@siteed/expo-audio-studio 2.1.1 → 2.3.0

package/CHANGELOG.md

@@ -8,6 +8,17 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 ## [Unreleased]
 
 
+## [2.3.0] - 2025-03-29
+### Changed
+- fix: always generate a new UUID unless filename is provided (#182) ([f98a9a5](https://github.com/deeeed/expo-audio-stream/commit/f98a9a52393829e6c4a79aee3575fbfcc9416c19))
+- refactor(audio-studio): introduce constants for silence threshold and WAV header size (#188) ([e8aa329](https://github.com/deeeed/expo-audio-stream/commit/e8aa3298bd6ba029d38898360b7df26b3fd5485f))
+- docs: enhance installation and API reference documentation for phone call handling (#187) ([fcaece1](https://github.com/deeeed/expo-audio-stream/commit/fcaece18cf046d970b9659f3f12a19deb096bceb))
+## [2.2.0] - 2025-03-28
+### Changed
+- refactor(audio-studio): implement platform-specific CRC32 handling ([b61a3d7](https://github.com/deeeed/expo-audio-stream/commit/b61a3d743914e66888ec6cc4cb8e010ff1992698))
+- chore: update Expo dependencies and remove invalid design-system version ([16e5007](https://github.com/deeeed/expo-audio-stream/commit/16e50077690b55977c22fbcb08be75834146ff47))
+- fix: linting issues ([741589d](https://github.com/deeeed/expo-audio-stream/commit/741589d60485a2d049e7adf529d3fd2b999fa098))
+- chore(expo-audio-studio): release @siteed/expo-audio-studio@2.1.1 ([1b17ac6](https://github.com/deeeed/expo-audio-stream/commit/1b17ac6e103f2ca50f29668b3ddaaf57a4b4b7d3))
 ## [2.1.1] - 2025-03-04
 ### Changed
 - feat: Rename`@siteed/expo-audio-stream` to `@siteed/expo-audio-studio` (#160) ([1b99191](https://github.com/deeeed/expo-audio-stream/commit/1b9919143413a900aefed94c20fc9a8b0e6050d3))
@@ -171,7 +182,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - Feature: Audio features extraction during recording.
 - Feature: Consistent WAV PCM recording format across all platforms.
 
-[unreleased]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-studio@2.1.1...HEAD
+[unreleased]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-studio@2.3.0...HEAD
+[2.3.0]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-studio@2.2.0...@siteed/expo-audio-studio@2.3.0
+[2.2.0]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-studio@2.1.1...@siteed/expo-audio-studio@2.2.0
 [2.1.1]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-studio@2.1.0...@siteed/expo-audio-studio@2.1.1
 [2.1.0]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-stream@2.0.1...@siteed/expo-audio-stream@2.1.0
 [2.0.1]: https://github.com/deeeed/expo-audio-stream/compare/@siteed/expo-audio-stream@2.0.0...@siteed/expo-audio-stream@2.0.1

package/android/src/main/AndroidManifest.xml

@@ -1,6 +1,6 @@
+<?xml version="1.0" encoding="utf-8"?>
 <manifest xmlns:android="http://schemas.android.com/apk/res/android">
     <!-- Permissions will be merged into the app's manifest -->
-    <uses-permission android:name="android.permission.READ_PHONE_STATE" />
     <uses-permission android:name="android.permission.RECORD_AUDIO"/>
     <uses-permission android:name="android.permission.FOREGROUND_SERVICE"/>
     <uses-permission android:name="android.permission.FOREGROUND_SERVICE_MICROPHONE"/>

package/android/src/main/java/net/siteed/audiostream/AudioRecorderManager.kt

@@ -37,7 +37,8 @@ class AudioRecorderManager(
     private val filesDir: File,
     private val permissionUtils: PermissionUtils,
     private val audioDataEncoder: AudioDataEncoder,
-    private val eventSender: EventSender
+    private val eventSender: EventSender,
+    private val enablePhoneStateHandling: Boolean = true
 ) {
     private var audioRecord: AudioRecord? = null
     private var bufferSizeInBytes = 0
@@ -348,8 +349,10 @@ class AudioRecorderManager(
     @RequiresApi(Build.VERSION_CODES.R)
     fun startRecording(options: Map<String, Any?>, promise: Promise) {
         try {
-            // Initialize phone state listener
-            initializePhoneStateListener()
+            // Initialize phone state listener only if enabled
+            if (enablePhoneStateHandling) {
+                initializePhoneStateListener()
+            }
 
             // Request audio focus
             if (!requestAudioFocus()) {
@@ -479,7 +482,8 @@ class AudioRecorderManager(
             return false
         }
 
-        if (!permissionUtils.checkPhoneStatePermission()) {
+        // Only check phone state permission if enabled
+        if (enablePhoneStateHandling && !permissionUtils.checkPhoneStatePermission()) {
             Log.w(Constants.TAG, "READ_PHONE_STATE permission not granted, phone call interruption handling will be disabled")
             // Don't reject here, just log warning as this is optional
         }

package/android/src/main/java/net/siteed/audiostream/ExpoAudioStreamModule.kt

@@ -16,6 +16,7 @@ import java.util.zip.CRC32
 class ExpoAudioStreamModule : Module(), EventSender {
     private lateinit var audioRecorderManager: AudioRecorderManager
     private lateinit var audioProcessor: AudioProcessor
+    private var enablePhoneStateHandling: Boolean = true // Default to true for backward compatibility
 
     private val audioFileHandler by lazy { 
         AudioFileHandler(appContext.reactContext?.filesDir ?: throw IllegalStateException("React context not available")) 
@@ -174,10 +175,14 @@ class ExpoAudioStreamModule : Module(), EventSender {
         AsyncFunction("requestPermissionsAsync") { promise: Promise ->
             try {
                 val permissions = mutableListOf(
-                    Manifest.permission.RECORD_AUDIO,
-                    Manifest.permission.READ_PHONE_STATE
+                    Manifest.permission.RECORD_AUDIO
                 )
 
+                // Only add phone state permission if enabled
+                if (enablePhoneStateHandling) {
+                    permissions.add(Manifest.permission.READ_PHONE_STATE)
+                }
+
                 // Add foreground service permission for Android 14+
                 if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.UPSIDE_DOWN_CAKE) {
                     Log.d(Constants.TAG, "Adding FOREGROUND_SERVICE_MICROPHONE permission request")
@@ -198,10 +203,14 @@ class ExpoAudioStreamModule : Module(), EventSender {
 
         AsyncFunction("getPermissionsAsync") { promise: Promise ->
             val permissions = mutableListOf(
-                Manifest.permission.RECORD_AUDIO,
-                Manifest.permission.READ_PHONE_STATE
+                Manifest.permission.RECORD_AUDIO
             )
 
+            // Only add phone state permission if enabled
+            if (enablePhoneStateHandling) {
+                permissions.add(Manifest.permission.READ_PHONE_STATE)
+            }
+
             if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.UPSIDE_DOWN_CAKE) {
                 permissions.add(Manifest.permission.FOREGROUND_SERVICE_MICROPHONE)
             }
@@ -714,20 +723,18 @@ class ExpoAudioStreamModule : Module(), EventSender {
     }
 
     private fun initializeManager() {
-        val androidContext =
-            appContext.reactContext ?: throw IllegalStateException("Android context not available")
-        val permissionUtils = PermissionUtils(androidContext)
-        val audioEncoder = AudioDataEncoder()
-        audioRecorderManager =
-            AudioRecorderManager(androidContext, androidContext.filesDir, permissionUtils, audioEncoder, this)
-        audioRecorderManager = AudioRecorderManager.initialize(
-            androidContext,
-            androidContext.filesDir,
+        val filesDir = appContext.reactContext?.filesDir ?: throw IllegalStateException("React context not available")
+        val permissionUtils = PermissionUtils(appContext.reactContext!!)
+        val audioDataEncoder = AudioDataEncoder()
+        audioRecorderManager = AudioRecorderManager(
+            appContext.reactContext!!,
+            filesDir,
             permissionUtils,
-            audioEncoder,
-            this
+            audioDataEncoder,
+            this,
+            enablePhoneStateHandling
         )
-        audioProcessor = AudioProcessor(androidContext.filesDir)
+        audioProcessor = AudioProcessor(filesDir)
     }
 
 

package/build/AudioAnalysis/extractAudioAnalysis.d.ts

@@ -1,3 +1,9 @@
+/**
+ * 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 { ConsoleLike } from '../ExpoAudioStream.types';
 import { AudioAnalysis, AudioFeaturesOptions, DecodingConfig } from './AudioAnalysis.types';
 import { WavFileInfo } from '../utils/getWavFileInfo';

package/build/AudioAnalysis/extractAudioAnalysis.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"extractAudioAnalysis.d.ts","sourceRoot":"","sources":["../../src/AudioAnalysis/extractAudioAnalysis.ts"],"names":[],"mappings":"AACA;;;;;GAKG;AACH,OAAO,EAAE,WAAW,EAAE,MAAM,0BAA0B,CAAA;AAGtD,OAAO,EACH,aAAa,EACb,oBAAoB,EAEpB,cAAc,EACjB,MAAM,uBAAuB,CAAA;AAI9B,OAAO,EAAkB,WAAW,EAAE,MAAM,yBAAyB,CAAA;AAerE,MAAM,WAAW,4BAA4B;IACzC,OAAO,CAAC,EAAE,MAAM,CAAA;IAChB,WAAW,CAAC,EAAE,WAAW,CAAA;IACzB,WAAW,CAAC,EAAE,WAAW,CAAA;IACzB,QAAQ,CAAC,EAAE,MAAM,CAAA;IACjB,UAAU,CAAC,EAAE,MAAM,CAAA;IACnB,UAAU,CAAC,EAAE,MAAM,CAAA;IACnB,gBAAgB,CAAC,EAAE,MAAM,CAAA;IACzB,QAAQ,CAAC,EAAE,MAAM,CAAA;IACjB,MAAM,CAAC,EAAE,MAAM,CAAA;IACf,iBAAiB,CAAC,EAAE,MAAM,CAAA;IAC1B,QAAQ,CAAC,EAAE,oBAAoB,CAAA;IAC/B,mBAAmB,CAAC,EAAE,MAAM,CAAA;IAC5B,MAAM,CAAC,EAAE,WAAW,CAAA;IACpB,eAAe,CAAC,EAAE,cAAc,CAAA;CACnC;AAGD,UAAU,kBAAkB;IACxB,OAAO,CAAC,EAAE,MAAM,CAAA;IAChB,WAAW,CAAC,EAAE,WAAW,CAAA;IACzB;;OAEG;IACH,iBAAiB,CAAC,EAAE,MAAM,CAAA;IAC1B,QAAQ,CAAC,EAAE,oBAAoB,CAAA;IAC/B,eAAe,CAAC,EAAE,cAAc,CAAA;IAChC,MAAM,CAAC,EAAE,WAAW,CAAA;CACvB;AAGD,UAAU,gBAAiB,SAAQ,kBAAkB;IACjD,WAAW,CAAC,EAAE,MAAM,CAAA;IACpB,SAAS,CAAC,EAAE,MAAM,CAAA;IAClB,QAAQ,CAAC,EAAE,KAAK,CAAA;IAChB,MAAM,CAAC,EAAE,KAAK,CAAA;CACjB;AAGD,UAAU,gBAAiB,SAAQ,kBAAkB;IACjD,QAAQ,CAAC,EAAE,MAAM,CAAA;IACjB,MAAM,CAAC,EAAE,MAAM,CAAA;IACf,WAAW,CAAC,EAAE,KAAK,CAAA;IACnB,SAAS,CAAC,EAAE,KAAK,CAAA;CACpB;AAED;;;;;GAKG;AACH,MAAM,MAAM,yBAAyB,GAAG,gBAAgB,GAAG,gBAAgB,CAAA;AAE3E;;;;;;;GAOG;AACH,wBAAsB,oBAAoB,CACtC,KAAK,EAAE,yBAAyB,GACjC,OAAO,CAAC,aAAa,CAAC,CAkHxB;AAED;;;;;;GAMG;AACH,eAAO,MAAM,qBAAqB,yIAY/B,4BAA4B,KAAG,QAAQ,aAAa,CAkGtD,CAAA"}

package/build/AudioAnalysis/extractAudioAnalysis.js

@@ -1,15 +1,8 @@
-// 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 crc32 from '../utils/crc32';
 import { getWavFileInfo } from '../utils/getWavFileInfo';
 import { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web';
 function calculateCRC32ForDataPoint(data) {

