\n    if (this.recordBitDepth !== this.exportBitDepth) {\n      if (this.exportBitDepth === 16) {\n        finalBuffer = this.floatTo16BitPCM(resampledBuffer);\n      } else if (this.exportBitDepth === 32) {\n        finalBuffer = this.floatTo32BitPCM(resampledBuffer);\n      }\n    }\n\n    const originalSize = mergedBuffer.byteLength;\n    const resampledSize = resampledBuffer.byteLength;\n    const finalSize = finalBuffer.byteLength;\n\n    // Clear the new recorded buffers after they have been processed\n    this.newRecBuffer.length = 0;\n\n    // Post the message to the main thread\n    // The first argument is the message data, containing the encoded WAV buffer\n    // The second argument is the transfer list, which transfers ownership of the ArrayBuffer\n    // to the main thread, avoiding the need to copy the buffer and improving performance\n    // this.port.postMessage({ recordedData: encodedWav.buffer, sampleRate: this.recordSampleRate }, [encodedWav.buffer]);\n    this.port.postMessage({\n      command: \"newData\",\n      recordedData: finalBuffer.buffer,\n      sampleRate: this.exportSampleRate,\n      bitDepth: this.exportBitDepth,\n    });\n  }\n\n  async getAllRecordedData() {\n    const length = this.recordedBuffers.reduce(\n      (acc, buffer) => acc + buffer.length,\n      0,\n    );\n    const mergedBuffer = this.mergeBuffers(this.recordedBuffers, length);\n    const resampledBuffer = await this.resampleBuffer(\n      mergedBuffer,\n      this.exportSampleRate,\n    );\n    // Convert to the desired bit depth if necessary\n    let finalBuffer = resampledBuffer;\n    if (this.recordBitDepth !== this.exportBitDepth) {\n      if (this.exportBitDepth === 16) {\n        finalBuffer = this.floatTo16BitPCM(resampledBuffer);\n      } else if (this.exportBitDepth === 32) {\n        finalBuffer = this.floatTo32BitPCM(resampledBuffer);\n      }\n    }\n\n    const originalSize = mergedBuffer.byteLength;\n    const resampledSize = resampledBuffer.byteLength;\n    const finalSize = finalBuffer.byteLength;\n\n    this.recordedBuffers.length = 0; // Clear the buffers after extraction\n\n    return finalBuffer.buffer;\n  }\n}\n\nregisterProcessor(\"recorder-processor\", RecorderProcessor);\n`;\n"]}
+{"version":3,"file":"inlineAudioWebWorker.web.js","sourceRoot":"","sources":["../../src/workers/inlineAudioWebWorker.web.tsx"],"names":[],"mappings":"AAAA,MAAM,CAAC,MAAM,oBAAoB,GAAG;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;CAiPnC,CAAA","sourcesContent":["export const InlineAudioWebWorker = `\nconst DEFAULT_BIT_DEPTH = 32;\nconst DEFAULT_SAMPLE_RATE = 44100;\n\nclass RecorderProcessor extends AudioWorkletProcessor {\n  constructor() {\n    super();\n    this.recordedBuffers = []; // Float32Array\n    this.newRecBuffer = []; // Float32Array\n    this.resampledBuffer = []; // Float32Array\n    this.exportIntervalSamples = 0;\n    this.samplesSinceLastExport = 0;\n    this.recordSampleRate = DEFAULT_SAMPLE_RATE; // To be overwritten\n    this.exportSampleRate = DEFAULT_SAMPLE_RATE; // To be overwritten\n    this.recordBitDepth = DEFAULT_BIT_DEPTH; // Default to 32-bit depth\n    this.exportBitDepth = DEFAULT_BIT_DEPTH; // To be overwritten\n    this.numberOfChannels = 1; // Default to 1 channel (mono)\n    this.isRecording = true;\n    this.port.onmessage = this.handleMessage.bind(this);\n  }\n\n  handleMessage(event) {\n    switch (event.data.command) {\n      case \"init\":\n        this.recordSampleRate = event.data.recordSampleRate;\n        this.exportSampleRate =\n          event.data.exportSampleRate || event.data.recordSampleRate;\n        this.exportIntervalSamples =\n          this.recordSampleRate * (event.data.interval / 1000);\n        if (event.data.numberOfChannels) {\n          this.numberOfChannels = event.data.numberOfChannels;\n        }\n        if (event.data.recordBitDepth) {\n          this.recordBitDepth = event.data.recordBitDepth;\n        }\n        this.exportBitDepth =\n          event.data.exportBitDepth || this.recordBitDepth || DEFAULT_BIT_DEPTH;\n        break;\n      case \"stop\":\n        this.isRecording = false;\n        this.getAllRecordedData()\n          .then((fullRecordedData) => {\n            this.port.postMessage({\n              command: \"recordedData\",\n              recordedData: fullRecordedData,\n              bitDepth: this.exportBitDepth,\n              sampleRate: this.exportSampleRate,\n            });\n            return fullRecordedData;\n          })\n          .catch((error) => {\n            console.error(\"Error extracting recorded data:\", error);\n          });\n        break;\n    }\n  }\n\n  process(inputs, outputs, parameters) {\n    if (!this.isRecording) return true;\n    const input = inputs[0];\n    if (input.length > 0) {\n      const newBuffer = new Float32Array(input[0]);\n      this.newRecBuffer.push(newBuffer);\n      this.recordedBuffers.push(newBuffer);\n      this.samplesSinceLastExport += newBuffer.length;\n\n      if (this.samplesSinceLastExport >= this.exportIntervalSamples) {\n        this.exportNewData();\n        this.samplesSinceLastExport = 0;\n      }\n    }\n    return true;\n  }\n\n  mergeBuffers(bufferArray, recLength) {\n    const result = new Float32Array(recLength);\n    let offset = 0;\n    for (let i = 0; i < bufferArray.length; i++) {\n      result.set(bufferArray[i], offset);\n      