@siteed/expo-audio-studio 2.4.0 → 2.5.0

package/build/ExpoAudioStream.web.js.map

@@ -1 +1 @@
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src/ExpoAudioStreamModule.web.ts\nimport { LegacyEventEmitter } from 'expo-modules-core'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport {\n    AudioRecording,\n    AudioStreamStatus,\n    BitDepth,\n    ConsoleLike,\n    RecordingConfig,\n    StartRecordingResult,\n} from './ExpoAudioStream.types'\nimport { WebRecorder } from './WebRecorder.web'\nimport { AudioEventPayload } from './events'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\n\nexport interface EmitAudioEventProps {\n    data: Float32Array\n    position: number\n    compression?: {\n        data: Blob\n        size: number\n        totalSize: number\n        mimeType: string\n        format: string\n        bitrate: number\n    }\n}\nexport type EmitAudioEventFunction = (_: EmitAudioEventProps) => void\nexport type EmitAudioAnalysisFunction = (_: AudioAnalysis) => void\n\nexport interface ExpoAudioStreamWebProps {\n    logger?: ConsoleLike\n    audioWorkletUrl: string\n    featuresExtratorUrl: string\n    maxBufferSize?: number // Maximum number of chunks to keep in memory\n}\n\nexport class ExpoAudioStreamWeb extends LegacyEventEmitter {\n    customRecorder: WebRecorder | null\n    audioChunks: Float32Array[]\n    isRecording: boolean\n    isPaused: boolean\n    recordingStartTime: number\n    pausedTime: number\n    currentDurationMs: number\n    currentSize: number\n    currentInterval: number\n    currentIntervalAnalysis: number\n    lastEmittedSize: number\n    lastEmittedTime: number\n    lastEmittedCompressionSize: number\n    lastEmittedAnalysisTime: number\n    streamUuid: string | null\n    extension: 'webm' | 'wav' = 'wav' // Default extension is 'wav'\n    recordingConfig?: RecordingConfig\n    bitDepth: BitDepth // Bit depth of the audio\n    audioWorkletUrl: string\n    featuresExtratorUrl: string\n    logger?: ConsoleLike\n    latestPosition: number = 0\n    totalCompressedSize: number = 0\n    private readonly maxBufferSize: number\n\n    constructor({\n        audioWorkletUrl,\n        featuresExtratorUrl,\n        logger,\n        maxBufferSize = 100, // Default to storing last 100 chunks (1 chunk = 0.5 seconds)\n    }: ExpoAudioStreamWebProps) {\n        const mockNativeModule = {\n            addListener: () => {\n                // Not used on web\n            },\n            removeListeners: () => {\n                // Not used on web\n            },\n        }\n        super(mockNativeModule) // Pass the mock native module to the parent class\n\n        this.logger = logger\n        this.customRecorder = null\n        this.audioChunks = []\n        this.isRecording = false\n        this.isPaused = false\n        this.recordingStartTime = 0\n        this.pausedTime = 0\n        this.currentDurationMs = 0\n        this.currentSize = 0\n        this.bitDepth = 32 // Default\n        this.currentInterval = 1000 // Default interval in ms\n        this.currentIntervalAnalysis = 500 // Default analysis interval in ms\n        this.lastEmittedSize = 0\n        this.lastEmittedTime = 0\n        this.latestPosition = 0\n        this.lastEmittedCompressionSize = 0\n        this.lastEmittedAnalysisTime = 0\n        this.streamUuid = null // Initialize UUID on first recording start\n        this.audioWorkletUrl = audioWorkletUrl\n        this.featuresExtratorUrl = featuresExtratorUrl\n        this.maxBufferSize = maxBufferSize\n    }\n\n    // Utility to handle user media stream\n    async getMediaStream() {\n        try {\n            return await navigator.mediaDevices.getUserMedia({ audio: true })\n        } catch (error) {\n            this.logger?.error('Failed to get media stream:', error)\n            throw error\n        }\n    }\n\n    // Start recording with options\n    async startRecording(\n        recordingConfig: RecordingConfig = {}\n    ): Promise<StartRecordingResult> {\n        if (this.isRecording) {\n            throw new Error('Recording is already in progress')\n        }\n\n        this.bitDepth = encodingToBitDepth({\n            encoding: recordingConfig.encoding ?? 