package/build/AudioAnalysis/extractAudioAnalysis.js.map

@@ -1 +1 @@
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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 } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\nimport { isWeb } from '../constants'\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DataPoint,\n    DecodingConfig,\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"]}
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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 { ConsoleLike } from '../ExpoAudioStream.types'\nimport ExpoAudioStreamModule from '../ExpoAudioStreamModule'\nimport { isWeb } from '../constants'\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DataPoint,\n    DecodingConfig,\n} from './AudioAnalysis.types'\nimport { processAudioBuffer } from '../utils/audioProcessing'\nimport { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'\nimport crc32 from '../utils/crc32'\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"]}

package/build/AudioAnalysis/extractMelSpectrogram.js.map

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The web implementation is a placeholder.\n */\n\nimport { ExpoAudioStreamModule } from '..'\nimport { isWeb } from '../constants'\nimport {\n    ExtractMelSpectrogramOptions,\n    MelSpectrogram,\n} from './AudioAnalysis.types'\nimport {\n    processAudioBuffer,\n    ProcessedAudioData,\n} from '../utils/audioProcessing'\n\n/**\n * Extracts a mel spectrogram from audio data\n *\n * @experimental This feature is experimental and currently only available on Android.\n * The iOS implementation will throw an \"UNSUPPORTED_PLATFORM\" error.\n * The web implementation is a placeholder that returns dummy data.\n */\nexport async function extractMelSpectrogram(\n    options: ExtractMelSpectrogramOptions\n): Promise<MelSpectrogram> {\n    const {\n        fileUri,\n        arrayBuffer,\n        windowSizeMs,\n        hopLengthMs,\n        nMels,\n        fMin = 0,\n        fMax,\n        windowType = 'hann',\n        normalize = false,\n        logScale = true,\n        decodingOptions,\n        startTimeMs,\n        endTimeMs,\n        logger,\n    } = options\n\n    if (isWeb) {\n        // Create audio context\n        const audioContext = new (window.AudioContext ||\n            (window as any).webkitAudioContext)()\n\n        try {\n            // Process audio data using the existing utility\n            const processedAudio: ProcessedAudioData = await processAudioBuffer(\n                {\n                    arrayBuffer,\n                    fileUri,\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate || 16000,\n                    targetChannels: decodingOptions?.targetChannels || 1,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                    startTimeMs,\n                    endTimeMs,\n                    audioContext,\n                    logger: options.logger,\n                }\n            )\n\n            // Calculate window and hop size in samples\n            const sampleRate = processedAudio.sampleRate\n            const windowSize = Math.floor((windowSizeMs * sampleRate) / 1000)\n            const hopLength = Math.floor((hopLengthMs * sampleRate) / 1000)\n            const maxFreq = fMax || sampleRate / 2\n\n            // Extract the mel spectrogram from the processed audio\n            const spectrogram = computeMelSpectrogram(\n                processedAudio.channelData,\n                sampleRate,\n                nMels,\n                windowSize,\n                hopLength,\n                fMin,\n                maxFreq,\n                windowType,\n                normalize,\n                logScale\n            )\n\n            const timeSteps = spectrogram.length\n\n            return {\n                spectrogram,\n                sampleRate,\n                nMels,\n                timeSteps,\n                durationMs: processedAudio.durationMs,\n            }\n        } catch (error) {\n            logger?.error('Error extracting mel spectrogram:', error)\n            throw error\n        } finally {\n            // Close the audio context\n            await audioContext.close()\n        }\n    }\n    return ExpoAudioStreamModule.extractMelSpectrogram(options)\n}\n\n/**\n * Computes a mel spectrogram from audio data\n *\n * @experimental This is a placeholder implementation that returns dummy data.\n * The actual implementation will be added in a future version.\n */\nfunction computeMelSpectrogram(\n    audioData: Float32Array,\n    sampleRate: number,\n    nMels: number,\n    windowSize: number,\n    hopLength: number,\n    fMin: number,\n    fMax: number,\n    windowType: 'hann' | 'hamming',\n    normalize: boolean,\n    logScale: boolean\n): number[][] {\n    // Placeholder for the actual implementation\n    // This would include:\n    // 1. Windowing the audio data using the specified window type\n    // 2. Computing the STFT (Short-Time Fourier Transform)\n    // 3. Converting to power spectrogram\n    // 4. Applying mel filterbanks\n    // 5. Taking the logarithm if logScale is true\n    // 6. Normalizing if normalize is true\n\n    // For now, return a dummy implementation\n    const numFrames =\n        Math.floor((audioData.length - windowSize) / hopLength) + 1\n    const spectrogram: number[][] = []\n\n    // Create dummy mel spectrogram data\n    for (let i = 0; i < numFrames; i++) {\n        spectrogram.push(Array(nMels).fill(0))\n    }\n\n    return spectrogram\n}\n"]}

package/build/ExpoAudioStreamModule.js

@@ -1,8 +1,8 @@
-import crc32 from 'crc-32';
 import { requireNativeModule } from 'expo-modules-core';
 import { Platform } from 'react-native';
 import { ExpoAudioStreamWeb, } from './ExpoAudioStream.web';
 import { processAudioBuffer } from './utils/audioProcessing';
+import crc32 from './utils/crc32';
 import { writeWavHeader } from './utils/writeWavHeader';
 // eslint-disable-next-line @typescript-eslint/no-explicit-any
 let ExpoAudioStreamModule;

