@siteed/expo-audio-stream 1.17.0 → 2.0.0

package/build/AudioAnalysis/extractAudioAnalysis.js

@@ -1,131 +1,130 @@
+// packages/expo-audio-stream/src/AudioAnalysis/extractAudioAnalysis.ts
+/**
+ * This module provides functions for extracting and analyzing audio data.
+ * - `extractAudioAnalysis`: For detailed analysis with customizable ranges and decoding options.
+ * - `extractWavAudioAnalysis`: For analyzing WAV files without decoding, preserving original PCM values.
+ * - `extractPreview`: For generating quick previews of audio waveforms, optimized for UI rendering.
+ */
+import crc32 from 'crc-32';
 import ExpoAudioStreamModule from '../ExpoAudioStreamModule';
 import { isWeb } from '../constants';
+import { processAudioBuffer } from '../utils/audioProcessing';
 import { convertPCMToFloat32 } from '../utils/convertPCMToFloat32';
 import { getWavFileInfo } from '../utils/getWavFileInfo';
 import { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web';
-export async function extractAudioFromAnyFormat({ fileUri, arrayBuffer, mimeType, decodingOptions, startTime, endTime, ...restProps }) {
+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 {
-            // Get the audio data
-            let audioBuffer;
-            if (arrayBuffer) {
-                audioBuffer = arrayBuffer;
-            }
-            else if (fileUri) {
-                const response = await fetch(fileUri);
-                if (!response.ok) {
-                    throw new Error(`Failed to fetch fileUri: ${response.statusText}`);
-                }
-                audioBuffer = await response.arrayBuffer();
-            }
-            else {
-                throw new Error('Either arrayBuffer or fileUri must be provided');
-            }
-            // Create audio context with target sample rate if specified
+            // Create AudioContext here
             const audioContext = new (window.AudioContext ||
                 window.webkitAudioContext)({
-                sampleRate: decodingOptions?.targetSampleRate,
-            });
-            // Decode the audio data
-            const decodedAudioBuffer = await audioContext.decodeAudioData(audioBuffer);
-            // Calculate the actual duration in milliseconds
-            const fullDurationMs = decodedAudioBuffer.duration * 1000;
-            const effectiveDurationMs = endTime
-                ? endTime - (startTime || 0)
-                : fullDurationMs - (startTime || 0);
-            // Create a new buffer for the selected range
-            const rangeLength = decodedAudioBuffer.length;
-            const rangeBuffer = new AudioBuffer({
-                length: rangeLength,
-                numberOfChannels: decodedAudioBuffer.numberOfChannels,
-                sampleRate: decodedAudioBuffer.sampleRate,
+                sampleRate: decodingOptions?.targetSampleRate ?? 16000,
             });
-            // Copy the selected range
-            for (let channel = 0; channel < decodedAudioBuffer.numberOfChannels; channel++) {
-                const channelData = decodedAudioBuffer.getChannelData(channel);
-                const rangeData = channelData.slice(0, rangeLength);
-                rangeBuffer.copyToChannel(rangeData, channel);
-            }
-            // Use the range buffer instead of the full buffer
-            let processedBuffer = rangeBuffer;
-            // Get original properties
-            const originalSampleRate = processedBuffer.sampleRate;
-            const length = processedBuffer.length;
-            // Determine target format
-            const targetChannels = decodingOptions?.targetChannels ??
-                processedBuffer.numberOfChannels;
-            const targetSampleRate = decodingOptions?.targetSampleRate ?? processedBuffer.sampleRate;
-            // Create offline context for resampling if needed
-            if (targetSampleRate !== originalSampleRate) {
-                const offlineCtx = new OfflineAudioContext(targetChannels, (length * targetSampleRate) / originalSampleRate, targetSampleRate);
-                const source = offlineCtx.createBufferSource();
-                source.buffer = processedBuffer;
-                source.connect(offlineCtx.destination);
-                source.start();
-                processedBuffer = await offlineCtx.startRendering();
-            }
-            // Convert to the desired format
-            const newLength = processedBuffer.length;
-            let wavBuffer;
-            // Create appropriate buffer based on target bit depth
-            switch (decodingOptions?.targetBitDepth) {
-                case 16:
-                    wavBuffer = new Int16Array(newLength * targetChannels);
-                    break;
-                case 8:
-                    wavBuffer = new Int8Array(newLength * targetChannels);
-                    break;
-                case 32:
-                default:
-                    wavBuffer = new Float32Array(newLength * targetChannels);
-                    break;
+            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 and initialize the worker
+                const blob = new Blob([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,
+                        // enableLogging: !!logger,
+                        features,
+                    });
+                });
             }
-            // Interleave channels and handle bit depth conversion
-            const numChannels = Math.min(processedBuffer.numberOfChannels, targetChannels);
-            for (let channel = 0; channel < numChannels; channel++) {
-                const channelData = processedBuffer.getChannelData(channel);
-                for (let i = 0; i < newLength; i++) {
-                    let sample = channelData[i];
-                    // Normalize if requested
-                    if (decodingOptions?.normalizeAudio) {
-                        sample = Math.max(-1, Math.min(1, sample));
-                    }
-                    // Convert sample based on target bit depth