offset += bufferArray[i].length;\n    }\n    return result;\n  }\n\n  floatTo16BitPCM(input) {\n    const output = new Int16Array(input.length);\n    for (let i = 0; i < input.length; i++) {\n      const s = Math.max(-1, Math.min(1, input[i]));\n      output[i] = s < 0 ? s * 0x8000 : s * 0x7fff;\n    }\n    console.debug(\n      \"Float to 16-bit PCM conversion complete. Output byte length:\",\n      output.byteLength,\n    );\n    return output;\n  }\n\n  floatTo32BitPCM(input) {\n    const output = new Int32Array(input.length);\n    for (let i = 0; i < input.length; i++) {\n      const s = Math.max(-1, Math.min(1, input[i]));\n      output[i] = s < 0 ? s * 0x80000000 : s * 0x7fffffff;\n    }\n    console.debug(\n      \"Float to 32-bit PCM conversion complete. Output byte length:\",\n      output.byteLength,\n    );\n    return output;\n  }\n\n  resample(samples, targetSampleRate) {\n    if (this.recordSampleRate === targetSampleRate) {\n      return samples;\n    }\n    const resampledBuffer = new Float32Array(\n      (samples.length * targetSampleRate) / this.recordSampleRate,\n    );\n    const ratio = this.recordSampleRate / targetSampleRate;\n    let offset = 0;\n    for (let i = 0; i < resampledBuffer.length; i++) {\n      const nextOffset = Math.floor((i + 1) * ratio);\n      let accum = 0;\n      let count = 0;\n      for (let j = offset; j < nextOffset && j < samples.length; j++) {\n        accum += samples[j];\n        count++;\n      }\n      resampledBuffer[i] = accum / count;\n      offset = nextOffset;\n    }\n    return resampledBuffer;\n  }\n\n  async resampleBuffer(buffer, targetSampleRate) {\n    if (typeof OfflineAudioContext === \"undefined\") {\n      return this.resample(buffer, targetSampleRate);\n    }\n\n    if (this.recordSampleRate === targetSampleRate) {\n      return buffer;\n    }\n    const offlineContext = new OfflineAudioContext(\n      this.numberOfChannels,\n      buffer.length,\n      this.recordSampleRate,\n    );\n    const sourceBuffer = offlineContext.createBuffer(\n      this.numberOfChannels,\n      buffer.length,\n      this.recordSampleRate,\n    );\n    sourceBuffer.copyToChannel(buffer, 0);\n\n    const bufferSource = offlineContext.createBufferSource();\n    bufferSource.buffer = sourceBuffer;\n    bufferSource.connect(offlineContext.destination);\n    bufferSource.start();\n\n    const renderedBuffer = await offlineContext.startRendering();\n\n    const resampledBuffer = new Float32Array(renderedBuffer.length);\n    renderedBuffer.copyFromChannel(resampledBuffer, 0);\n\n    return resampledBuffer;\n  }\n\n  async exportNewData() {\n    // Calculate the total length of the new recorded buffers\n    const length = this.newRecBuffer.reduce(\n      (acc, buffer) => acc + buffer.length,\n      0,\n    );\n\n    // Merge all new recorded buffers into a single buffer\n    const mergedBuffer = this.mergeBuffers(this.newRecBuffer, length);\n\n    const resampledBuffer = await this.resampleBuffer(\n      mergedBuffer,\n      this.exportSampleRate,\n    );\n\n    let finalBuffer = resampledBuffer; // Float32Array\n    if (this.recordBitDepth !== this.exportBitDepth) {\n      if (this.exportBitDepth === 16) {\n        finalBuffer = this.floatTo16BitPCM(resampledBuffer);\n      } else if (this.exportBitDepth === 32) {\n        finalBuffer = this.floatTo32BitPCM(resampledBuffer);\n      }\n    }\n\n    const originalSize = mergedBuffer.byteLength;\n    const resampledSize = resampledBuffer.byteLength;\n    const finalSize = finalBuffer.byteLength;\n\n    // Clear the new recorded buffers after they have been processed\n    this.newRecBuffer.length = 0;\n\n    // Post the message to the main thread\n    // The first argument is the message data, containing the encoded WAV buffer\n    // The second argument is the transfer list, which transfers ownership of the ArrayBuffer\n    // to the main thread, avoiding the need to copy the buffer and improving performance\n    // this.port.postMessage({ recordedData: encodedWav.buffer, sampleRate: this.recordSampleRate }, [encodedWav.buffer]);\n    this.port.postMessage({\n      command: \"newData\",\n      recordedData: finalBuffer.buffer,\n      sampleRate: this.exportSampleRate,\n      bitDepth: this.exportBitDepth,\n    });\n  }\n\n  async getAllRecordedData() {\n    const length = this.recordedBuffers.reduce(\n      (acc, buffer) => acc + buffer.length,\n      0,\n    );\n    const mergedBuffer = this.mergeBuffers(this.recordedBuffers, length);\n    const resampledBuffer = await this.resampleBuffer(\n      mergedBuffer,\n      this.exportSampleRate,\n    );\n    // Convert to the desired bit depth if necessary\n    let finalBuffer = resampledBuffer;\n    if (this.recordBitDepth !== this.exportBitDepth) {\n      if (this.exportBitDepth === 16) {\n        finalBuffer = this.floatTo16BitPCM(resampledBuffer);\n      } else if (this.exportBitDepth === 32) {\n        finalBuffer = this.floatTo32BitPCM(resampledBuffer);\n      }\n    }\n\n    const originalSize = mergedBuffer.byteLength;\n    const resampledSize = resampledBuffer.byteLength;\n    const finalSize = finalBuffer.byteLength;\n\n    this.recordedBuffers.length = 0; // Clear the buffers after extraction\n\n    return finalBuffer.buffer;\n  }\n}\n\nregisterProcessor(\"recorder-processor\", RecorderProcessor);\n`\n"]}