package/build/ExpoAudioStreamModule.js.map

@@ -1 +1 @@
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crc32 from 'crc-32'\nimport { requireNativeModule } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport {\n    ExtractAudioDataOptions,\n    ExtractedAudioData,\n    BitDepth,\n    TrimAudioOptions,\n    TrimAudioResult,\n} from './ExpoAudioStream.types'\nimport {\n    ExpoAudioStreamWeb,\n    ExpoAudioStreamWebProps,\n} from './ExpoAudioStream.web'\nimport { processAudioBuffer } from './utils/audioProcessing'\nimport { writeWavHeader } from './utils/writeWavHeader'\n\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nlet ExpoAudioStreamModule: any\n\nif (Platform.OS === 'web') {\n    let instance: ExpoAudioStreamWeb | null = null\n\n    ExpoAudioStreamModule = (webProps: ExpoAudioStreamWebProps) => {\n        if (!instance) {\n            instance = new ExpoAudioStreamWeb(webProps)\n        }\n        return instance\n    }\n    ExpoAudioStreamModule.requestPermissionsAsync = async () => {\n        try {\n            const stream = await navigator.mediaDevices.getUserMedia({\n                audio: true,\n            })\n            stream.getTracks().forEach((track) => track.stop())\n            return {\n                status: 'granted',\n                expires: 'never',\n                canAskAgain: true,\n                granted: true,\n            }\n        } catch {\n            return {\n                status: 'denied',\n                expires: 'never',\n                canAskAgain: true,\n                granted: false,\n            }\n        }\n    }\n    ExpoAudioStreamModule.getPermissionsAsync = async () => {\n        let maybeStatus: string | null = null\n\n        if (navigator?.permissions?.query) {\n            try {\n                const { state } = await navigator.permissions.query({\n                    name: 'microphone' as PermissionName,\n                })\n                maybeStatus = state\n            } catch {\n                maybeStatus = null\n            }\n        }\n\n        switch (maybeStatus) {\n            case 'granted':\n                return {\n                    status: 'granted',\n                    expires: 'never',\n                    canAskAgain: true,\n                    granted: true,\n                }\n            case 'denied':\n                return {\n                    status: 'denied',\n                    expires: 'never',\n                    canAskAgain: true,\n                    granted: false,\n                }\n            default:\n                return await ExpoAudioStreamModule.requestPermissionsAsync()\n        }\n    }\n    ExpoAudioStreamModule.extractAudioData = async (\n        options: ExtractAudioDataOptions\n    ): Promise<ExtractedAudioData> => {\n        try {\n            const {\n                fileUri,\n                position,\n                length,\n                startTimeMs,\n                endTimeMs,\n                decodingOptions,\n                includeNormalizedData,\n                includeBase64Data,\n                includeWavHeader = false,\n                logger,\n            } = options\n\n            logger?.debug('EXTRACT AUDIO - Step 1: Initial request', {\n                fileUri,\n                extractionParams: {\n                    position,\n                    length,\n                    startTimeMs,\n                    endTimeMs,\n                },\n                decodingOptions: {\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate ?? 16000,\n                    targetChannels: decodingOptions?.targetChannels ?? 1,\n                    targetBitDepth: decodingOptions?.targetBitDepth ?? 16,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                },\n                outputOptions: {\n                    includeNormalizedData,\n                    includeBase64Data,\n                    includeWavHeader,\n                },\n            })\n\n            // Process the audio using shared helper function\n            const processedBuffer = await processAudioBuffer({\n                fileUri,\n                targetSampleRate: decodingOptions?.targetSampleRate ?? 16000,\n                targetChannels: decodingOptions?.targetChannels ?? 1,\n                normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                position,\n                length,\n                startTimeMs,\n                endTimeMs,\n                logger,\n            })\n\n            logger?.debug('EXTRACT AUDIO - Step 2: Audio processing complete', {\n                processedData: {\n                    samples: processedBuffer.samples,\n                    sampleRate: processedBuffer.sampleRate,\n                    channels: processedBuffer.channels,\n                    durationMs: processedBuffer.durationMs,\n                },\n            })\n\n            const channelData = processedBuffer.channelData\n            const bitDepth = (decodingOptions?.targetBitDepth ?? 16) as BitDepth\n            const bytesPerSample = bitDepth / 8\n            const numSamples = processedBuffer.samples\n\n            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion setup', {\n                channelData: {\n                    length: channelData.length,\n                    first: channelData[0],\n                    last: channelData[channelData.length - 1],\n                },\n                calculation: {\n                    bitDepth,\n                    bytesPerSample,\n                    numSamples,\n                    expectedBytes: numSamples * bytesPerSample,\n                },\n            })\n\n            // Create PCM data with correct length based on original byte length\n            const pcmData = new Uint8Array(numSamples * bytesPerSample)\n            let offset = 0\n\n            // Convert Float32 samples to PCM format\n            for (let i = 0; i < numSamples; i++) {\n                const sample = channelData[i]\n                const value = Math.max(-1, Math.min(1, sample))\n                // Convert to 16-bit signed integer\n                let intValue = Math.round(value * 32767)\n\n                // Handle negative values correctly\n                if (intValue < 0) {\n                    intValue = 65536 + intValue\n                }\n\n                // Write as little-endian\n                pcmData[offset++] = intValue & 255 // Low byte\n                pcmData[offset++] = (intValue >> 8) & 255 // High byte\n            }\n\n            const durationMs = Math.round(\n                (numSamples / processedBuffer.sampleRate) * 1000\n            )\n\n            logger?.debug('EXTRACT AUDIO - Step 4: Final output', {\n                pcmData: {\n                    length: pcmData.length,\n                    first: pcmData[0],\n                    last: pcmData[pcmData.length - 1],\n                },\n                timing: {\n                    numSamples,\n                    sampleRate: processedBuffer.sampleRate,\n                    durationMs,\n                    shouldBe3000ms: endTimeMs\n                        ? endTimeMs - (startTimeMs ?? 0) === 3000\n                        : undefined,\n                },\n            })\n\n            const result: ExtractedAudioData = {\n                pcmData: new Uint8Array(pcmData.buffer),\n                sampleRate: processedBuffer.sampleRate,\n                channels: processedBuffer.channels,\n                bitDepth,\n                durationMs,\n                format: `pcm_${bitDepth}bit` as const,\n                samples: numSamples,\n            }\n\n            // Add WAV header if requested\n            if (includeWavHeader) {\n                logger?.debug('EXTRACT AUDIO - Step 4: Adding WAV header', {\n                    originalLength: pcmData.length,\n                    newLength: result.pcmData.length,\n                    firstBytes: Array.from(result.pcmData.slice(0, 44)), // WAV header is 44 bytes\n                })\n                const wavBuffer = writeWavHeader({\n                    buffer: pcmData.buffer.slice(0, pcmData.length),\n                    sampleRate: processedBuffer.sampleRate,\n                    numChannels: processedBuffer.channels,\n                    bitDepth,\n                })\n                result.pcmData = new Uint8Array(wavBuffer)\n                result.hasWavHeader = true\n            }\n\n            if (includeNormalizedData) {\n                // // Simple approach: Create normalized data directly from the PCM data\n                // // Just convert to -1 to 1 range without any amplification\n                // const normalizedData = new Float32Array(numSamples)\n\n                // // Convert the PCM data to float values\n                // for (let i = 0; i < numSamples; i++) {\n                //     // Get the 16-bit PCM value (little endian)\n                //     const lowByte = pcmData[i * 2]\n                //     const highByte = pcmData[i * 2 + 1]\n                //     const pcmValue = (highByte << 8) | lowByte\n\n                //     // Convert to signed 16-bit value\n                //     const signedValue =\n                //         pcmValue > 32767 ? pcmValue - 65536 : pcmValue\n\n                //     // Normalize to float between -1 and 1\n                //     normalizedData[i] = signedValue / 32768.0\n                // }\n                // Store the normalized data in the result\n                result.normalizedData = channelData\n            }\n\n            if (includeBase64Data) {\n                // Convert the PCM data to a base64 string\n                const binary = Array.from(new Uint8Array(pcmData.buffer))\n                    .map((b) => String.fromCharCode(b))\n                    .join('')\n                result.base64Data = btoa(binary)\n            }\n\n            if (options.computeChecksum) {\n                result.checksum = crc32.buf(pcmData)\n            }\n\n            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion complete', {\n                pcmStats: {\n                    length: pcmData.length,\n                    bytesPerSample,\n                    totalSamples: numSamples,\n                    firstBytes: Array.from(pcmData.slice(0, 16)),\n                    lastBytes: Array.from(pcmData.slice(-16)),\n                },\n            })\n\n            return result\n        } catch (error) {\n            options.logger?.error('EXTRACT AUDIO - Error:', error)\n            throw error\n        }\n    }\n\n    ExpoAudioStreamModule.trimAudio = async (\n        options: TrimAudioOptions\n    ): Promise<TrimAudioResult> => {\n        try {\n            const startTime = performance.now()\n            const {\n                fileUri,\n                mode = 'single',\n                startTimeMs,\n                endTimeMs,\n                ranges,\n                outputFileName,\n                outputFormat,\n            } = options\n\n            // Validate inputs\n            if (!fileUri) {\n                throw new Error('fileUri is required')\n            }\n\n            if (\n                mode === 'single' &&\n                startTimeMs === undefined &&\n                endTimeMs === undefined\n            ) {\n                throw new Error(\n                    'At least one of startTimeMs or endTimeMs must be provided in single mode'\n                )\n            }\n\n            if (\n                (mode === 'keep' || mode === 'remove') &&\n                (!ranges || ranges.length === 0)\n            ) {\n                throw new Error(\n                    'ranges must be provided and non-empty for keep or remove modes'\n                )\n            }\n\n            // Create AudioContext\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)()\n\n            // First, load the entire audio file to get its properties\n            const response = await fetch(fileUri)\n            const arrayBuffer = await response.arrayBuffer()\n            const originalAudioBuffer =\n                await audioContext.decodeAudioData(arrayBuffer)\n\n            // Get original audio properties\n            const originalSampleRate = originalAudioBuffer.sampleRate\n            const originalChannels = originalAudioBuffer.numberOfChannels\n\n            // Add more detailed logging\n            console.log(`Original audio details:`, {\n                sampleRate: originalSampleRate,\n                channels: originalChannels,\n                duration: originalAudioBuffer.duration,\n                length: originalAudioBuffer.length,\n                // Log a few samples to verify content\n                firstSamples: Array.from(\n                    originalAudioBuffer.getChannelData(0).slice(0, 5)\n                ),\n            })\n\n            // Determine output format - use original values as defaults if not specified\n            let format = outputFormat?.format || 'wav'\n            const targetSampleRate =\n                outputFormat?.sampleRate || originalSampleRate\n            const targetChannels = outputFormat?.channels || originalChannels\n            const targetBitDepth = outputFormat?.bitDepth || 16\n\n            // Get file info from the URL\n            const filename =\n                outputFileName ||\n                fileUri.split('/').pop() ||\n                'trimmed-audio.wav'\n\n            // Process based on mode\n            let resultBuffer: AudioBuffer\n\n            // Report initial progress\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 10,\n            })\n\n            if (mode === 'single') {\n                // Single mode: extract a single range\n                // Use original sample rate and channels for extraction to preserve quality\n                const { buffer } = await processAudioBuffer({\n                    fileUri,\n                    targetSampleRate, // Use the requested sample rate\n                    targetChannels,\n                    normalizeAudio: false,\n                    startTimeMs,\n                    endTimeMs,\n                    audioContext,\n                })\n\n                console.log(`Processed buffer details:`, {\n                    sampleRate: buffer.sampleRate,\n                    channels: buffer.numberOfChannels,\n                    duration: buffer.duration,\n                    length: buffer.length,\n                    // Log a few samples to verify content\n                    firstSamples: Array.from(\n                        buffer.getChannelData(0).slice(0, 5)\n                    ),\n                })\n\n                resultBuffer = buffer\n\n                // If we need to change sample rate or channels, do it after extraction\n                if (\n                    targetSampleRate !== originalSampleRate ||\n                    targetChannels !== originalChannels\n                ) {\n                    console.log(\n                        `Resampling from ${originalSampleRate}Hz to ${targetSampleRate}Hz`\n                    )\n                    resultBuffer = await resampleAudioBuffer(\n                        audioContext,\n                        buffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            } else {\n                // For keep or remove modes\n                const fullDuration = originalAudioBuffer.duration * 1000 // in ms\n\n                type ProcessSegment = {\n                    startTimeMs: number\n                    endTimeMs: number\n                }\n\n                let segmentsToProcess: ProcessSegment[] = []\n\n                if (mode === 'keep') {\n                    // For keep mode, use the ranges directly\n                    segmentsToProcess = ranges!\n                } else {\n                    // mode === 'remove'\n                    // For remove mode, invert the ranges\n                    const sortedRanges = [...ranges!].sort(\n                        (a, b) => a.startTimeMs - b.startTimeMs\n                    )\n\n                    // Add segment from start to first range if needed\n                    if (\n                        sortedRanges.length > 0 &&\n                        sortedRanges[0].startTimeMs > 0\n                    ) {\n                        segmentsToProcess.push({\n                            startTimeMs: 0,\n                            endTimeMs: sortedRanges[0].startTimeMs,\n                        })\n                    }\n\n                    // Add segments between ranges\n                    for (let i = 0; i < sortedRanges.length - 1; i++) {\n                        segmentsToProcess.push({\n                            startTimeMs: sortedRanges[i].endTimeMs,\n                            endTimeMs: sortedRanges[i + 1].startTimeMs,\n                        })\n                    }\n\n                    // Add segment from last range to end if needed\n                    if (\n                        sortedRanges.length > 0 &&\n                        sortedRanges[sortedRanges.length - 1].endTimeMs <\n                            fullDuration\n                    ) {\n                        segmentsToProcess.push({\n                            startTimeMs:\n                                sortedRanges[sortedRanges.length - 1].endTimeMs,\n                            endTimeMs: fullDuration,\n                        })\n                    }\n                }\n\n                // Filter out empty or invalid segments\n                segmentsToProcess = segmentsToProcess.filter(\n                    (segment) =>\n                        segment.startTimeMs < segment.endTimeMs &&\n                        segment.endTimeMs - segment.startTimeMs > 1\n                ) // 1ms minimum\n\n                if (segmentsToProcess.length === 0) {\n                    throw new Error(\n                        'No valid segments to process after filtering ranges'\n                    )\n                }\n\n                // Process each segment using original sample rate and channels\n                const segmentBuffers: AudioBuffer[] = []\n\n                for (let i = 0; i < segmentsToProcess.length; i++) {\n                    const segment = segmentsToProcess[i]\n\n                    // Report progress for each segment\n                    ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                        progress:\n                            10 +\n                            Math.round((i / segmentsToProcess.length) * 40),\n                    })\n\n                    // Use processAudioBuffer to extract this segment\n                    const { buffer: segmentBuffer } = await processAudioBuffer({\n                        fileUri,\n                        targetSampleRate: originalSampleRate, // Use original sample rate\n                        targetChannels: originalChannels, // Use original channels\n                        normalizeAudio: false,\n                        startTimeMs: segment.startTimeMs,\n                        endTimeMs: segment.endTimeMs,\n                        audioContext,\n                    })\n\n                    segmentBuffers.push(segmentBuffer)\n                }\n\n                // Concatenate all segments\n                const totalSamples = segmentBuffers.reduce(\n                    (sum, buffer) => sum + buffer.length,\n                    0\n                )\n\n                // Create buffer with original properties first\n                const concatenatedBuffer = audioContext.createBuffer(\n                    originalChannels,\n                    totalSamples,\n                    originalSampleRate\n                )\n\n                let offset = 0\n                for (const segmentBuffer of segmentBuffers) {\n                    for (\n                        let channel = 0;\n                        channel < originalChannels;\n                        channel++\n                    ) {\n                        const outputData =\n                            concatenatedBuffer.getChannelData(channel)\n                        const segmentData =\n                            segmentBuffer.getChannelData(channel)\n\n                        for (let i = 0; i < segmentBuffer.length; i++) {\n                            outputData[offset + i] = segmentData[i]\n                        }\n                    }\n                    offset += segmentBuffer.length\n                }\n\n                resultBuffer = concatenatedBuffer\n\n                // If we need to change sample rate or channels, do it after concatenation\n                if (\n                    targetSampleRate !