npm - @siteed/expo-audio-studio - Versions diffs - 2.1.1 → 2.2.0 - Mend

@siteed/expo-audio-studio 2.1.1 → 2.2.0

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.

Files changed (30) hide show

package/CHANGELOG.md +12 -1
package/build/AudioAnalysis/extractAudioAnalysis.d.ts +6 -0
package/build/AudioAnalysis/extractAudioAnalysis.d.ts.map +1 -1
package/build/AudioAnalysis/extractAudioAnalysis.js +1 -8
package/build/AudioAnalysis/extractAudioAnalysis.js.map +1 -1
package/build/AudioAnalysis/extractMelSpectrogram.js.map +1 -1
package/build/ExpoAudioStreamModule.js +1 -1
package/build/ExpoAudioStreamModule.js.map +1 -1
package/build/utils/crc32.d.ts +7 -0
package/build/utils/crc32.d.ts.map +1 -0
package/build/utils/crc32.js +12 -0
package/build/utils/crc32.js.map +1 -0
package/build/utils/crc32.native.d.ts +2 -0
package/build/utils/crc32.native.d.ts.map +1 -0
package/build/utils/crc32.native.js +5 -0
package/build/utils/crc32.native.js.map +1 -0
package/build/utils/crc32.web.d.ts +4 -0
package/build/utils/crc32.web.d.ts.map +1 -0
package/build/utils/crc32.web.js +3 -0
package/build/utils/crc32.web.js.map +1 -0
package/ios/DecodingConfig.swift +19 -7
package/ios/ExpoAudioStream.podspec +1 -1
package/package.json +6 -3
package/src/AudioAnalysis/extractAudioAnalysis.ts +1 -2
package/src/AudioAnalysis/extractMelSpectrogram.ts +2 -2
package/src/ExpoAudioStreamModule.ts +1 -1
package/src/types/crc-32.d.ts +9 -0
package/src/utils/crc32.native.ts +4 -0
package/src/utils/crc32.ts +21 -0
package/src/utils/crc32.web.ts +3 -0

package/CHANGELOG.md CHANGED Viewed

@@ -8,6 +8,16 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 ## [Unreleased]
+## [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))
+- feat: Enhance Audio Processing and Transcription Capabilities in Playground App (#171) ([1ec6026](https://github.com/deeeed/expo-audio-stream/commit/1ec6026ff75fc3ff7122b5df72e8dcd15ce848bd))
+- feat: Enhance Essentia Integration with iOS and Android Build Improvements (#169) ([422fd50](https://github.com/deeeed/expo-audio-stream/commit/422fd501a5ec71f30df660d56559bc410084b797))
+- feat: Enhance Audio Analysis with Mel Spectrogram Comparison and Pipeline Support (#164) ([541e13c](https://github.com/deeeed/expo-audio-stream/commit/541e13c6e01b8ff9947bc69dc7c29ffed6d8ee07))
+- feat: Integrate Essentia Audio Analysis Library into React Native for Android (#163) ([4cac310](https://github.com/deeeed/expo-audio-stream/commit/4cac310e4af47ddda528dee0f2840e3a336c6823))
+- 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 +181,8 @@ 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@2.2.0...HEAD
+[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/build/AudioAnalysis/extractAudioAnalysis.d.ts CHANGED Viewed

@@ -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 CHANGED Viewed

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package/build/AudioAnalysis/extractAudioAnalysis.js CHANGED Viewed

@@ -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) {

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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"]}
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 { 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. 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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 CHANGED Viewed

	@@ -1 +1 @@
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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. 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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 CHANGED Viewed

@@ -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 CHANGED Viewed

	@@ -1 +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"]}
1	+ 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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 ADDED Viewed

@@ -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 ADDED Viewed

	@@ -0,0 +1 @@
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 ADDED Viewed

@@ -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 ADDED Viewed

@@ -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 ADDED Viewed

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

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

	@@ -0,0 +1 @@
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 ADDED Viewed

@@ -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 ADDED Viewed

	@@ -0,0 +1 @@
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 ADDED Viewed

@@ -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 ADDED Viewed

	@@ -0,0 +1 @@
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 ADDED Viewed

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

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

	@@ -0,0 +1 @@
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/DecodingConfig.swift CHANGED Viewed

@@ -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 CHANGED Viewed

@@ -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/package.json CHANGED Viewed

@@ -1,6 +1,6 @@
 {
   "name": "@siteed/expo-audio-studio",
-  "version": "2.1.1",
+  "version": "2.2.0",
   "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/src/AudioAnalysis/extractAudioAnalysis.ts CHANGED Viewed

@@ -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 CHANGED Viewed

@@ -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 CHANGED Viewed

@@ -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 ADDED Viewed

@@ -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 ADDED Viewed

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

package/src/utils/crc32.ts ADDED Viewed

@@ -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 ADDED Viewed

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