'pcm_32bit',\n        })\n\n        const audioContext = new (window.AudioContext ||\n            // eslint-disable-next-line @typescript-eslint/ban-ts-comment\n            // @ts-ignore - Allow webkitAudioContext for Safari\n            window.webkitAudioContext)()\n        const stream = await this.getMediaStream()\n\n        const source = audioContext.createMediaStreamSource(stream)\n\n        this.customRecorder = new WebRecorder({\n            logger: this.logger,\n            audioContext,\n            source,\n            recordingConfig,\n            emitAudioEventCallback: ({\n                data,\n                position,\n                compression,\n            }: EmitAudioEventProps) => {\n                // Keep only the latest chunks based on maxBufferSize\n                this.audioChunks.push(new Float32Array(data))\n                if (this.audioChunks.length > this.maxBufferSize) {\n                    this.audioChunks.shift() // Remove oldest chunk\n                }\n                this.currentSize += data.byteLength\n                this.emitAudioEvent({ data, position, compression })\n                this.lastEmittedTime = Date.now()\n                this.lastEmittedSize = this.currentSize\n                this.lastEmittedCompressionSize = compression?.size ?? 0\n            },\n            emitAudioAnalysisCallback: (audioAnalysisData: AudioAnalysis) => {\n                this.logger?.log(`Emitted AudioAnalysis:`, audioAnalysisData)\n                this.emit('AudioAnalysis', audioAnalysisData)\n            },\n        })\n        await this.customRecorder.init()\n        this.customRecorder.start()\n\n        // // Set a timer to stop recording after 5 seconds\n        // setTimeout(() => {\n        //   logger.log(\"AUTO Stopping recording\");\n        //   this.customRecorder?.stopAndPlay();\n        //   this.isRecording = false;\n        // }, 3000);\n\n        this.isRecording = true\n        this.recordingConfig = recordingConfig\n        this.recordingStartTime = Date.now()\n        this.pausedTime = 0\n        this.isPaused = false\n        this.lastEmittedSize = 0\n        this.lastEmittedTime = 0\n        this.lastEmittedCompressionSize = 0\n        this.currentInterval = recordingConfig.interval ?? 1000\n        this.currentIntervalAnalysis = recordingConfig.intervalAnalysis ?? 500\n        this.lastEmittedAnalysisTime = Date.now()\n\n        // Use custom filename if provided, otherwise fallback to timestamp\n        if (recordingConfig.filename) {\n            // Remove any existing extension from the filename\n            this.streamUuid = recordingConfig.filename.replace(/\\.[^/.]+$/, '')\n        } else {\n            this.streamUuid = Date.now().toString()\n        }\n\n        const fileUri = `${this.streamUuid}.${this.extension}`\n        const streamConfig: StartRecordingResult = {\n            fileUri,\n            mimeType: `audio/${this.extension}`,\n            bitDepth: this.bitDepth,\n            channels: recordingConfig.channels ?? 1,\n            sampleRate: recordingConfig.sampleRate ?? 44100,\n            compression: recordingConfig.compression\n                ? {\n                      ...recordingConfig.compression,\n                      bitrate: recordingConfig.compression?.bitrate ?? 128000,\n                      size: 0,\n                      mimeType: 'audio/webm',\n                      format: recordingConfig.compression?.format ?? 'opus',\n                      compressedFileUri: '',\n                  }\n                : undefined,\n        }\n        return streamConfig\n    }\n\n    emitAudioEvent({ data, position, compression }: EmitAudioEventProps) {\n        const fileUri = `${this.streamUuid}.${this.extension}`\n        if (compression?.size) {\n            this.lastEmittedCompressionSize = compression.size\n            this.totalCompressedSize = compression.totalSize\n        }\n        this.latestPosition = position\n        this.currentDurationMs = position * 1000 // Convert position (in seconds) to ms\n\n        const audioEventPayload: AudioEventPayload = {\n            fileUri,\n            mimeType: `audio/${this.extension}`,\n            lastEmittedSize: this.lastEmittedSize,\n            deltaSize: data.byteLength,\n            position,\n            totalSize: this.currentSize,\n            buffer: data,\n            streamUuid: this.streamUuid ?? '',\n            compression: compression\n                ? {\n                      data: compression?.data,\n                      totalSize: this.totalCompressedSize,\n                      eventDataSize: compression?.size ?? 