== originalSampleRate ||\n                    targetChannels !== originalChannels\n                ) {\n                    console.log(\n                        `Resampling concatenated buffer from ${originalSampleRate}Hz to ${targetSampleRate}Hz`\n                    )\n                    resultBuffer = await resampleAudioBuffer(\n                        audioContext,\n                        concatenatedBuffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            }\n\n            // Report progress (50% - processing complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 50,\n            })\n\n            // Encode the result based on the requested format\n            let outputData: ArrayBuffer\n            let outputMimeType: string\n            let compressionInfo: any = null\n\n            // Check if AAC was requested on web and show a warning\n            if (format === 'aac' && Platform.OS === 'web') {\n                console.warn(\n                    'AAC format is not supported on web platforms. Falling back to OPUS format.'\n                )\n                format = 'opus'\n            }\n\n            if (format === 'wav') {\n                // Create a properly interleaved buffer for WAV format\n                // For WAV, we need to convert Float32Array to Int16Array (for 16-bit audio)\n                const numSamples =\n                    resultBuffer.length * resultBuffer.numberOfChannels\n                const interleavedData = new Int16Array(numSamples)\n\n                // Log detailed information about the buffer before encoding\n                console.log(`Creating WAV file:`, {\n                    bufferSampleRate: resultBuffer.sampleRate,\n                    bufferChannels: resultBuffer.numberOfChannels,\n                    bufferLength: resultBuffer.length,\n                    targetSampleRate,\n                    targetChannels,\n                    targetBitDepth,\n                    // Log a few samples to verify content\n                    firstSamples: Array.from(\n                        resultBuffer.getChannelData(0).slice(0, 5)\n                    ),\n                })\n\n                // Interleave channels properly\n                for (let i = 0; i < resultBuffer.length; i++) {\n                    for (\n                        let channel = 0;\n                        channel < resultBuffer.numberOfChannels;\n                        channel++\n                    ) {\n                        // Convert float (-1.0 to 1.0) to int16 (-32768 to 32767)\n                        const floatSample =\n                            resultBuffer.getChannelData(channel)[i]\n                        // Clamp the value to -1.0 to 1.0\n                        const clampedSample = Math.max(\n                            -1.0,\n                            Math.min(1.0, floatSample)\n                        )\n                        // Convert to int16\n                        const intSample = Math.round(clampedSample * 32767)\n                        // Store in interleaved buffer\n                        interleavedData[\n                            i * resultBuffer.numberOfChannels + channel\n                        ] = intSample\n                    }\n                }\n\n                // Convert Int16Array to ArrayBuffer for WAV header\n                const rawBuffer = interleavedData.buffer\n\n                // IMPORTANT: Make sure we're using the ACTUAL sample rate of the buffer\n                // not just what was requested in the options\n                console.log(\n                    `Creating WAV with ${resultBuffer.numberOfChannels} channels at ${resultBuffer.sampleRate}Hz`\n                )\n\n                outputData = writeWavHeader({\n                    buffer: rawBuffer as ArrayBuffer,\n                    sampleRate: resultBuffer.sampleRate, // Use the actual buffer's sample rate\n                    numChannels: resultBuffer.numberOfChannels,\n                    bitDepth: targetBitDepth as BitDepth,\n                })\n                outputMimeType = 'audio/wav'\n            } else if (format === 'opus' || format === 'aac') {\n                try {\n                    // Try to use MediaRecorder for compressed formats\n                    const { data, bitrate } = await encodeCompressedAudio(\n                        resultBuffer,\n                        format,\n                        outputFormat?.bitrate\n                    )\n\n                    outputData = data\n                    outputMimeType =\n                        format === 'opus' ? 'audio/webm' : 'audio/aac'\n                    compressionInfo = {\n                        format,\n                        bitrate,\n                        size: data.byteLength,\n                    }\n                } catch (error) {\n                    console.warn(\n                        `Failed to encode to ${format}, falling back to WAV: ${error}`\n                    )\n\n                    // Same WAV encoding as above\n                    const wavData = new Float32Array(\n                        resultBuffer.length * resultBuffer.numberOfChannels\n                    )\n\n                    for (let i = 0; i < resultBuffer.length; i++) {\n                        for (\n                            let channel = 0;\n                            channel < resultBuffer.numberOfChannels;\n                            channel++\n                        ) {\n                            wavData[\n                                i * resultBuffer.numberOfChannels + channel\n                            ] = resultBuffer.getChannelData(channel)[i]\n                        }\n                    }\n\n                    outputData = writeWavHeader({\n                        buffer: wavData.buffer as ArrayBuffer,\n                        sampleRate: resultBuffer.sampleRate,\n                        numChannels: resultBuffer.numberOfChannels,\n                        bitDepth: targetBitDepth as BitDepth,\n                    })\n                    outputMimeType = 'audio/wav'\n                }\n            } else {\n                // Default to WAV for unsupported formats\n                console.warn(\n                    `Format ${format} not supported on web, using WAV instead`\n                )\n\n                // Same WAV encoding as above\n                const wavData = new Float32Array(\n                    resultBuffer.length * resultBuffer.numberOfChannels\n                )\n\n                for (let i = 0; i < resultBuffer.length; i++) {\n                    for (\n                        let channel = 0;\n                        channel < resultBuffer.numberOfChannels;\n                        channel++\n                    ) {\n                        wavData[i * resultBuffer.numberOfChannels + channel] =\n                            resultBuffer.getChannelData(channel)[i]\n                    }\n                }\n\n                outputData = writeWavHeader({\n                    buffer: wavData.buffer as ArrayBuffer,\n                    sampleRate: resultBuffer.sampleRate,\n                    numChannels: resultBuffer.numberOfChannels,\n                    bitDepth: targetBitDepth as BitDepth,\n                })\n                outputMimeType = 'audio/wav'\n            }\n\n            // Report progress (90% - encoding complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 90,\n            })\n\n            // Create a blob and URL for the result\n            const blob = new Blob([outputData], { type: outputMimeType })\n            const outputUri = URL.createObjectURL(blob)\n\n            // Calculate processing time\n            const processingTimeMs = performance.now() - startTime\n\n            // Report progress (100% - complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 100,\n            })\n\n            // Create result object\n            const result: TrimAudioResult = {\n                uri: outputUri,\n                filename,\n                durationMs: Math.round(resultBuffer.duration * 1000),\n                size: outputData.byteLength,\n                sampleRate: resultBuffer.sampleRate,\n                channels: resultBuffer.numberOfChannels,\n                bitDepth: targetBitDepth,\n                mimeType: outputMimeType,\n                processingInfo: {\n                    durationMs: processingTimeMs,\n                },\n            }\n\n            // Add compression info if available\n            if (compressionInfo) {\n                result.compression = compressionInfo\n            }\n\n            return result\n        } catch (error) {\n            console.error('Error in trimAudio:', error)\n            throw error\n        }\n    }\n\n    // Add a sendEvent method for web\n    ExpoAudioStreamModule.sendEvent = (eventName: string, params: any) => {\n        // This will be picked up by the LegacyEventEmitter in trimAudio.ts\n        if (\n            ExpoAudioStreamModule.listeners &&\n            ExpoAudioStreamModule.listeners[eventName]\n        ) {\n            ExpoAudioStreamModule.listeners[eventName].forEach(\n                (listener: Function) => {\n                    listener(params)\n                }\n            )\n        }\n    }\n\n    // Initialize listeners object\n    ExpoAudioStreamModule.listeners = {}\n\n    // Add methods for event listeners that LegacyEventEmitter will use\n    ExpoAudioStreamModule.addListener = (\n        eventName: string,\n        listener: Function\n    ) => {\n        if (!ExpoAudioStreamModule.listeners[eventName]) {\n            ExpoAudioStreamModule.listeners[eventName] = []\n        }\n        ExpoAudioStreamModule.listeners[eventName].push(listener)\n\n        // Return an object with a remove method\n        return {\n            remove: () => {\n                const index =\n                    ExpoAudioStreamModule.listeners[eventName].indexOf(listener)\n                if (index !== -1) {\n                    ExpoAudioStreamModule.listeners[eventName].splice(index, 1)\n                }\n            },\n        }\n    }\n\n    ExpoAudioStreamModule.removeAllListeners = (eventName: string) => {\n        if (ExpoAudioStreamModule.listeners[eventName]) {\n            delete ExpoAudioStreamModule.listeners[eventName]\n        }\n    }\n}\n\n// Move the encodeCompressedAudio function outside the if block to fix the ESLint error\nasync function encodeCompressedAudio(\n    buffer: AudioBuffer,\n    format: 'opus' | 'aac',\n    bitrate?: number\n): Promise<{ data: ArrayBuffer; bitrate: number }> {\n    return new Promise((resolve, reject) => {\n        try {\n            // On web, always use opus if aac is requested\n            const actualFormat =\n                Platform.OS === 'web' && format === 'aac' ? 'opus' : format\n\n            // Check if MediaRecorder supports the requested format\n            const mimeType =\n                actualFormat === 'opus' ? 'audio/webm;codecs=opus' : 'audio/aac'\n            if (!MediaRecorder.isTypeSupported(mimeType)) {\n                throw new Error(`MediaRecorder does not support ${mimeType}`)\n            }\n\n            // Create a new AudioContext and source\n            const ctx = new (window.AudioContext ||\n                (window as any).webkitAudioContext)()\n            const source = ctx.createBufferSource()\n            source.buffer = buffer\n\n            // Create a MediaStreamDestination to capture the audio\n            const destination = ctx.createMediaStreamDestination()\n            source.connect(destination)\n\n            // Create a MediaRecorder with the requested format\n            const recorder = new MediaRecorder(destination.stream, {\n                mimeType,\n                audioBitsPerSecond:\n                    bitrate || (actualFormat === 'opus' ? 32000 : 64000),\n            })\n\n            const chunks: Blob[] = []\n\n            recorder.ondataavailable = (e) => {\n                if (e.data.size > 0) {\n                    chunks.push(e.data)\n                }\n            }\n\n            recorder.onstop = async () => {\n                try {\n                    const blob = new Blob(chunks, { type: mimeType })\n                    const arrayBuffer = await blob.arrayBuffer()\n\n                    // Get the actual bitrate used\n                    const actualBitrate = Math.round(\n                        (arrayBuffer.byteLength * 8) / buffer.duration\n                    )\n\n                    resolve({\n                        data: arrayBuffer,\n                        bitrate: actualBitrate / 1000, // Convert to kbps\n                    })\n\n                    // Clean up\n                    ctx.close()\n                } catch (error) {\n                    reject(error)\n                }\n            }\n\n            // Start recording and playback\n            recorder.start()\n            source.start(0)\n\n            // Stop recording when the buffer finishes playing\n            setTimeout(() => {\n                recorder.stop()\n                source.stop()\n            }, buffer.duration * 1000)\n        } catch (error) {\n            reject(error)\n        }\n    })\n}\n\n// Improved resampleAudioBuffer function\nasync function resampleAudioBuffer(\n    context: AudioContext,\n    buffer: AudioBuffer,\n    targetSampleRate: number,\n    targetChannels: number\n): Promise<AudioBuffer> {\n    // If no change needed, return the original buffer\n    if (\n        buffer.sampleRate === targetSampleRate &&\n        buffer.numberOfChannels === targetChannels\n    ) {\n        return buffer\n    }\n\n    console.log(\n        `Resampling: ${buffer.sampleRate}Hz → ${targetSampleRate}Hz, ${buffer.numberOfChannels} → ${targetChannels} channels`\n    )\n\n    // Calculate the new length based on the sample rate change\n    const newLength = Math.round(\n        (buffer.length * targetSampleRate) / buffer.sampleRate\n    )\n\n    // Create an offline context for resampling\n    const offlineContext = new OfflineAudioContext(\n        targetChannels,\n        newLength,\n        targetSampleRate\n    )\n\n    // Create a source node\n    const source = offlineContext.createBufferSource()\n    source.buffer = buffer\n\n    // If we need to change channel count\n    if (buffer.numberOfChannels !== targetChannels) {\n        if (targetChannels === 1 && buffer.numberOfChannels > 1) {\n            // Downmix to mono\n            const merger = offlineContext.createChannelMerger(1)\n\n            // Create a gain node to reduce volume when downmixing to prevent clipping\n            const gainNode = offlineContext.createGain()\n            gainNode.gain.value = 1.0 / buffer.numberOfChannels\n\n            source.connect(gainNode)\n            gainNode.connect(merger)\n            merger.connect(offlineContext.destination)\n        } else if (targetChannels === 2 && buffer.numberOfChannels === 1) {\n            // Upmix mono to stereo (duplicate the channel)\n            const splitter = offlineContext.createChannelSplitter(1)\n            const merger = offlineContext.createChannelMerger(2)\n\n            source.connect(splitter)\n            splitter.connect(merger, 0, 0)\n            splitter.connect(merger, 0, 1)\n            merger.connect(offlineContext.destination)\n        } else {\n            // For other cases, just connect and let the system handle it\n            source.connect(offlineContext.destination)\n        }\n    } else {\n        // No channel conversion needed\n        source.connect(offlineContext.destination)\n    }\n\n    // Start rendering\n    source.start(0)\n    const resampledBuffer = await offlineContext.startRendering()\n\n    console.log(\n        `Resampling complete: ${resampledBuffer.length} samples at ${resampledBuffer.sampleRate}Hz`\n    )\n\n    return resampledBuffer\n}\n\nif (Platform.OS !== 'web') {\n    ExpoAudioStreamModule = requireNativeModule('ExpoAudioStream')\n}\n\nexport default ExpoAudioStreamModule\n"]}