-                    if (decodingOptions?.targetBitDepth === 16) {
-                        sample = sample * 32767; // Convert to 16-bit range
-                    }
-                    else if (decodingOptions?.targetBitDepth === 8) {
-                        sample = sample * 127; // Convert to 8-bit range
-                    }
-                    wavBuffer[i * targetChannels + channel] = sample;
-                }
+            finally {
+                await audioContext.close();
             }
-            // Pass the duration to extractAudioAnalysis
-            return await extractAudioAnalysis({
-                arrayBuffer: wavBuffer.buffer,
-                bitDepth: decodingOptions?.targetBitDepth ?? 32,
-                skipWavHeader: true,
-                sampleRate: targetSampleRate,
-                numberOfChannels: targetChannels,
-                durationMs: effectiveDurationMs,
-                ...restProps,
-            });
         }
         catch (error) {
-            console.error('Failed to process audio:', error);
+            logger?.error('Failed to process audio:', error);
             throw error;
         }
     }
     else {
-        // For native platforms, pass through all options
-        return await extractAudioAnalysis({
-            fileUri,
-            decodingOptions,
-            ...restProps,
-        });
+        return await ExpoAudioStreamModule.extractAudioAnalysis(props);
     }
 }
-export const extractAudioAnalysis = async ({ fileUri, pointsPerSecond = 20, arrayBuffer, bitDepth, skipWavHeader = true, durationMs, sampleRate, numberOfChannels, algorithm = 'rms', features, featuresExtratorUrl, logger, }) => {
+/**
+ * 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');
@@ -141,7 +140,7 @@ export const extractAudioAnalysis = async ({ fileUri, pointsPerSecond = 20, arra
         }
         // Create a new copy of the ArrayBuffer to avoid detachment issues
         const bufferCopy = arrayBuffer.slice(0);
-        logger?.log(`extractAudioAnalysis skipWavHeader=${skipWavHeader} bitDepth=${bitDepth} len=${bufferCopy.byteLength}`, bufferCopy.slice(0, 100));
+        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`);
@@ -152,21 +151,18 @@ export const extractAudioAnalysis = async ({ fileUri, pointsPerSecond = 20, arra
         const { pcmValues: channelData, min, max, } = await convertPCMToFloat32({
             buffer: arrayBuffer,
             bitDepth: actualBitDepth,
-            skipWavHeader,
         });
-        logger?.log(`extractAudioAnalysis skipWaveHeader=${skipWavHeader} convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`);
+        logger?.log(`extractAudioAnalysis convertPCMToFloat32 length=${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) => {
-            let worker;
-            if (featuresExtratorUrl) {
-                worker = new Worker(new URL(featuresExtratorUrl, window.location.href));
-            }
-            else {
-                const blob = new Blob([InlineFeaturesExtractor], {
-                    type: 'application/javascript',
-                });
-                const url = URL.createObjectURL(blob);
-                worker = new Worker(url);
-            }
+            const blob = new Blob([InlineFeaturesExtractor], {
+                type: 'application/javascript',
+            });
+            const url = URL.createObjectURL(blob);
+            const worker = new Worker(url);
             worker.onmessage = (event) => {
                 resolve(event.data.result);
             };
@@ -175,10 +171,10 @@ export const extractAudioAnalysis = async ({ fileUri, pointsPerSecond = 20, arra
             };
             worker.postMessage({
                 command: 'process',
-                channelData,
+                channelData: constrainedChannelData,
                 sampleRate,
-                pointsPerSecond,
-                algorithm,
+                segmentDurationMs,
+                logger,
                 bitDepth,
                 fullAudioDurationMs: durationMs,
                 numberOfChannels,
@@ -191,54 +187,43 @@ export const extractAudioAnalysis = async ({ fileUri, pointsPerSecond = 20, arra
         }
         logger?.log(`extractAudioAnalysis`, {
             fileUri,
-            pointsPerSecond,
-            algorithm,
+            segmentDurationMs,
         });
         const res = await ExpoAudioStreamModule.extractAudioAnalysis({
             fileUri,
-            pointsPerSecond,
-            skipWavHeader,
-            algorithm,
+            segmentDurationMs,
             features,
+            position,
+            length,
         });
         logger?.log(`extractAudioAnalysis`, res);
         return res;
     }
 };
-export async function extractPreview({ fileUri, numberOfPoints, algorithm = 'rms', startTime, endTime, decodingOptions, }) {
-    if (isWeb) {
-        // For web, we can reuse the existing extractAudioFromAnyFormat with modified parameters
-        const analysis = await extractAudioFromAnyFormat({
-            fileUri,
-            algorithm,
-            decodingOptions,
-            startTime, // Pass startTime
-            endTime, // Pass endTime
-            pointsPerSecond: (numberOfPoints ?? 100) /
-                ((endTime ? endTime - (startTime || 0) : 1000) / 1000), // Adjust points per second calculation
-        });
-        // Convert AudioAnalysis to AudioPreview format and adjust duration
-        return {
-            pointsPerSecond: analysis.pointsPerSecond,
-            durationMs: endTime
-                ? endTime - (startTime || 0)
-                : analysis.durationMs, // Use range duration if specified
-            amplitudeRange: analysis.amplitudeRange,
-            dataPoints: analysis.dataPoints.map((point) => ({
-                id: point.id,
-                amplitude: point.amplitude,
-                startTime: point.startTime,
-                endTime: point.endTime,
-            })),
-        };
-    }
-    return await ExpoAudioStreamModule.extractPreview({