package/expo-module.config.json

@@ -1,18 +1,9 @@
 {
-  "platforms": [
-    "ios",
-    "tvos",
-    "android",
-    "web"
-  ],
-  "ios": {
-    "modules": [
-      "ExpoAudioStreamModule"
-    ]
-  },
-  "android": {
-    "modules": [
-      "net.siteed.audiostream.ExpoAudioStreamModule"
-    ]
-  }
-}
+    "platforms": ["ios", "tvos", "android", "web"],
+    "ios": {
+        "modules": ["ExpoAudioStreamModule"]
+    },
+    "android": {
+        "modules": ["net.siteed.audiostream.ExpoAudioStreamModule"]
+    }
+}

package/ios/AudioStreamManager.swift

@@ -379,6 +379,7 @@ class AudioStreamManager: NSObject {
             
             let result = RecordingResult(
                 fileUri: fileURL.absoluteString,
+                filename: fileURL.lastPathComponent,
                 mimeType: mimeType,
                 duration: duration,
                 size: fileSize,

package/ios/ExpoAudioStreamModule.swift

@@ -197,6 +197,7 @@ public class ExpoAudioStreamModule: Module, AudioStreamManagerDelegate {
                 // Convert RecordingResult to a dictionary
                 let resultDict: [String: Any] = [
                     "fileUri": recordingResult.fileUri,
+                    "filename": recordingResult.filename,
                     "durationMs": recordingResult.duration,
                     "size": recordingResult.size,
                     "channels": recordingResult.channels,

package/ios/RecordingResult.swift

@@ -2,6 +2,7 @@
 
 struct RecordingResult {
     var fileUri: String
+    var filename: String
     var mimeType: String
     var duration: Int64
     var size: Int64