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{ requireNativeModule } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport {\n    ExtractAudioDataOptions,\n    ExtractedAudioData,\n    BitDepth,\n    TrimAudioOptions,\n    TrimAudioResult,\n} from './ExpoAudioStream.types'\nimport {\n    ExpoAudioStreamWeb,\n    ExpoAudioStreamWebProps,\n} from './ExpoAudioStream.web'\nimport { processAudioBuffer } from './utils/audioProcessing'\nimport crc32 from './utils/crc32'\nimport { writeWavHeader } from './utils/writeWavHeader'\n\n// eslint-disable-next-line @typescript-eslint/no-explicit-any\nlet ExpoAudioStreamModule: any\n\nif (Platform.OS === 'web') {\n    let instance: ExpoAudioStreamWeb | null = null\n\n    ExpoAudioStreamModule = (webProps: ExpoAudioStreamWebProps) => {\n        if (!instance) {\n            instance = new ExpoAudioStreamWeb(webProps)\n        }\n        return instance\n    }\n    ExpoAudioStreamModule.requestPermissionsAsync = async () => {\n        try {\n            const stream = await navigator.mediaDevices.getUserMedia({\n                audio: true,\n            })\n            stream.getTracks().forEach((track) => track.stop())\n            return {\n                status: 'granted',\n                expires: 'never',\n                canAskAgain: true,\n                granted: true,\n            }\n        } catch {\n            return {\n                status: 'denied',\n                expires: 'never',\n                canAskAgain: true,\n                granted: false,\n            }\n        }\n    }\n    ExpoAudioStreamModule.getPermissionsAsync = async () => {\n        let maybeStatus: string | null = null\n\n        if (navigator?.permissions?.query) {\n            try {\n                const { state } = await navigator.permissions.query({\n                    name: 'microphone' as PermissionName,\n                })\n                maybeStatus = state\n            } catch {\n                maybeStatus = null\n            }\n        }\n\n        switch (maybeStatus) {\n            case 'granted':\n                return {\n                    status: 'granted',\n                    expires: 'never',\n                    canAskAgain: true,\n                    granted: true,\n                }\n            case 'denied':\n                return {\n                    status: 'denied',\n                    expires: 'never',\n                    canAskAgain: true,\n                    granted: false,\n                }\n            default:\n                return await ExpoAudioStreamModule.requestPermissionsAsync()\n        }\n    }\n    ExpoAudioStreamModule.extractAudioData = async (\n        options: ExtractAudioDataOptions\n    ): Promise<ExtractedAudioData> => {\n        try {\n            const {\n                fileUri,\n                position,\n                length,\n                startTimeMs,\n                endTimeMs,\n                decodingOptions,\n                includeNormalizedData,\n                includeBase64Data,\n                includeWavHeader = false,\n                logger,\n            } = options\n\n            logger?.debug('EXTRACT AUDIO - Step 1: Initial request', {\n                fileUri,\n                extractionParams: {\n                    position,\n                    length,\n                    startTimeMs,\n                    endTimeMs,\n                },\n                decodingOptions: {\n                    targetSampleRate:\n                        decodingOptions?.targetSampleRate ?? 16000,\n                    targetChannels: decodingOptions?.targetChannels ?? 1,\n                    targetBitDepth: decodingOptions?.targetBitDepth ?? 16,\n                    normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                },\n                outputOptions: {\n                    includeNormalizedData,\n                    includeBase64Data,\n                    includeWavHeader,\n                },\n            })\n\n            // Process the audio using shared helper function\n            const processedBuffer = await processAudioBuffer({\n                fileUri,\n                targetSampleRate: decodingOptions?.targetSampleRate ?? 16000,\n                targetChannels: decodingOptions?.targetChannels ?? 1,\n                normalizeAudio: decodingOptions?.normalizeAudio ?? false,\n                position,\n                length,\n                startTimeMs,\n                endTimeMs,\n                logger,\n            })\n\n            logger?.debug('EXTRACT AUDIO - Step 2: Audio processing complete', {\n                processedData: {\n                    samples: processedBuffer.samples,\n                    sampleRate: processedBuffer.sampleRate,\n                    channels: processedBuffer.channels,\n                    durationMs: processedBuffer.durationMs,\n                },\n            })\n\n            const channelData = processedBuffer.channelData\n            const bitDepth = (decodingOptions?.targetBitDepth ?? 16) as BitDepth\n            const bytesPerSample = bitDepth / 8\n            const numSamples = processedBuffer.samples\n\n            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion setup', {\n                channelData: {\n                    length: channelData.length,\n                    first: channelData[0],\n                    last: channelData[channelData.length - 1],\n                },\n                calculation: {\n                    bitDepth,\n                    bytesPerSample,\n                    numSamples,\n                    expectedBytes: numSamples * bytesPerSample,\n                },\n            })\n\n            // Create PCM data with correct length based on original byte length\n            const pcmData = new Uint8Array(numSamples * bytesPerSample)\n            let offset = 0\n\n            // Convert Float32 samples to PCM format\n            for (let i = 0; i < numSamples; i++) {\n                const sample = channelData[i]\n                const value = Math.max(-1, Math.min(1, sample))\n                // Convert to 16-bit signed integer\n                let intValue = Math.round(value * 32767)\n\n                // Handle negative values correctly\n                if (intValue < 0) {\n                    intValue = 65536 + intValue\n                }\n\n                // Write as little-endian\n                pcmData[offset++] = intValue & 255 // Low byte\n                pcmData[offset++] = (intValue >> 8) & 255 // High byte\n            }\n\n            const durationMs = Math.round(\n                (numSamples / processedBuffer.sampleRate) * 1000\n            )\n\n            logger?.debug('EXTRACT AUDIO - Step 4: Final output', {\n                pcmData: {\n                    length: pcmData.length,\n                    first: pcmData[0],\n                    last: pcmData[pcmData.length - 1],\n                },\n                timing: {\n                    numSamples,\n                    sampleRate: processedBuffer.sampleRate,\n                    durationMs,\n                    shouldBe3000ms: endTimeMs\n                        ? endTimeMs - (startTimeMs ?? 0) === 3000\n                        : undefined,\n                },\n            })\n\n            const result: ExtractedAudioData = {\n                pcmData: new Uint8Array(pcmData.buffer),\n                sampleRate: processedBuffer.sampleRate,\n                channels: processedBuffer.channels,\n                bitDepth,\n                durationMs,\n                format: `pcm_${bitDepth}bit` as const,\n                samples: numSamples,\n            }\n\n            // Add WAV header if requested\n            if (includeWavHeader) {\n                logger?.debug('EXTRACT AUDIO - Step 4: Adding WAV header', {\n                    originalLength: pcmData.length,\n                    newLength: result.pcmData.length,\n                    firstBytes: Array.from(result.pcmData.slice(0, 44)), // WAV header is 44 bytes\n                })\n                const wavBuffer = writeWavHeader({\n                    buffer: pcmData.buffer.slice(0, pcmData.length),\n                    sampleRate: processedBuffer.sampleRate,\n                    numChannels: processedBuffer.channels,\n                    bitDepth,\n                })\n                result.pcmData = new Uint8Array(wavBuffer)\n                result.hasWavHeader = true\n            }\n\n            if (includeNormalizedData) {\n                // // Simple approach: Create normalized data directly from the PCM data\n                // // Just convert to -1 to 1 range without any amplification\n                // const normalizedData = new Float32Array(numSamples)\n\n                // // Convert the PCM data to float values\n                // for (let i = 0; i < numSamples; i++) {\n                //     // Get the 16-bit PCM value (little endian)\n                //     const lowByte = pcmData[i * 2]\n                //     const highByte = pcmData[i * 2 + 1]\n                //     const pcmValue = (highByte << 8) | lowByte\n\n                //     // Convert to signed 16-bit value\n                //     const signedValue =\n                //         pcmValue > 32767 ? pcmValue - 65536 : pcmValue\n\n                //     // Normalize to float between -1 and 1\n                //     normalizedData[i] = signedValue / 32768.0\n                // }\n                // Store the normalized data in the result\n                result.normalizedData = channelData\n            }\n\n            if (includeBase64Data) {\n                // Convert the PCM data to a base64 string\n                const binary = Array.from(new Uint8Array(pcmData.buffer))\n                    .map((b) => String.fromCharCode(b))\n                    .join('')\n                result.base64Data = btoa(binary)\n            }\n\n            if (options.computeChecksum) {\n                result.checksum = crc32.buf(pcmData)\n            }\n\n            logger?.debug('EXTRACT AUDIO - Step 3: PCM conversion complete', {\n                pcmStats: {\n                    length: pcmData.length,\n                    bytesPerSample,\n                    totalSamples: numSamples,\n                    firstBytes: Array.from(pcmData.slice(0, 16)),\n                    lastBytes: Array.from(pcmData.slice(-16)),\n                },\n            })\n\n            return result\n        } catch (error) {\n            options.logger?.error('EXTRACT AUDIO - Error:', error)\n            throw error\n        }\n    }\n\n    ExpoAudioStreamModule.trimAudio = async (\n        options: TrimAudioOptions\n    ): Promise<TrimAudioResult> => {\n        try {\n            const startTime = performance.now()\n            const {\n                fileUri,\n                mode = 'single',\n                startTimeMs,\n                endTimeMs,\n                ranges,\n                outputFileName,\n                outputFormat,\n            } = options\n\n            // Validate inputs\n            if (!fileUri) {\n                throw new Error('fileUri is required')\n            }\n\n            if (\n                mode === 'single' &&\n                startTimeMs === undefined &&\n                endTimeMs === undefined\n            ) {\n                throw new Error(\n                    'At least one of startTimeMs or endTimeMs must be provided in single mode'\n                )\n            }\n\n            if (\n                (mode === 'keep' || mode === 'remove') &&\n                (!ranges || ranges.length === 0)\n            ) {\n                throw new Error(\n                    'ranges must be provided and non-empty for keep or remove modes'\n                )\n            }\n\n            // Create AudioContext\n            const audioContext = new (window.AudioContext ||\n                (window as any).webkitAudioContext)()\n\n            // First, load the entire audio file to get its properties\n            const response = await fetch(fileUri)\n            const arrayBuffer = await response.arrayBuffer()\n            const originalAudioBuffer =\n                await audioContext.decodeAudioData(arrayBuffer)\n\n            // Get original audio properties\n            const originalSampleRate = originalAudioBuffer.sampleRate\n            const originalChannels = originalAudioBuffer.numberOfChannels\n\n            // Add more detailed logging\n            console.log(`Original audio details:`, {\n                sampleRate: originalSampleRate,\n                channels: originalChannels,\n                duration: originalAudioBuffer.duration,\n                length: originalAudioBuffer.length,\n                // Log a few samples to verify content\n                firstSamples: Array.from(\n                    originalAudioBuffer.getChannelData(0).slice(0, 5)\n                ),\n            })\n\n            // Determine output format - use original values as defaults if not specified\n            let format = outputFormat?.format || 'wav'\n            const targetSampleRate =\n                outputFormat?.sampleRate || originalSampleRate\n            const targetChannels = outputFormat?.channels || originalChannels\n            const targetBitDepth = outputFormat?.bitDepth || 16\n\n            // Get file info from the URL\n            const filename =\n                outputFileName ||\n                fileUri.split('/').pop() ||\n                'trimmed-audio.wav'\n\n            // Process based on mode\n            let resultBuffer: AudioBuffer\n\n            // Report initial progress\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 10,\n            })\n\n            if (mode === 'single') {\n                // Single mode: extract a single range\n                // Use original sample rate and channels for extraction to preserve quality\n                const { buffer } = await processAudioBuffer({\n                    fileUri,\n                    targetSampleRate, // Use the requested sample rate\n                    targetChannels,\n                    normalizeAudio: false,\n                    startTimeMs,\n                    endTimeMs,\n                    audioContext,\n                })\n\n                console.log(`Processed buffer details:`, {\n                    sampleRate: buffer.sampleRate,\n                    channels: buffer.numberOfChannels,\n                    duration: buffer.duration,\n                    length: buffer.length,\n                    // Log a few samples to verify content\n                    firstSamples: Array.from(\n                        buffer.getChannelData(0).slice(0, 5)\n                    ),\n                })\n\n                resultBuffer = buffer\n\n                // If we need to change sample rate or channels, do it after extraction\n                if (\n                    targetSampleRate !== originalSampleRate ||\n                    targetChannels !== originalChannels\n                ) {\n                    console.log(\n                        `Resampling from ${originalSampleRate}Hz to ${targetSampleRate}Hz`\n                    )\n                    resultBuffer = await resampleAudioBuffer(\n                        audioContext,\n                        buffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            } else {\n                // For keep or remove modes\n                const fullDuration = originalAudioBuffer.duration * 1000 // in ms\n\n                type ProcessSegment = {\n                    startTimeMs: number\n                    endTimeMs: number\n                }\n\n                let segmentsToProcess: ProcessSegment[] = []\n\n                if (mode === 'keep') {\n                    // For keep mode, use the ranges directly\n                    segmentsToProcess = ranges!\n                } else {\n                    // mode === 'remove'\n                    // For remove mode, invert the ranges\n                    const sortedRanges = [...ranges!].sort(\n                        (a, b) => a.startTimeMs - b.startTimeMs\n                    )\n\n                    // Add segment from start to first range if needed\n                    if (\n                        sortedRanges.length > 0 &&\n                        sortedRanges[0].startTimeMs > 0\n                    ) {\n                        segmentsToProcess.push({\n                            startTimeMs: 0,\n                            endTimeMs: sortedRanges[0].startTimeMs,\n                        })\n                    }\n\n                    // Add segments between ranges\n                    for (let i = 0; i < sortedRanges.length - 1; i++) {\n                        segmentsToProcess.push({\n                            startTimeMs: sortedRanges[i].endTimeMs,\n                            endTimeMs: sortedRanges[i + 1].startTimeMs,\n                        })\n                    }\n\n                    // Add segment from last range to end if needed\n                    if (\n                        sortedRanges.length > 0 &&\n                        sortedRanges[sortedRanges.length - 1].endTimeMs <\n                            fullDuration\n                    ) {\n                        segmentsToProcess.push({\n                            startTimeMs:\n                                sortedRanges[sortedRanges.length - 1].endTimeMs,\n                            endTimeMs: fullDuration,\n                        })\n                    }\n                }\n\n                // Filter out empty or invalid segments\n                segmentsToProcess = segmentsToProcess.filter(\n                    (segment) =>\n                        segment.startTimeMs < segment.endTimeMs &&\n                        segment.endTimeMs - segment.startTimeMs > 1\n                ) // 1ms minimum\n\n                if (segmentsToProcess.length === 0) {\n                    throw new Error(\n                        'No valid segments to process after filtering ranges'\n                    )\n                }\n\n                // Process each segment using original sample rate and channels\n                const segmentBuffers: AudioBuffer[] = []\n\n                for (let i = 0; i < segmentsToProcess.length; i++) {\n                    const segment = segmentsToProcess[i]\n\n                    // Report progress for each segment\n                    ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                        progress:\n                            10 +\n                            Math.round((i / segmentsToProcess.length) * 40),\n                    })\n\n                    // Use processAudioBuffer to extract this segment\n                    const { buffer: segmentBuffer } = await processAudioBuffer({\n                        fileUri,\n                        targetSampleRate: originalSampleRate, // Use original sample rate\n                        targetChannels: originalChannels, // Use original channels\n                        normalizeAudio: false,\n                        startTimeMs: segment.startTimeMs,\n                        endTimeMs: segment.endTimeMs,\n                        audioContext,\n                    })\n\n                    segmentBuffers.push(segmentBuffer)\n                }\n\n                // Concatenate all segments\n                const totalSamples = segmentBuffers.reduce(\n                    (sum, buffer) => sum + buffer.length,\n                    0\n                )\n\n                // Create buffer with original properties first\n                const concatenatedBuffer = audioContext.createBuffer(\n                    originalChannels,\n                    totalSamples,\n                    originalSampleRate\n                )\n\n                let offset = 0\n                for (const segmentBuffer of segmentBuffers) {\n                    for (\n                        let channel = 0;\n                        channel < originalChannels;\n                        channel++\n                    ) {\n                        const outputData =\n                            concatenatedBuffer.getChannelData(channel)\n                        const segmentData =\n                            segmentBuffer.getChannelData(channel)\n\n                        for (let i = 0; i < segmentBuffer.length; i++) {\n                            outputData[offset + i] = segmentData[i]\n                        }\n                    }\n                    offset += segmentBuffer.length\n                }\n\n                resultBuffer = concatenatedBuffer\n\n                // If we need to change sample rate or channels, do it after concatenation\n                if (\n                    targetSampleRate !