+/**
+ * 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,
-        numberOfPoints,
-        algorithm,
-        startTime,
-        endTime,
+        startTimeMs,
+        endTimeMs,
+        logger,
+        segmentDurationMs,
         decodingOptions,
     });
+    // Transform the result into AudioPreview format
+    return analysis;
 }
+export const extractAudioData = async (props) => {
+    return await ExpoAudioStreamModule.extractAudioData(props);
+};
 //# sourceMappingURL=extractAudioAnalysis.js.map

package/build/AudioAnalysis/extractAudioAnalysis.js.map

@@ -1 +1 @@
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packages/expo-audio-stream/src/AudioAnalysis/extractAudioAnalysis.ts\nimport { ConsoleLike } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\nimport { isWeb } from '../constants'\nimport {\n    AmplitudeAlgorithm,\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    AudioPreview,\n    DecodingConfig,\n    PreviewOptions,\n} from './AudioAnalysis.types'\nimport { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'\nimport { getWavFileInfo, WavFileInfo } from '../utils/getWavFileInfo'\nimport { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web'\n\nexport interface ExtractAudioAnalysisProps {\n    fileUri?: string // should provide either fileUri or arrayBuffer\n    wavMetadata?: WavFileInfo\n    arrayBuffer?: ArrayBuffer\n    bitDepth?: number\n    skipWavHeader?: boolean\n    durationMs?: number\n    sampleRate?: number\n    numberOfChannels?: number\n    algorithm?: AmplitudeAlgorithm\n    position?: number // Optional number of bytes to skip. Default is 0\n    length?: number // Optional number of bytes to read.\n    pointsPerSecond?: 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\nexport interface ExtractAudioFromAnyFormatProps\n    extends ExtractAudioAnalysisProps {\n    mimeType?: string\n    decodingOptions?: DecodingConfig\n    startTime?: number // Add start time in milliseconds\n    endTime?: number // Add end time in milliseconds\n}\n\nexport async function extractAudioFromAnyFormat({\n    fileUri,\n    arrayBuffer,\n    mimeType,\n    decodingOptions,\n    startTime,\n    endTime,\n    ...restProps\n}: ExtractAudioFromAnyFormatProps): Promise<AudioAnalysis> {\n    if (isWeb) {\n        try {\n            // Get the audio data\n            let audioBuffer: ArrayBuffer\n            if (arrayBuffer) {\n                audioBuffer = arrayBuffer\n            } else if (fileUri) {\n                const response = await fetch(fileUri)\n                if (!response.ok) {\n                    throw new Error(\n                        `Failed to fetch fileUri: ${response.statusText}`\n                    )\n                }\n                audioBuffer = await response.arrayBuffer()\n            } else {\n                throw new Error(\n                    'Either arrayBuffer or fileUri must be provided'\n                )\n            }\n\n            // Create audio context with target sample rate if specified\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)({\n                sampleRate: decodingOptions?.targetSampleRate,\n            })\n\n            // Decode the audio data\n            const decodedAudioBuffer =\n                await audioContext.decodeAudioData(audioBuffer)\n\n            // Calculate the actual duration in milliseconds\n            const fullDurationMs = decodedAudioBuffer.duration * 1000\n            const effectiveDurationMs = endTime\n                ? endTime - (startTime || 0)\n                : fullDurationMs - (startTime || 0)\n\n            // Create a new buffer for the selected range\n            const rangeLength = decodedAudioBuffer.length\n            const rangeBuffer = new AudioBuffer({\n                length: rangeLength,\n                numberOfChannels: decodedAudioBuffer.numberOfChannels,\n                sampleRate: decodedAudioBuffer.sampleRate,\n            })\n\n            // Copy the selected range\n            for (\n                let channel = 0;\n                channel < decodedAudioBuffer.numberOfChannels;\n                channel++\n            ) {\n                const channelData = decodedAudioBuffer.getChannelData(channel)\n                const rangeData = channelData.slice(0, rangeLength)\n                rangeBuffer.copyToChannel(rangeData, channel)\n            }\n\n            // Use the range buffer instead of the full buffer\n            let processedBuffer = rangeBuffer\n\n            // Get original properties\n            const originalSampleRate = processedBuffer.sampleRate\n            const length = processedBuffer.length\n\n            // Determine target format\n            const targetChannels =\n                decodingOptions?.targetChannels ??\n                processedBuffer.numberOfChannels\n            const targetSampleRate =\n                decodingOptions?.targetSampleRate ?? processedBuffer.sampleRate\n\n            // Create offline context for resampling if needed\n            if (targetSampleRate !== originalSampleRate) {\n                const offlineCtx = new OfflineAudioContext(\n                    targetChannels,\n                    (length * targetSampleRate) / originalSampleRate,\n                    targetSampleRate\n                )\n                const source = offlineCtx.createBufferSource()\n                source.buffer = processedBuffer\n                source.connect(offlineCtx.destination)\n                source.start()\n                processedBuffer = await offlineCtx.startRendering()\n            }\n\n            // Convert to the desired format\n            const newLength = processedBuffer.length\n            let wavBuffer: Float32Array | Int16Array | Int8Array\n\n            // Create appropriate buffer based on target bit depth\n            switch (decodingOptions?.targetBitDepth) {\n                case 16:\n                    wavBuffer = new Int16Array(newLength * targetChannels)\n                    break\n                case 