0,\n                      position,\n                  }\n                : undefined,\n        }\n\n        this.emit('AudioData', audioEventPayload)\n    }\n\n    // Stop recording\n    async stopRecording(): Promise<AudioRecording> {\n        if (!this.customRecorder) {\n            throw new Error('Recorder is not initialized')\n        }\n\n        this.logger?.debug('[Stop] Starting stop process')\n        const startTime = performance.now()\n\n        try {\n            this.logger?.debug('[Stop] Stopping recorder')\n            const { compressedBlob } = await this.customRecorder.stop()\n\n            this.isRecording = false\n            this.isPaused = false\n            this.currentDurationMs = Date.now() - this.recordingStartTime\n\n            let compression: AudioRecording['compression']\n            let fileUri = `${this.streamUuid}.${this.extension}`\n            let mimeType = `audio/${this.extension}`\n\n            if (compressedBlob && this.recordingConfig?.compression?.enabled) {\n                const compressedUri = URL.createObjectURL(compressedBlob)\n                compression = {\n                    compressedFileUri: compressedUri,\n                    size: compressedBlob.size,\n                    mimeType: 'audio/webm',\n                    format: 'opus',\n                    bitrate: this.recordingConfig.compression.bitrate ?? 128000,\n                }\n                // Use compressed values when compression is enabled\n                fileUri = compressedUri\n                mimeType = 'audio/webm'\n            }\n\n            this.logger?.debug(\n                `[Stop] Completed stop process in ${performance.now() - startTime}ms`,\n                {\n                    durationMs: this.currentDurationMs,\n                    compressedSize: compression?.size,\n                }\n            )\n\n            // Use the stored streamUuid (which contains our custom filename) for the final filename\n            const filename = `${this.streamUuid}.${this.extension}`\n            const result: AudioRecording = {\n                fileUri,\n                filename, // This will now use our custom filename\n                bitDepth: this.bitDepth,\n                createdAt: this.recordingStartTime,\n                channels: this.recordingConfig?.channels ?? 1,\n                sampleRate: this.recordingConfig?.sampleRate ?? 44100,\n                durationMs: this.currentDurationMs,\n                size: this.currentSize,\n                mimeType,\n                compression,\n            }\n\n            // Reset after creating the result\n            this.streamUuid = null\n\n            return result\n        } catch (error) {\n            this.logger?.error('[Stop] Error stopping recording:', error)\n            throw error\n        }\n    }\n\n    // Pause recording\n    async pauseRecording() {\n        if (!this.isRecording || this.isPaused) {\n            throw new Error('Recording is not active or already paused')\n        }\n\n        if (this.customRecorder) {\n            this.customRecorder.pause()\n        }\n        this.isPaused = true\n        this.pausedTime = Date.now()\n    }\n\n    // Resume recording\n    async resumeRecording() {\n        if (!this.isPaused) {\n            throw new Error('Recording is not paused')\n        }\n\n        if (this.customRecorder) {\n            this.customRecorder.resume()\n        }\n        this.isPaused = false\n        this.recordingStartTime += Date.now() - this.pausedTime\n    }\n\n    // Get current status\n    status() {\n        const status: AudioStreamStatus = {\n            isRecording: this.isRecording,\n            isPaused: this.isPaused,\n            durationMs: this.currentDurationMs,\n            size: this.currentSize,\n            interval: this.currentInterval,\n            intervalAnalysis: this.currentIntervalAnalysis,\n            mimeType: `audio/${this.extension}`,\n            compression: this.recordingConfig?.compression?.enabled\n                ? {\n                      size: this.totalCompressedSize,\n                      mimeType: 'audio/webm',\n                      format: this.recordingConfig.compression.format ?? 'opus',\n                      bitrate:\n                          this.recordingConfig.compression.bitrate ?? 128000,\n                      compressedFileUri: `${this.streamUuid}.webm`,\n                  }\n                : undefined,\n        }\n        return status\n    }\n}\n"]}