== originalSampleRate ||\n                    targetChannels !== originalChannels\n                ) {\n                    console.log(\n                        `Resampling concatenated buffer from ${originalSampleRate}Hz to ${targetSampleRate}Hz`\n                    )\n                    resultBuffer = await resampleAudioBuffer(\n                        audioContext,\n                        concatenatedBuffer,\n                        targetSampleRate,\n                        targetChannels\n                    )\n                }\n            }\n\n            // Report progress (50% - processing complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 50,\n            })\n\n            // Encode the result based on the requested format\n            let outputData: ArrayBuffer\n            let outputMimeType: string\n            let compressionInfo: any = null\n\n            // Check if AAC was requested on web and show a warning\n            if (format === 'aac' && Platform.OS === 'web') {\n                console.warn(\n                    'AAC format is not supported on web platforms. Falling back to OPUS format.'\n                )\n                format = 'opus'\n            }\n\n            if (format === 'wav') {\n                // Create a properly interleaved buffer for WAV format\n                // For WAV, we need to convert Float32Array to Int16Array (for 16-bit audio)\n                const numSamples =\n                    resultBuffer.length * resultBuffer.numberOfChannels\n                const interleavedData = new Int16Array(numSamples)\n\n                // Log detailed information about the buffer before encoding\n                console.log(`Creating WAV file:`, {\n                    bufferSampleRate: resultBuffer.sampleRate,\n                    bufferChannels: resultBuffer.numberOfChannels,\n                    bufferLength: resultBuffer.length,\n                    targetSampleRate,\n                    targetChannels,\n                    targetBitDepth,\n                    // Log a few samples to verify content\n                    firstSamples: Array.from(\n                        resultBuffer.getChannelData(0).slice(0, 5)\n                    ),\n                })\n\n                // Interleave channels properly\n                for (let i = 0; i < resultBuffer.length; i++) {\n                    for (\n                        let channel = 0;\n                        channel < resultBuffer.numberOfChannels;\n                        channel++\n                    ) {\n                        // Convert float (-1.0 to 1.0) to int16 (-32768 to 32767)\n                        const floatSample =\n                            resultBuffer.getChannelData(channel)[i]\n                        // Clamp the value to -1.0 to 1.0\n                        const clampedSample = Math.max(\n                            -1.0,\n                            Math.min(1.0, floatSample)\n                        )\n                        // Convert to int16\n                        const intSample = Math.round(clampedSample * 32767)\n                        // Store in interleaved buffer\n                        interleavedData[\n                            i * resultBuffer.numberOfChannels + channel\n                        ] = intSample\n                    }\n                }\n\n                // Convert Int16Array to ArrayBuffer for WAV header\n                const rawBuffer = interleavedData.buffer\n\n                // IMPORTANT: Make sure we're using the ACTUAL sample rate of the buffer\n                // not just what was requested in the options\n                console.log(\n                    `Creating WAV with ${resultBuffer.numberOfChannels} channels at ${resultBuffer.sampleRate}Hz`\n                )\n\n                outputData = writeWavHeader({\n                    buffer: rawBuffer as ArrayBuffer,\n                    sampleRate: resultBuffer.sampleRate, // Use the actual buffer's sample rate\n                    numChannels: resultBuffer.numberOfChannels,\n                    bitDepth: targetBitDepth as BitDepth,\n                })\n                outputMimeType = 'audio/wav'\n            } else if (format === 'opus' || format === 'aac') {\n                try {\n                    // Try to use MediaRecorder for compressed formats\n                    const { data, bitrate } = await encodeCompressedAudio(\n                        resultBuffer,\n                        format,\n                        outputFormat?.bitrate\n                    )\n\n                    outputData = data\n                    outputMimeType =\n                        format === 'opus' ? 'audio/webm' : 'audio/aac'\n                    compressionInfo = {\n                        format,\n                        bitrate,\n                        size: data.byteLength,\n                    }\n                } catch (error) {\n                    console.warn(\n                        `Failed to encode to ${format}, falling back to WAV: ${error}`\n                    )\n\n                    // Same WAV encoding as above\n                    const wavData = new Float32Array(\n                        resultBuffer.length * resultBuffer.numberOfChannels\n                    )\n\n                    for (let i = 0; i < resultBuffer.length; i++) {\n                        for (\n                            let channel = 0;\n                            channel < resultBuffer.numberOfChannels;\n                            channel++\n                        ) {\n                            wavData[\n                                i * resultBuffer.numberOfChannels + channel\n                            ] = resultBuffer.getChannelData(channel)[i]\n                        }\n                    }\n\n                    outputData = writeWavHeader({\n                        buffer: wavData.buffer as ArrayBuffer,\n                        sampleRate: resultBuffer.sampleRate,\n                        numChannels: resultBuffer.numberOfChannels,\n                        bitDepth: targetBitDepth as BitDepth,\n                    })\n                    outputMimeType = 'audio/wav'\n                }\n            } else {\n                // Default to WAV for unsupported formats\n                console.warn(\n                    `Format ${format} not supported on web, using WAV instead`\n                )\n\n                // Same WAV encoding as above\n                const wavData = new Float32Array(\n                    resultBuffer.length * resultBuffer.numberOfChannels\n                )\n\n                for (let i = 0; i < resultBuffer.length; i++) {\n                    for (\n                        let channel = 0;\n                        channel < resultBuffer.numberOfChannels;\n                        channel++\n                    ) {\n                        wavData[i * resultBuffer.numberOfChannels + channel] =\n                            resultBuffer.getChannelData(channel)[i]\n                    }\n                }\n\n                outputData = writeWavHeader({\n                    buffer: wavData.buffer as ArrayBuffer,\n                    sampleRate: resultBuffer.sampleRate,\n                    numChannels: resultBuffer.numberOfChannels,\n                    bitDepth: targetBitDepth as BitDepth,\n                })\n                outputMimeType = 'audio/wav'\n            }\n\n            // Report progress (90% - encoding complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 90,\n            })\n\n            // Create a blob and URL for the result\n            const blob = new Blob([outputData], { type: outputMimeType })\n            const outputUri = URL.createObjectURL(blob)\n\n            // Calculate processing time\n            const processingTimeMs = performance.now() - startTime\n\n            // Report progress (100% - complete)\n            ExpoAudioStreamModule.sendEvent('TrimProgress', {\n                progress: 100,\n            })\n\n            // Create result object\n            const result: TrimAudioResult = {\n                uri: outputUri,\n                filename,\n                durationMs: Math.round(resultBuffer.duration * 1000),\n                size: outputData.byteLength,\n                sampleRate: resultBuffer.sampleRate,\n                channels: resultBuffer.numberOfChannels,\n                bitDepth: targetBitDepth,\n                mimeType: outputMimeType,\n                processingInfo: {\n                    durationMs: processingTimeMs,\n                },\n            }\n\n            // Add compression info if available\n            if (compressionInfo) {\n                result.compression = compressionInfo\n            }\n\n            return result\n        } catch (error) {\n            console.error('Error in trimAudio:', error)\n            throw error\n        }\n    }\n\n    // Add a sendEvent method for web\n    ExpoAudioStreamModule.sendEvent = (eventName: string, params: any) => {\n        // This will be picked up by the LegacyEventEmitter in trimAudio.ts\n        if (\n            ExpoAudioStreamModule.listeners &&\n            ExpoAudioStreamModule.listeners[eventName]\n        ) {\n            ExpoAudioStreamModule.listeners[eventName].forEach(\n                (listener: Function) => {\n                    listener(params)\n                }\n            )\n        }\n    }\n\n    // Initialize listeners object\n    ExpoAudioStreamModule.listeners = {}\n\n    // Add methods for event listeners that LegacyEventEmitter will use\n    ExpoAudioStreamModule.addListener = (\n        eventName: string,\n        listener: Function\n    ) => {\n        if (!ExpoAudioStreamModule.listeners[eventName]) {\n            ExpoAudioStreamModule.listeners[eventName] = []\n        }\n        ExpoAudioStreamModule.listeners[eventName].push(listener)\n\n        // Return an object with a remove method\n        return {\n            remove: () => {\n                const index =\n                    ExpoAudioStreamModule.listeners[eventName].indexOf(listener)\n                if (index !== -1) {\n                    ExpoAudioStreamModule.listeners[eventName].splice(index, 1)\n                }\n            },\n        }\n    }\n\n    ExpoAudioStreamModule.removeAllListeners = (eventName: string) => {\n        if (ExpoAudioStreamModule.listeners[eventName]) {\n            delete ExpoAudioStreamModule.listeners[eventName]\n        }\n    }\n}\n\n// Move the encodeCompressedAudio function outside the if block to fix the ESLint error\nasync function encodeCompressedAudio(\n    buffer: AudioBuffer,\n    format: 'opus' | 'aac',\n    bitrate?: number\n): Promise<{ data: ArrayBuffer; bitrate: number }> {\n    return new Promise((resolve, reject) => {\n        try {\n            // On web, always use opus if aac is requested\n            const actualFormat =\n                Platform.OS === 'web' && format === 'aac' ? 'opus' : format\n\n            // Check if MediaRecorder supports the requested format\n            const mimeType =\n                actualFormat === 'opus' ? 'audio/webm;codecs=opus' : 'audio/aac'\n            if (!MediaRecorder.isTypeSupported(mimeType)) {\n                throw new Error(`MediaRecorder does not support ${mimeType}`)\n            }\n\n            // Create a new AudioContext and source\n            const ctx = new (window.AudioContext ||\n                (window as any).webkitAudioContext)()\n            const source = ctx.createBufferSource()\n            source.buffer = buffer\n\n            // Create a MediaStreamDestination to capture the audio\n            const destination = ctx.createMediaStreamDestination()\n            source.connect(destination)\n\n            // Create a MediaRecorder with the requested format\n            const recorder = new MediaRecorder(destination.stream, {\n                mimeType,\n                audioBitsPerSecond:\n                    bitrate || (actualFormat === 'opus' ? 32000 : 64000),\n            })\n\n            const chunks: Blob[] = []\n\n            recorder.ondataavailable = (e) => {\n                if (e.data.size > 0) {\n                    chunks.push(e.data)\n                }\n            }\n\n            recorder.onstop = async () => {\n                try {\n                    const blob = new Blob(chunks, { type: mimeType })\n                    const arrayBuffer = await blob.arrayBuffer()\n\n                    // Get the actual bitrate used\n                    const actualBitrate = Math.round(\n                        (arrayBuffer.byteLength * 8) / buffer.duration\n                    )\n\n                    resolve({\n                        data: arrayBuffer,\n                        bitrate: actualBitrate / 1000, // Convert to kbps\n                    })\n\n                    // Clean up\n                    ctx.close()\n                } catch (error) {\n                    reject(error)\n                }\n            }\n\n            // Start recording and playback\n            recorder.start()\n            source.start(0)\n\n            // Stop recording when the buffer finishes playing\n            setTimeout(() => {\n                recorder.stop()\n                source.stop()\n            }, buffer.duration * 1000)\n        } catch (error) {\n            reject(error)\n        }\n    })\n}\n\n// Improved resampleAudioBuffer function\nasync function resampleAudioBuffer(\n    context: AudioContext,\n    buffer: AudioBuffer,\n    targetSampleRate: number,\n    targetChannels: number\n): Promise<AudioBuffer> {\n    // If no change needed, return the original buffer\n    if (\n        buffer.sampleRate === targetSampleRate &&\n        buffer.numberOfChannels === targetChannels\n    ) {\n        return buffer\n    }\n\n    console.log(\n        `Resampling: ${buffer.sampleRate}Hz → ${targetSampleRate}Hz, ${buffer.numberOfChannels} → ${targetChannels} channels`\n    )\n\n    // Calculate the new length based on the sample rate change\n    const newLength = Math.round(\n        (buffer.length * targetSampleRate) / buffer.sampleRate\n    )\n\n    // Create an offline context for resampling\n    const offlineContext = new OfflineAudioContext(\n        targetChannels,\n        newLength,\n        targetSampleRate\n    )\n\n    // Create a source node\n    const source = offlineContext.createBufferSource()\n    source.buffer = buffer\n\n    // If we need to change channel count\n    if (buffer.numberOfChannels !== targetChannels) {\n        if (targetChannels === 1 && buffer.numberOfChannels > 1) {\n            // Downmix to mono\n            const merger = offlineContext.createChannelMerger(1)\n\n            // Create a gain node to reduce volume when downmixing to prevent clipping\n            const gainNode = offlineContext.createGain()\n            gainNode.gain.value = 1.0 / buffer.numberOfChannels\n\n            source.connect(gainNode)\n            gainNode.connect(merger)\n            merger.connect(offlineContext.destination)\n        } else if (targetChannels === 2 && buffer.numberOfChannels === 1) {\n            // Upmix mono to stereo (duplicate the channel)\n            const splitter = offlineContext.createChannelSplitter(1)\n            const merger = offlineContext.createChannelMerger(2)\n\n            source.connect(splitter)\n            splitter.connect(merger, 0, 0)\n            splitter.connect(merger, 0, 1)\n            merger.connect(offlineContext.destination)\n        } else {\n            // For other cases, just connect and let the system handle it\n            source.connect(offlineContext.destination)\n        }\n    } else {\n        // No channel conversion needed\n        source.connect(offlineContext.destination)\n    }\n\n    // Start rendering\n    source.start(0)\n    const resampledBuffer = await offlineContext.startRendering()\n\n    console.log(\n        `Resampling complete: ${resampledBuffer.length} samples at ${resampledBuffer.sampleRate}Hz`\n    )\n\n    return resampledBuffer\n}\n\nif (Platform.OS !== 'web') {\n    ExpoAudioStreamModule = requireNativeModule('ExpoAudioStream')\n}\n\nexport default ExpoAudioStreamModule\n"]}