8:\n                    wavBuffer = new Int8Array(newLength * targetChannels)\n                    break\n                case 32:\n                default:\n                    wavBuffer = new Float32Array(newLength * targetChannels)\n                    break\n            }\n\n            // Interleave channels and handle bit depth conversion\n            const numChannels = Math.min(\n                processedBuffer.numberOfChannels,\n                targetChannels\n            )\n            for (let channel = 0; channel < numChannels; channel++) {\n                const channelData = processedBuffer.getChannelData(channel)\n                for (let i = 0; i < newLength; i++) {\n                    let sample = channelData[i]\n\n                    // Normalize if requested\n                    if (decodingOptions?.normalizeAudio) {\n                        sample = Math.max(-1, Math.min(1, sample))\n                    }\n\n                    // Convert sample based on target bit depth\n                    if (decodingOptions?.targetBitDepth === 16) {\n                        sample = sample * 32767 // Convert to 16-bit range\n                    } else if (decodingOptions?.targetBitDepth === 8) {\n                        sample = sample * 127 // Convert to 8-bit range\n                    }\n\n                    wavBuffer[i * targetChannels + channel] = sample\n                }\n            }\n\n            // Pass the duration to extractAudioAnalysis\n            return await extractAudioAnalysis({\n                arrayBuffer: wavBuffer.buffer as ArrayBuffer,\n                bitDepth: decodingOptions?.targetBitDepth ?? 32,\n                skipWavHeader: true,\n                sampleRate: targetSampleRate,\n                numberOfChannels: targetChannels,\n                durationMs: effectiveDurationMs,\n                ...restProps,\n            })\n        } catch (error) {\n            console.error('Failed to process audio:', error)\n            throw error\n        }\n    } else {\n        // For native platforms, pass through all options\n        return await extractAudioAnalysis({\n            fileUri,\n            decodingOptions,\n            ...restProps,\n        })\n    }\n}\n\nexport const extractAudioAnalysis = async ({\n    fileUri,\n    pointsPerSecond = 20,\n    arrayBuffer,\n    bitDepth,\n    skipWavHeader = true,\n    durationMs,\n    sampleRate,\n    numberOfChannels,\n    algorithm = 'rms',\n    features,\n    featuresExtratorUrl,\n    logger,\n}: ExtractAudioAnalysisProps): 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: ${response.statusText}`\n                )\n            }\n\n            arrayBuffer = await response.arrayBuffer()\n            logger?.log(`fetched fileUri`, arrayBuffer.byteLength, arrayBuffer)\n        }\n\n        // Create a new copy of the ArrayBuffer to avoid detachment issues\n        const bufferCopy = arrayBuffer.slice(0)\n        logger?.log(\n            `extractAudioAnalysis skipWavHeader=${skipWavHeader} 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=${actualBitDepth}`)\n\n        const {\n            pcmValues: channelData,\n            min,\n            max,\n        } = await convertPCMToFloat32({\n            buffer: arrayBuffer,\n            bitDepth: actualBitDepth,\n            skipWavHeader,\n        })\n        logger?.log(\n            `extractAudioAnalysis skipWaveHeader=${skipWavHeader} convertPCMToFloat32 length=${channelData.length} range: [ ${min} :: ${max} ]`\n        )\n\n        return new Promise((resolve, reject) => {\n            let worker: Worker\n            if (featuresExtratorUrl) {\n                worker = new Worker(\n                    new URL(featuresExtratorUrl, window.location.href)\n                )\n            } else {\n                const blob = new Blob([InlineFeaturesExtractor], {\n                    type: 'application/javascript',\n                })\n                const url = URL.createObjectURL(blob)\n                worker = new Worker(url)\n            }\n\n            worker.onmessage = (event) => {\n                resolve(event.data.result)\n            }\n\n            worker.onerror = (error) => {\n                reject(error)\n            }\n\n            worker.postMessage({\n                command: 'process',\n                channelData,\n                sampleRate,\n                pointsPerSecond,\n                algorithm,\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            pointsPerSecond,\n            algorithm,\n        })\n        const res = await ExpoAudioStreamModule.extractAudioAnalysis({\n            fileUri,\n            pointsPerSecond,\n            skipWavHeader,\n            algorithm,\n            features,\n        })\n        logger?.log(`extractAudioAnalysis`, res)\n        return res\n    }\n}\n\nexport async function extractPreview({\n    fileUri,\n    numberOfPoints,\n    algorithm = 'rms',\n    startTime,\n    endTime,\n    decodingOptions,\n}: PreviewOptions): Promise<AudioPreview> {\n    if (isWeb) {\n        // For web, we can reuse the existing extractAudioFromAnyFormat with modified parameters\n        const analysis = await extractAudioFromAnyFormat({\n            fileUri,\n            algorithm,\n            decodingOptions,\n            startTime, // Pass startTime\n            endTime, // Pass endTime\n            pointsPerSecond:\n                (numberOfPoints ?? 100) /\n                ((endTime ? endTime - (startTime || 0) : 1000) / 1000), // Adjust points per second calculation\n        })\n\n        // Convert AudioAnalysis to AudioPreview format and adjust duration\n        return {\n            pointsPerSecond: analysis.pointsPerSecond,\n            durationMs: endTime\n                ? endTime - (startTime || 0)\n                : analysis.durationMs, // Use range duration if specified\n            amplitudeRange: analysis.amplitudeRange,\n            dataPoints: analysis.dataPoints.map((point) => ({\n                id: point.id,\n                amplitude: point.amplitude,\n                startTime: point.startTime,\n                endTime: point.endTime,\n            })),\n        }\n    }\n\n    return await ExpoAudioStreamModule.extractPreview({\n        fileUri,\n        numberOfPoints,\n        algorithm,\n        startTime,\n        endTime,\n        decodingOptions,\n    })\n}\n"]}