package/build/utils/crc32.d.ts

@@ -0,0 +1,7 @@
+interface CRC32 {
+    (data: string | Uint8Array): number;
+    buf(data: Uint8Array): number;
+}
+declare let crc32Implementation: CRC32;
+export default crc32Implementation;
+//# sourceMappingURL=crc32.d.ts.map

package/build/utils/crc32.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"crc32.d.ts","sourceRoot":"","sources":["../../src/utils/crc32.ts"],"names":[],"mappings":"AAEA,UAAU,KAAK;IACX,CAAC,IAAI,EAAE,MAAM,GAAG,UAAU,GAAG,MAAM,CAAC;IACpC,GAAG,CAAC,IAAI,EAAE,UAAU,GAAG,MAAM,CAAC;CACjC;AAED,QAAA,IAAI,mBAAmB,EAAE,KAAK,CAAA;AAa9B,eAAe,mBAAmB,CAAA"}

package/build/utils/crc32.js

@@ -0,0 +1,12 @@
+import { Platform } from 'react-native';
+let crc32Implementation;
+if (Platform.OS === 'web') {
+    // eslint-disable-next-line @typescript-eslint/no-var-requires
+    crc32Implementation = require('crc-32');
+}
+else {
+    // No-op implementation for native platforms
+    crc32Implementation = Object.assign(() => 0, { buf: () => 0 });
+}
+export default crc32Implementation;
+//# sourceMappingURL=crc32.js.map

package/build/utils/crc32.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"crc32.js","sourceRoot":"","sources":["../../src/utils/crc32.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,QAAQ,EAAE,MAAM,cAAc,CAAA;AAOvC,IAAI,mBAA0B,CAAA;AAE9B,IAAI,QAAQ,CAAC,EAAE,KAAK,KAAK,EAAE,CAAC;IACxB,8DAA8D;IAC9D,mBAAmB,GAAG,OAAO,CAAC,QAAQ,CAAC,CAAA;AAC3C,CAAC;KAAM,CAAC;IACJ,4CAA4C;IAC5C,mBAAmB,GAAG,MAAM,CAAC,MAAM,CAC/B,GAAG,EAAE,CAAC,CAAC,EACP,EAAE,GAAG,EAAE,GAAG,EAAE,CAAC,CAAC,EAAE,CACnB,CAAA;AACL,CAAC;AAED,eAAe,mBAAmB,CAAA","sourcesContent":["import { Platform } from 'react-native'\n\ninterface CRC32 {\n    (data: string | Uint8Array): number;\n    buf(data: Uint8Array): number;\n}\n\nlet crc32Implementation: CRC32\n\nif (Platform.OS === 'web') {\n    // eslint-disable-next-line @typescript-eslint/no-var-requires\n    crc32Implementation = require('crc-32')\n} else {\n    // No-op implementation for native platforms\n    crc32Implementation = Object.assign(\n        () => 0,\n        { buf: () => 0 }\n    )\n}\n\nexport default crc32Implementation\n"]}

package/build/utils/crc32.native.d.ts

@@ -0,0 +1,2 @@
+export default function crc32(): number;
+//# sourceMappingURL=crc32.native.d.ts.map

package/build/utils/crc32.native.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"crc32.native.d.ts","sourceRoot":"","sources":["../../src/utils/crc32.native.ts"],"names":[],"mappings":"AACA,MAAM,CAAC,OAAO,UAAU,KAAK,IAAI,MAAM,CAEtC"}

package/build/utils/crc32.native.js

@@ -0,0 +1,5 @@
+// No-op implementation for native platforms
+export default function crc32() {
+    return 0;
+}
+//# sourceMappingURL=crc32.native.js.map

package/build/utils/crc32.native.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"crc32.native.js","sourceRoot":"","sources":["../../src/utils/crc32.native.ts"],"names":[],"mappings":"AAAA,4CAA4C;AAC5C,MAAM,CAAC,OAAO,UAAU,KAAK;IACzB,OAAO,CAAC,CAAC;AACb,CAAC","sourcesContent":["// No-op implementation for native platforms\nexport default function crc32(): number {\n    return 0;\n}"]}

package/build/utils/crc32.web.d.ts

@@ -0,0 +1,4 @@
+/// <reference path="../../src/types/crc-32.d.ts" />
+import crc32 from 'crc-32';
+export default crc32;
+//# sourceMappingURL=crc32.web.d.ts.map

package/build/utils/crc32.web.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"crc32.web.d.ts","sourceRoot":"","sources":["../../src/utils/crc32.web.ts"],"names":[],"mappings":";AAAA,OAAO,KAAK,MAAM,QAAQ,CAAA;AAE1B,eAAe,KAAK,CAAA"}

package/build/utils/crc32.web.js

@@ -0,0 +1,3 @@
+import crc32 from 'crc-32';
+export default crc32;
+//# sourceMappingURL=crc32.web.js.map

package/build/utils/crc32.web.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"crc32.web.js","sourceRoot":"","sources":["../../src/utils/crc32.web.ts"],"names":[],"mappings":"AAAA,OAAO,KAAK,MAAM,QAAQ,CAAA;AAE1B,eAAe,KAAK,CAAA","sourcesContent":["import crc32 from 'crc-32'\n\nexport default crc32\n"]}

package/ios/AudioProcessor.swift

@@ -5,6 +5,9 @@ import Accelerate
 import AVFoundation
 import QuartzCore
 
+// Constants
+private let SILENCE_THRESHOLD_RMS: Float = 0.01
+
 public struct TrimResult {
     let uri: String
     let filename: String
@@ -416,7 +419,7 @@ public class AudioProcessor {
     ) -> DataPoint {
         let sumSquares: Float = segment.reduce(0) { $0 + $1 * $1 }
         let rms = sqrt(sumSquares / Float(segment.count))
-        let silent = rms < 0.01
+        let silent = rms < SILENCE_THRESHOLD_RMS
         let dB = Float(20 * log10(Double(rms)))
         
         let features = computeFeatures(
@@ -590,7 +593,7 @@ public class AudioProcessor {
                     amplitude: localMax,      // Always use peak amplitude
                     rms: rms,                // Use calculated RMS value
                     dB: Float(20 * log10(Double(rms))),  // Use RMS for dB calculation
-                    silent: rms < 0.01,      // Use RMS for silence detection
+                    silent: rms < SILENCE_THRESHOLD_RMS,      // Use RMS for silence detection
                     features: computeFeatures(
                         segmentData: Array(UnsafeBufferPointer(start: summedData, count: Int(framesToRead))),
                         sampleRate: sampleRate,
@@ -599,7 +602,7 @@ public class AudioProcessor {
                         segmentLength: Int(framesToRead),
                         featureOptions: featureOptions
                     ),
-                    speech: SpeechFeatures(isActive: rms >= 0.01),
+                    speech: SpeechFeatures(isActive: rms >= SILENCE_THRESHOLD_RMS),
                     startTime: startTime,
                     endTime: endTime,
                     startPosition: Int(currentFrame),
@@ -1229,7 +1232,7 @@ public class AudioProcessor {
                                             featureOptions: featureOptions)
                 
                 let rms = features.rms
-                let silent = rms < 0.01
+                let silent = rms < SILENCE_THRESHOLD_RMS
                 let dB = Float(20 * log10(Double(rms)))
                 
                 let dataPoint = DataPoint(

package/ios/AudioStreamManager.swift

@@ -12,6 +12,9 @@ import UIKit
 import MediaPlayer
 import UserNotifications
 
+// Constants
+internal let WAV_HEADER_SIZE: Int64 = 44  // Standard WAV header is 44 bytes
+
 // Helper to convert to little-endian byte array
 extension UInt32 {
     var littleEndianBytes: [UInt8] {
@@ -458,9 +461,8 @@ class AudioStreamManager: NSObject {
         let baseFilename: String
         if let existingFilename = recordingSettings?.filename {
             baseFilename = existingFilename
-        } else if let existingUUID = recordingUUID {
-            baseFilename = existingUUID.uuidString
         } else {
+            // Always create a new UUID for recording unless a filename is provided
             let newUUID = UUID()
             recordingUUID = newUUID
             baseFilename = newUUID.uuidString
@@ -1096,17 +1098,17 @@ class AudioStreamManager: NSObject {
                 - Exists: true
                 - Size: \(wavFileSize) bytes
                 - Duration: \(finalDuration) seconds
-                - Expected minimum size: 44 bytes (WAV header)
+                - Expected minimum size: \(WAV_HEADER_SIZE) bytes (WAV header)
                 """)
             
             // Return nil if the file is too small
-            if wavFileSize <= 44 {
-                Logger.debug("Recording file is too small (≤ 44 bytes), likely no audio data was recorded")
+            if wavFileSize <= WAV_HEADER_SIZE {
+                Logger.debug("Recording file is too small (≤ \(WAV_HEADER_SIZE) bytes), likely no audio data was recorded")
                 return nil
             }
             
             // Update the WAV header with the correct file size
-            updateWavHeader(fileURL: fileURL, totalDataSize: wavFileSize - 44)
+            updateWavHeader(fileURL: fileURL, totalDataSize: wavFileSize - WAV_HEADER_SIZE)
             
             // Validate compressed file if enabled
             var compression: CompressedRecordingInfo?
@@ -1372,7 +1374,7 @@ class AudioStreamManager: NSObject {
     ///   - fileURL: The URL of the WAV file.
     ///   - totalDataSize: The total size of the audio data.
     private func updateWavHeader(fileURL: URL, totalDataSize: Int64) {
-        // Prevent negative values - minimum WAV file size should be at least the header size (44 bytes)
+        // Prevent negative values - minimum WAV file size should be at least the header size (WAV_HEADER_SIZE bytes)
         guard totalDataSize >= 0 else {
             Logger.debug("Invalid file size: total data size is negative")
             return
@@ -1383,7 +1385,7 @@ class AudioStreamManager: NSObject {
             defer { fileHandle.closeFile() }
 
             // Calculate sizes
-            let fileSize = totalDataSize + 44 - 8 // Total file size minus 8 bytes for 'RIFF' and size field itself
+            let fileSize = totalDataSize + WAV_HEADER_SIZE - 8 // Total file size minus 8 bytes for 'RIFF' and size field itself
             let dataSize = totalDataSize // Size of the 'data' sub-chunk
 
             // Update RIFF chunk size at offset 4
@@ -1585,8 +1587,19 @@ class AudioStreamManager: NSObject {
                     guard let self = self else { return }
                     if let processor = self.audioProcessor {
                         Logger.debug("Processing audio buffer of size: \(dataToProcess.count)")
+                        
+                        // Strip WAV header from the first buffer to avoid false amplitude detection
+                        let dataToAnalyze: Data
+                        if self.totalDataSizeAnalysis == 0 && dataToProcess.count > Int(WAV_HEADER_SIZE) {
+                            // This is the first buffer and may contain the WAV header
+                            dataToAnalyze = dataToProcess.subdata(in: Int(WAV_HEADER_SIZE)..<dataToProcess.count)
+                            Logger.debug("Removed WAV header (\(WAV_HEADER_SIZE) bytes) from first buffer for analysis")
+                        } else {
+                            dataToAnalyze = dataToProcess
+                        }
+                        
                         let processingResult = processor.processAudioBuffer(
-                            data: dataToProcess,
+                            data: dataToAnalyze,
                             sampleRate: Float(settings.sampleRate),
                             segmentDurationMs: settings.segmentDurationMs,
                             featureOptions: settings.featureOptions ?? [:],
@@ -1600,9 +1613,10 @@ class AudioStreamManager: NSObject {
                             }
                         }
 
-                            // Update state after emission
+                        // Update state after emission
                         self.lastEmissionTimeAnalysis = currentTime
-                        self.lastEmittedSizeAnalysis = totalDataSizeAnalysis
+                        // Update the total analysis data size to mark that we've processed data
+                        self.totalDataSizeAnalysis = self.totalDataSize
                         accumulatedAnalysisData.removeAll()
                     }
                 }

package/ios/DecodingConfig.swift

@@ -8,12 +8,24 @@
 import AVFoundation
 
 public struct DecodingConfig {
-    let targetSampleRate: Double?
-    let targetChannels: Int?
-    let targetBitDepth: Int?
-    let normalizeAudio: Bool
+    public let targetSampleRate: Double?
+    public let targetChannels: Int?
+    public let targetBitDepth: Int?
+    public let normalizeAudio: Bool
     
-    static func fromDictionary(_ dict: [String: Any]?) -> DecodingConfig {
+    public init(
+        targetSampleRate: Double?,
+        targetChannels: Int?,
+        targetBitDepth: Int?,
+        normalizeAudio: Bool = false
+    ) {
+        self.targetSampleRate = targetSampleRate
+        self.targetChannels = targetChannels
+        self.targetBitDepth = targetBitDepth
+        self.normalizeAudio = normalizeAudio
+    }
+    
+    public static func fromDictionary(_ dict: [String: Any]?) -> DecodingConfig {
         guard let dict = dict else {
             return DecodingConfig.default
         }
@@ -26,7 +38,7 @@ public struct DecodingConfig {
         )
     }
     
-    static var `default`: DecodingConfig {
+    public static var `default`: DecodingConfig {
         return DecodingConfig(
             targetSampleRate: nil,
             targetChannels: nil,
@@ -35,7 +47,7 @@ public struct DecodingConfig {
         )
     }
     
-    func toAudioFormat(baseFormat: AVAudioFormat) -> AVAudioFormat {
+    public func toAudioFormat(baseFormat: AVAudioFormat) -> AVAudioFormat {
         let sampleRate = targetSampleRate ?? baseFormat.sampleRate
         let channels = targetChannels ?? Int(baseFormat.channelCount)
         

package/ios/ExpoAudioStream.podspec

@@ -10,7 +10,7 @@ Pod::Spec.new do |s|
   s.license        = package['license']
   s.author         = package['author']
   s.homepage       = package['homepage']
-  s.platforms      = { :ios => '13.4', :tvos => '13.4' }
+  s.platforms      = { :ios => '15.1', :tvos => '15.1' }
   s.swift_version  = '5.4'
   s.source         = { git: 'https://github.com/deeeed/expo-audio-stream' }
   s.static_framework = true

package/ios/ExpoAudioStreamModule.swift

@@ -2,9 +2,11 @@
 import ExpoModulesCore
 import AVFoundation
 
-let audioDataEvent: String = "AudioData"
-let audioAnalysisEvent: String = "AudioAnalysis"
-let recordingInterruptedEvent: String = "onRecordingInterrupted"
+// Constants
+private let audioDataEvent: String = "AudioData"
+private let audioAnalysisEvent: String = "AudioAnalysis"
+private let recordingInterruptedEvent: String = "onRecordingInterrupted"
+private let DEFAULT_SEGMENT_DURATION_MS = 100
 
 public class ExpoAudioStreamModule: Module, AudioStreamManagerDelegate {
     private var streamManager = AudioStreamManager()
@@ -60,7 +62,7 @@ public class ExpoAudioStreamModule: Module, AudioStreamManagerDelegate {
             
             let features = options["features"] as? [String: Bool] ?? [:]
             let featureOptions = self.extractFeatureOptions(from: features)
-            let segmentDurationMs = options["segmentDurationMs"] as? Int ?? 100 // Default value of 100ms
+            let segmentDurationMs = options["segmentDurationMs"] as? Int ?? DEFAULT_SEGMENT_DURATION_MS // Default value of 100ms
             
             DispatchQueue.global().async(execute: {
                 do {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@siteed/expo-audio-studio",
-  "version": "2.1.1",
+  "version": "2.3.0",
   "description": "Comprehensive audio processing library for React Native and Expo with recording, analysis, visualization, and streaming capabilities across iOS, Android, and web",
   "license": "MIT",
   "main": "build/index.js",
@@ -116,13 +116,16 @@
     "react": "*",
     "react-native": "*"
   },
+  "peerDependenciesMeta": {
+    "crc-32": {
+      "optional": true
+    }
+  },
   "publishConfig": {
     "access": "public",
     "registry": "https://registry.npmjs.org"
   },
   "dependencies": {
-    "@siteed/design-system": "^0.35.1",
-    "crc-32": "^1.2.2",
     "expo-modules-core": "~2.1.4"
   }
 }

package/plugin/build/index.js

@@ -113,10 +113,9 @@ const withRecordingPermission = (config, props) => {
         ];
         const optionalPermissions = [
             enableNotifications && 'android.permission.POST_NOTIFICATIONS',
-            enablePhoneStateHandling && 'android.permission.READ_PHONE_STATE',
+            enablePhoneStateHandling && 'android.permission.READ_PHONE_STATE', // Only add if enabled
             enableBackgroundAudio && 'android.permission.FOREGROUND_SERVICE',
-            enableBackgroundAudio &&
-                'android.permission.FOREGROUND_SERVICE_MICROPHONE',
+            enableBackgroundAudio && 'android.permission.FOREGROUND_SERVICE_MICROPHONE',
         ].filter(Boolean);
         const permissionsToAdd = [...basePermissions, ...optionalPermissions];
         debugLog('📋 Existing Android permissions:', config.modResults.manifest['uses-permission']?.map((p) => p.$?.['android:name']) || []);

package/plugin/src/index.ts

@@ -169,10 +169,9 @@ const withRecordingPermission: ConfigPlugin<AudioStreamPluginOptions> = (
 
         const optionalPermissions = [
             enableNotifications && 'android.permission.POST_NOTIFICATIONS',
-            enablePhoneStateHandling && 'android.permission.READ_PHONE_STATE',
+            enablePhoneStateHandling && 'android.permission.READ_PHONE_STATE', // Only add if enabled
             enableBackgroundAudio && 'android.permission.FOREGROUND_SERVICE',
-            enableBackgroundAudio &&
-                'android.permission.FOREGROUND_SERVICE_MICROPHONE',
+            enableBackgroundAudio && 'android.permission.FOREGROUND_SERVICE_MICROPHONE',
         ].filter(Boolean) as string[]
 
         const permissionsToAdd = [...basePermissions, ...optionalPermissions]

package/src/AudioAnalysis/extractAudioAnalysis.ts

@@ -5,8 +5,6 @@
  * - `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 { ConsoleLike } from '../ExpoAudioStream.types'
 import ExpoAudioStreamModule from '../ExpoAudioStreamModule'
 import { isWeb } from '../constants'
@@ -18,6 +16,7 @@ import {
 } from './AudioAnalysis.types'
 import { processAudioBuffer } from '../utils/audioProcessing'
 import { convertPCMToFloat32 } from '../utils/convertPCMToFloat32'
+import crc32 from '../utils/crc32'
 import { getWavFileInfo, WavFileInfo } from '../utils/getWavFileInfo'
 import { InlineFeaturesExtractor } from '../workers/InlineFeaturesExtractor.web'
 

package/src/AudioAnalysis/extractMelSpectrogram.ts

@@ -16,7 +16,7 @@ import {
 
 /**
  * Extracts a mel spectrogram from audio data
- * 
+ *
  * @experimental This feature is experimental and currently only available on Android.
  * The iOS implementation will throw an "UNSUPPORTED_PLATFORM" error.
  * The web implementation is a placeholder that returns dummy data.
@@ -105,7 +105,7 @@ export async function extractMelSpectrogram(
 
 /**
  * Computes a mel spectrogram from audio data
- * 
+ *
  * @experimental This is a placeholder implementation that returns dummy data.
  * The actual implementation will be added in a future version.
  */

package/src/ExpoAudioStreamModule.ts

@@ -1,4 +1,3 @@
-import crc32 from 'crc-32'
 import { requireNativeModule } from 'expo-modules-core'
 import { Platform } from 'react-native'
 
@@ -14,6 +13,7 @@ import {
     ExpoAudioStreamWebProps,
 } from './ExpoAudioStream.web'
 import { processAudioBuffer } from './utils/audioProcessing'
+import crc32 from './utils/crc32'
 import { writeWavHeader } from './utils/writeWavHeader'
 
 // eslint-disable-next-line @typescript-eslint/no-explicit-any

package/src/types/crc-32.d.ts

@@ -0,0 +1,9 @@
+declare module 'crc-32' {
+    interface CRC32 {
+        (data: string | Uint8Array): number;
+        buf(data: Uint8Array): number;
+    }
+    
+    const crc32: CRC32;
+    export default crc32;
+} 

package/src/utils/crc32.native.ts

@@ -0,0 +1,4 @@
+// No-op implementation for native platforms
+export default function crc32(): number {
+    return 0;
+}

package/src/utils/crc32.ts

@@ -0,0 +1,21 @@
+import { Platform } from 'react-native'
+
+interface CRC32 {
+    (data: string | Uint8Array): number;
+    buf(data: Uint8Array): number;
+}
+
+let crc32Implementation: CRC32
+
+if (Platform.OS === 'web') {
+    // eslint-disable-next-line @typescript-eslint/no-var-requires
+    crc32Implementation = require('crc-32')
+} else {
+    // No-op implementation for native platforms
+    crc32Implementation = Object.assign(
+        () => 0,
+        { buf: () => 0 }
+    )
+}
+
+export default crc32Implementation

package/src/utils/crc32.web.ts

@@ -0,0 +1,3 @@
+import crc32 from 'crc-32'
+
+export default crc32