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packages/expo-audio-stream/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 crc32 from 'crc-32'\n\nimport { ConsoleLike, ExtractAudioDataOptions } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\nimport { isWeb } from '../constants'\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DataPoint,\n    DecodingConfig,\n    PreviewOptions,\n} from './AudioAnalysis.types'\nimport { processAudioBuffer } from '../utils/audioProcessing'\nimport { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'\nimport { getWavFileInfo, WavFileInfo } from '../utils/getWavFileInfo'\nimport { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.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 and initialize the worker\n                const blob = new Blob([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                        // 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        return await ExpoAudioStreamModule.extractAudioAnalysis(props)\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([InlineFeaturesExtractor], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            const worker = new Worker(url)\n\n            worker.onmessage = (event) => {\n                resolve(event.data.result)\n            }\n\n            worker.onerror = (error) => {\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 ExpoAudioStreamModule.extractAudioAnalysis({\n            fileUri,\n            segmentDurationMs,\n            features,\n            position,\n            length,\n        })\n        logger?.log(`extractAudioAnalysis`, res)\n        return res\n    }\n}\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\nexport const extractAudioData = async (props: ExtractAudioDataOptions) => {\n    return await ExpoAudioStreamModule.extractAudioData(props)\n}"]}

package/build/ExpoAudioStream.types.d.ts

@@ -1,4 +1,4 @@
-import { AmplitudeAlgorithm, AudioAnalysis, AudioFeaturesOptions } from './AudioAnalysis/AudioAnalysis.types';
+import { AudioAnalysis, AudioFeaturesOptions, DecodingConfig } from './AudioAnalysis/AudioAnalysis.types';
 import { AudioAnalysisEvent } from './events';
 export interface CompressionInfo {
     size: number;
@@ -30,9 +30,11 @@ export interface AudioDataEvent {
 export type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit';
 export type SampleRate = 16000 | 44100 | 48000;
 export type BitDepth = 8 | 16 | 32;
+export type PCMFormat = `pcm_${BitDepth}bit`;
 export type ConsoleLike = {
     log: (message: string, ...args: unknown[]) => void;
     debug: (message: string, ...args: unknown[]) => void;
+    info: (message: string, ...args: unknown[]) => void;
     warn: (message: string, ...args: unknown[]) => void;
     error: (message: string, ...args: unknown[]) => void;
 };
@@ -99,8 +101,7 @@ export interface RecordingConfig {
     notification?: NotificationConfig;
     enableProcessing?: boolean;
     ios?: IOSConfig;
-    pointsPerSecond?: number;
-    algorithm?: AmplitudeAlgorithm;
+    segmentDurationMs?: number;
     features?: AudioFeaturesOptions;
     onAudioStream?: (_: AudioDataEvent) => Promise<void>;
     onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>;
@@ -146,6 +147,49 @@ export interface WaveformConfig {
     mirror?: boolean;
     height?: number;
 }
+export interface ExtractAudioDataOptions {
+    fileUri: string;
+    startTimeMs?: number;
+    endTimeMs?: number;
+    position?: number;
+    length?: number;
+    /** Include normalized audio data in [-1, 1] range */
+    includeNormalizedData?: boolean;
+    /** Include base64 encoded string representation of the audio data */
+    includeBase64Data?: boolean;
+    /** Include WAV header in the PCM data (makes it a valid WAV file) */
+    includeWavHeader?: boolean;
+    /** Logger for debugging - can pass console directly. */
+    logger?: ConsoleLike;
+    /** Compute the checksum of the pcm data */
+    computeChecksum?: boolean;
+    /** Target config for the normalized audio (Android and Web) */
+    decodingOptions?: DecodingConfig;
+}
+export interface ExtractedAudioData {
+    /** Raw PCM audio data */
+    pcmData: Uint8Array;
+    /** Normalized audio data in [-1, 1] range (when includeNormalizedData is true) */
+    normalizedData?: Float32Array;
+    /** Base64 encoded string representation of the audio data (when includeBase64Data is true) */
+    base64Data?: string;
+    /** Sample rate in Hz (e.g., 44100, 48000) */
+    sampleRate: number;
+    /** Number of audio channels (1 for mono, 2 for stereo) */
+    channels: number;
+    /** Bits per sample (8, 16, or 32) */
+    bitDepth: BitDepth;
+    /** Duration of the audio in milliseconds */
+    durationMs: number;
+    /** PCM format identifier (e.g., "pcm_16bit") */
+    format: PCMFormat;
+    /** Total number of audio samples per channel */
+    samples: number;
+    /** Whether the pcmData includes a WAV header */
+    hasWavHeader?: boolean;
+    /** CRC32 Checksum of pcm data */
+    checksum?: number;
+}
 export interface UseAudioRecorderState {
     startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>;
     stopRecording: () => Promise<AudioRecording | null>;

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package/build/ExpoAudioStream.types.js.map

@@ -1 +1 @@
-{"version":3,"file":"ExpoAudioStream.types.js","sourceRoot":"","sources":["../src/ExpoAudioStream.types.ts"],"names":[],"mappings":"","sourcesContent":["// packages/expo-audio-stream/src/ExpoAudioStream.types.ts\nimport {\n    AmplitudeAlgorithm,\n    AudioAnalysis,\n    AudioFeaturesOptions,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport { AudioAnalysisEvent } from './events'\n\nexport interface CompressionInfo {\n    size: number\n    mimeType: string\n    bitrate: number\n    format: string\n    compressedFileUri?: string\n}\n\nexport interface AudioStreamStatus {\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number\n    size: number\n    interval: number\n    intervalAnalysis: number\n    mimeType: string\n    compression?: CompressionInfo\n}\n\nexport interface AudioDataEvent {\n    data: string | Float32Array\n    position: number\n    fileUri: string\n    eventDataSize: number\n    totalSize: number\n    compression?: CompressionInfo & {\n        data?: string | Blob // Base64 (native) or Float32Array (web) encoded compressed data chunk\n    }\n}\n\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\nexport type SampleRate = 16000 | 44100 | 48000 \nexport type BitDepth = 8 | 16 | 32\n\nexport type ConsoleLike = {\n    log: (message: string, ...args: unknown[]) => void\n    debug: (message: string, ...args: unknown[]) => void\n    warn: (message: string, ...args: unknown[]) => void\n    error: (message: string, ...args: unknown[]) => void\n}\n\nexport interface Chunk {\n    text: string\n    timestamp: [number, number | null]\n}\n\nexport interface TranscriberData {\n    id: string\n    isBusy: boolean\n    text: string\n    startTime: number\n    endTime: number\n    chunks: Chunk[]\n}\n\nexport interface AudioRecording {\n    fileUri: string\n    filename: string\n    durationMs: number\n    size: number\n    mimeType: string\n    channels: number\n    bitDepth: BitDepth\n    sampleRate: SampleRate\n    createdAt?: number\n    transcripts?: TranscriberData[]\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    compression?: CompressionInfo & {\n        compressedFileUri: string\n    }\n}\n\nexport interface StartRecordingResult {\n    fileUri: string\n    mimeType: string\n    channels?: number\n    bitDepth?: BitDepth\n    sampleRate?: SampleRate\n    compression?: CompressionInfo & {\n        compressedFileUri: string\n    }\n}\n\nexport interface AudioSessionConfig {\n    category?:\n        | 'Ambient'\n        | 'SoloAmbient'\n        | 'Playback'\n        | 'Record'\n        | 'PlayAndRecord'\n        | 'MultiRoute'\n    mode?:\n        | 'Default'\n        | 'VoiceChat'\n        | 'VideoChat'\n        | 'GameChat'\n        | 'VideoRecording'\n        | 'Measurement'\n        | 'MoviePlayback'\n        | 'SpokenAudio'\n    categoryOptions?: (\n        | 'MixWithOthers'\n        | 'DuckOthers'\n        | 'InterruptSpokenAudioAndMixWithOthers'\n        | 'AllowBluetooth'\n        | 'AllowBluetoothA2DP'\n        | 'AllowAirPlay'\n        | 'DefaultToSpeaker'\n    )[]\n}\n\nexport interface IOSConfig {\n    audioSession?: AudioSessionConfig\n}\n\n// Add new type for interruption reasons\nexport type RecordingInterruptionReason =\n    | 'audioFocusLoss'\n    | 'audioFocusGain'\n    | 'phoneCall'\n    | 'phoneCallEnded'\n    | 'recordingStopped'\n\n// Add new interface for interruption events\nexport interface RecordingInterruptionEvent {\n    reason: RecordingInterruptionReason\n    isPaused: boolean\n}\n\nexport interface RecordingConfig {\n    // Sample rate for recording (16000, 44100, or 48000 Hz)\n    sampleRate?: SampleRate\n\n    // Number of audio channels (1 for mono, 2 for stereo)\n    channels?: 1 | 2\n\n    // Encoding type for the recording (pcm_32bit, pcm_16bit, pcm_8bit)\n    encoding?: EncodingType\n\n    // Interval in milliseconds at which to emit recording data\n    interval?: number\n\n    // Interval in milliseconds at which to emit analysis data\n    intervalAnalysis?: number\n\n    // Keep the device awake while recording (default is false)\n    keepAwake?: boolean\n\n    // Show a notification during recording (default is false)\n    showNotification?: boolean\n\n    // Show waveform in the notification (Android only, when showNotification is true)\n    showWaveformInNotification?: boolean\n\n    // Configuration for the notification\n    notification?: NotificationConfig\n\n    // Enable audio processing (default is false)\n    enableProcessing?: boolean\n\n    // iOS-specific configuration\n    ios?: IOSConfig\n\n    // Number of data points to extract per second of audio (default is 1000)\n    pointsPerSecond?: number\n\n    // Algorithm to use for amplitude computation (default is \"rms\")\n    algorithm?: AmplitudeAlgorithm\n\n    // Feature options to extract (default is empty)\n    features?: AudioFeaturesOptions\n\n    // Callback function to handle audio stream\n    onAudioStream?: (_: AudioDataEvent) => Promise<void>\n\n    // Callback function to handle audio features extraction results\n    onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>\n\n    compression?: {\n        enabled: boolean\n        format: 'aac' | 'opus'\n        bitrate?: number\n    }\n\n    // Whether to automatically resume recording after an interruption (default is false)\n    autoResumeAfterInterruption?: boolean\n\n    // Optional callback to handle recording interruptions\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n\n    // Optional output configuration\n    outputDirectory?: string // If not provided, uses default app directory\n    filename?: string // If not provided, uses UUID\n}\n\nexport interface NotificationConfig {\n    // Title of the notification\n    title?: string\n\n    // Main text content of the notification\n    text?: string\n\n    // Icon to be displayed in the notification (resource name or URI)\n    icon?: string\n\n    // Android-specific notification configuration\n    android?: {\n        // Unique identifier for the notification channel\n        channelId?: string\n\n        // User-visible name of the notification channel\n        channelName?: string\n\n        // User-visible description of the notification channel\n        channelDescription?: string\n\n        // Unique identifier for this notification\n        notificationId?: number\n\n        // List of actions that can be performed from the notification\n        actions?: NotificationAction[]\n\n        // Configuration for the waveform visualization in the notification\n        waveform?: WaveformConfig\n\n        // Color of the notification LED (if device supports it)\n        lightColor?: string\n\n        // Priority of the notification (affects how it's displayed)\n        priority?: 'min' | 'low' | 'default' | 'high' | 'max'\n\n        // Accent color for the notification (used for the app icon and buttons)\n        accentColor?: string\n    }\n\n    // iOS-specific notification configuration\n    ios?: {\n        // Identifier for the notification category (used for grouping similar notifications)\n        categoryIdentifier?: string\n    }\n}\n\nexport interface NotificationAction {\n    // Display title for the action\n    title: string\n\n    // Unique identifier for the action\n    identifier: string\n\n    // Icon to be displayed for the action (Android only)\n    icon?: string\n}\n\nexport interface WaveformConfig {\n    color?: string // The color of the waveform (e.g., \"#FFFFFF\" for white)\n    opacity?: number // Opacity of the waveform (0.0 - 1.0)\n    strokeWidth?: number // Width of the waveform line (default: 1.5)\n    style?: 'stroke' | 'fill' // Drawing style: \"stroke\" for outline, \"fill\" for solid\n    mirror?: boolean // Whether to mirror the waveform (symmetrical display)\n    height?: number // Height of the waveform view in dp (default: 64)\n}\n\nexport interface UseAudioRecorderState {\n    startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n    stopRecording: () => Promise<AudioRecording | null>\n    pauseRecording: () => Promise<void>\n    resumeRecording: () => Promise<void>\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number // Duration of the recording\n    size: number // Size in bytes of the recorded audio\n    compression?: CompressionInfo\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n"]}
+{"version":3,"file":"ExpoAudioStream.types.js","sourceRoot":"","sources":["../src/ExpoAudioStream.types.ts"],"names":[],"mappings":"","sourcesContent":["// packages/expo-audio-stream/src/ExpoAudioStream.types.ts\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DecodingConfig,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport { AudioAnalysisEvent } from './events'\n\nexport interface CompressionInfo {\n    size: number\n    mimeType: string\n    bitrate: number\n    format: string\n    compressedFileUri?: string\n}\n\nexport interface AudioStreamStatus {\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number\n    size: number\n    interval: number\n    intervalAnalysis: number\n    mimeType: string\n    compression?: CompressionInfo\n}\n\nexport interface AudioDataEvent {\n    data: string | Float32Array\n    position: number\n    fileUri: string\n    eventDataSize: number\n    totalSize: number\n    compression?: CompressionInfo & {\n        data?: string | Blob // Base64 (native) or Float32Array (web) encoded compressed data chunk\n    }\n}\n\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\nexport type SampleRate = 16000 | 44100 | 48000\nexport type BitDepth = 8 | 16 | 32\nexport type PCMFormat = `pcm_${BitDepth}bit`\n\nexport type ConsoleLike = {\n    log: (message: string, ...args: unknown[]) => void\n    debug: (message: string, ...args: unknown[]) => void\n    info: (message: string, ...args: unknown[]) => void\n    warn: (message: string, ...args: unknown[]) => void\n    error: (message: string, ...args: unknown[]) => void\n}\n\nexport interface Chunk {\n    text: string\n    timestamp: [number, number | null]\n}\n\nexport interface TranscriberData {\n    id: string\n    isBusy: boolean\n    text: string\n    startTime: number\n    endTime: number\n    chunks: Chunk[]\n}\n\nexport interface AudioRecording {\n    fileUri: string\n    filename: string\n    durationMs: number\n    size: number\n    mimeType: string\n    channels: number\n    bitDepth: BitDepth\n    sampleRate: SampleRate\n    createdAt?: number\n    transcripts?: TranscriberData[]\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    compression?: CompressionInfo & {\n        compressedFileUri: string\n    }\n}\n\nexport interface StartRecordingResult {\n    fileUri: string\n    mimeType: string\n    channels?: number\n    bitDepth?: BitDepth\n    sampleRate?: SampleRate\n    compression?: CompressionInfo & {\n        compressedFileUri: string\n    }\n}\n\nexport interface AudioSessionConfig {\n    category?:\n        | 'Ambient'\n        | 'SoloAmbient'\n        | 'Playback'\n        | 'Record'\n        | 'PlayAndRecord'\n        | 'MultiRoute'\n    mode?:\n        | 'Default'\n        | 'VoiceChat'\n        | 'VideoChat'\n        | 'GameChat'\n        | 'VideoRecording'\n        | 'Measurement'\n        | 'MoviePlayback'\n        | 'SpokenAudio'\n    categoryOptions?: (\n        | 'MixWithOthers'\n        | 'DuckOthers'\n        | 'InterruptSpokenAudioAndMixWithOthers'\n        | 'AllowBluetooth'\n        | 'AllowBluetoothA2DP'\n        | 'AllowAirPlay'\n        | 'DefaultToSpeaker'\n    )[]\n}\n\nexport interface IOSConfig {\n    audioSession?: AudioSessionConfig\n}\n\n// Add new type for interruption reasons\nexport type RecordingInterruptionReason =\n    | 'audioFocusLoss'\n    | 'audioFocusGain'\n    | 'phoneCall'\n    | 'phoneCallEnded'\n    | 'recordingStopped'\n\n// Add new interface for interruption events\nexport interface RecordingInterruptionEvent {\n    reason: RecordingInterruptionReason\n    isPaused: boolean\n}\n\nexport interface RecordingConfig {\n    // Sample rate for recording (16000, 44100, or 48000 Hz)\n    sampleRate?: SampleRate\n\n    // Number of audio channels (1 for mono, 2 for stereo)\n    channels?: 1 | 2\n\n    // Encoding type for the recording (pcm_32bit, pcm_16bit, pcm_8bit)\n    encoding?: EncodingType\n\n    // Interval in milliseconds at which to emit recording data\n    interval?: number\n\n    // Interval in milliseconds at which to emit analysis data\n    intervalAnalysis?: number\n\n    // Keep the device awake while recording (default is false)\n    keepAwake?: boolean\n\n    // Show a notification during recording (default is false)\n    showNotification?: boolean\n\n    // Show waveform in the notification (Android only, when showNotification is true)\n    showWaveformInNotification?: boolean\n\n    // Configuration for the notification\n    notification?: NotificationConfig\n\n    // Enable audio processing (default is false)\n    enableProcessing?: boolean\n\n    // iOS-specific configuration\n    ios?: IOSConfig\n\n    // Duration of each segment in milliseconds (default: 100)\n    segmentDurationMs?: number\n\n    // Feature options to extract (default is empty)\n    features?: AudioFeaturesOptions\n\n    // Callback function to handle audio stream\n    onAudioStream?: (_: AudioDataEvent) => Promise<void>\n\n    // Callback function to handle audio features extraction results\n    onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>\n\n    compression?: {\n        enabled: boolean\n        format: 'aac' | 'opus'\n        bitrate?: number\n    }\n\n    // Whether to automatically resume recording after an interruption (default is false)\n    autoResumeAfterInterruption?: boolean\n\n    // Optional callback to handle recording interruptions\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n\n    // Optional output configuration\n    outputDirectory?: string // If not provided, uses default app directory\n    filename?: string // If not provided, uses UUID\n}\n\nexport interface NotificationConfig {\n    // Title of the notification\n    title?: string\n\n    // Main text content of the notification\n    text?: string\n\n    // Icon to be displayed in the notification (resource name or URI)\n    icon?: string\n\n    // Android-specific notification configuration\n    android?: {\n        // Unique identifier for the notification channel\n        channelId?: string\n\n        // User-visible name of the notification channel\n        channelName?: string\n\n        // User-visible description of the notification channel\n        channelDescription?: string\n\n        // Unique identifier for this notification\n        notificationId?: number\n\n        // List of actions that can be performed from the notification\n        actions?: NotificationAction[]\n\n        // Configuration for the waveform visualization in the notification\n        waveform?: WaveformConfig\n\n        // Color of the notification LED (if device supports it)\n        lightColor?: string\n\n        // Priority of the notification (affects how it's displayed)\n        priority?: 'min' | 'low' | 'default' | 'high' | 'max'\n\n        // Accent color for the notification (used for the app icon and buttons)\n        accentColor?: string\n    }\n\n    // iOS-specific notification configuration\n    ios?: {\n        // Identifier for the notification category (used for grouping similar notifications)\n        categoryIdentifier?: string\n    }\n}\n\nexport interface NotificationAction {\n    // Display title for the action\n    title: string\n\n    // Unique identifier for the action\n    identifier: string\n\n    // Icon to be displayed for the action (Android only)\n    icon?: string\n}\n\nexport interface WaveformConfig {\n    color?: string // The color of the waveform (e.g., \"#FFFFFF\" for white)\n    opacity?: number // Opacity of the waveform (0.0 - 1.0)\n    strokeWidth?: number // Width of the waveform line (default: 1.5)\n    style?: 'stroke' | 'fill' // Drawing style: \"stroke\" for outline, \"fill\" for solid\n    mirror?: boolean // Whether to mirror the waveform (symmetrical display)\n    height?: number // Height of the waveform view in dp (default: 64)\n}\n\nexport interface ExtractAudioDataOptions {\n    fileUri: string\n    // Time-based range (mutually exclusive with byte-based range)\n    startTimeMs?: number\n    endTimeMs?: number\n    // Byte-based range (mutually exclusive with time-based range)\n    position?: number\n    length?: number\n    /** Include normalized audio data in [-1, 1] range */\n    includeNormalizedData?: boolean\n    /** Include base64 encoded string representation of the audio data */\n    includeBase64Data?: boolean\n    /** Include WAV header in the PCM data (makes it a valid WAV file) */\n    includeWavHeader?: boolean\n    /** Logger for debugging - can pass console directly. */\n    logger?: ConsoleLike\n    /** Compute the checksum of the pcm data */\n    computeChecksum?: boolean\n    /** Target config for the normalized audio (Android and Web) */\n    decodingOptions?: DecodingConfig\n}\n\nexport interface ExtractedAudioData {\n    /** Raw PCM audio data */\n    pcmData: Uint8Array\n    /** Normalized audio data in [-1, 1] range (when includeNormalizedData is true) */\n    normalizedData?: Float32Array\n    /** Base64 encoded string representation of the audio data (when includeBase64Data is true) */\n    base64Data?: string\n    /** Sample rate in Hz (e.g., 44100, 48000) */\n    sampleRate: number\n    /** Number of audio channels (1 for mono, 2 for stereo) */\n    channels: number\n    /** Bits per sample (8, 16, or 32) */\n    bitDepth: BitDepth\n    /** Duration of the audio in milliseconds */\n    durationMs: number\n    /** PCM format identifier (e.g., \"pcm_16bit\") */\n    format: PCMFormat\n    /** Total number of audio samples per channel */\n    samples: number\n    /** Whether the pcmData includes a WAV header */\n    hasWavHeader?: boolean\n    /** CRC32 Checksum of pcm data */\n    checksum?: number\n}\n\nexport interface UseAudioRecorderState {\n    startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n    stopRecording: () => Promise<AudioRecording | null>\n    pauseRecording: () => Promise<void>\n    resumeRecording: () => Promise<void>\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number // Duration of the recording\n    size: number // Size in bytes of the recorded audio\n    compression?: CompressionInfo\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n"]}