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src/ExpoAudioStreamModule.web.ts\nimport { LegacyEventEmitter } from 'expo-modules-core'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport {\n    AudioRecording,\n    AudioStreamStatus,\n    BitDepth,\n    ConsoleLike,\n    RecordingConfig,\n    RecordingInterruptionReason,\n    StartRecordingResult,\n} from './ExpoAudioStream.types'\nimport { WebRecorder } from './WebRecorder.web'\nimport { AudioEventPayload } from './events'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\n\nexport interface AudioStreamEvent {\n    type: string\n    device?: string\n    timestamp: Date\n}\n\nexport interface ExpoAudioStreamOptions {\n    logger?: ConsoleLike\n    eventCallback?: (event: AudioStreamEvent) => void\n}\n\nexport interface EmitAudioEventProps {\n    data: Float32Array\n    position: number\n    compression?: {\n        data: Blob\n        size: number\n        totalSize: number\n        mimeType: string\n        format: string\n        bitrate: number\n    }\n}\nexport type EmitAudioEventFunction = (_: EmitAudioEventProps) => void\nexport type EmitAudioAnalysisFunction = (_: AudioAnalysis) => void\n\nexport interface ExpoAudioStreamWebProps {\n    logger?: ConsoleLike\n    audioWorkletUrl: string\n    featuresExtratorUrl: string\n    maxBufferSize?: number // Maximum number of chunks to keep in memory\n}\n\nexport class ExpoAudioStreamWeb extends LegacyEventEmitter {\n    customRecorder: WebRecorder | null\n    audioChunks: Float32Array[]\n    isRecording: boolean\n    isPaused: boolean\n    recordingStartTime: number\n    pausedTime: number\n    currentDurationMs: number\n    currentSize: number\n    currentInterval: number\n    currentIntervalAnalysis: number\n    lastEmittedSize: number\n    lastEmittedTime: number\n    lastEmittedCompressionSize: number\n    lastEmittedAnalysisTime: number\n    streamUuid: string | null\n    extension: 'webm' | 'wav' = 'wav' // Default extension is 'wav'\n    recordingConfig?: RecordingConfig\n    bitDepth: BitDepth // Bit depth of the audio\n    audioWorkletUrl: string\n    featuresExtratorUrl: string\n    logger?: ConsoleLike\n    latestPosition: number = 0\n    totalCompressedSize: number = 0\n    private readonly maxBufferSize: number\n    private eventCallback?: (event: AudioStreamEvent) => void\n\n    constructor({\n        audioWorkletUrl,\n        featuresExtratorUrl,\n        logger,\n        maxBufferSize = 100, // Default to storing last 100 chunks (1 chunk = 0.5 seconds)\n    }: ExpoAudioStreamWebProps) {\n        const mockNativeModule = {\n            addListener: () => {},\n            removeListeners: () => {},\n        }\n        super(mockNativeModule) // Pass the mock native module to the parent class\n\n        this.logger = logger\n        this.customRecorder = null\n        this.audioChunks = []\n        this.isRecording = false\n        this.isPaused = false\n        this.recordingStartTime = 0\n        this.pausedTime = 0\n        this.currentDurationMs = 0\n        this.currentSize = 0\n        this.bitDepth = 32 // Default\n        this.currentInterval = 1000 // Default interval in ms\n        this.currentIntervalAnalysis = 500 // Default analysis interval in ms\n        this.lastEmittedSize = 0\n        this.lastEmittedTime = 0\n        this.latestPosition = 0\n        this.lastEmittedCompressionSize = 0\n        this.lastEmittedAnalysisTime = 0\n        this.streamUuid = null // Initialize UUID on first recording start\n        this.audioWorkletUrl = audioWorkletUrl\n        this.featuresExtratorUrl = featuresExtratorUrl\n        this.maxBufferSize = maxBufferSize\n    }\n\n    // Utility to handle user media stream\n    async getMediaStream() {\n        try {\n            return await navigator.mediaDevices.getUserMedia({ audio: true })\n        } catch (error) {\n            this.logger?.error('Failed to get media stream:', error)\n            throw error\n        }\n    }\n\n    // Prepare recording with options\n    async prepareRecording(\n        recordingConfig: RecordingConfig = {}\n    ): Promise<boolean> {\n        if (this.isRecording) {\n            this.logger?.warn(\n                'Cannot prepare: Recording is already in progress'\n            )\n            return false\n        }\n\n        try {\n            // Check permissions and initialize basic settings\n            await this.getMediaStream().then((stream) => {\n                // Just verify we can access the microphone by getting a stream, then release it\n                stream.getTracks().forEach((track) => track.stop())\n            })\n\n            this.bitDepth = encodingToBitDepth({\n                encoding: recordingConfig.encoding ?? 'pcm_32bit',\n            })\n\n            // Store recording configuration for later use\n            this.recordingConfig = recordingConfig\n\n            // Use custom filename if provided, otherwise fallback to timestamp\n            if (recordingConfig.filename) {\n                // Remove any existing extension from the filename\n                this.streamUuid = recordingConfig.filename.replace(\n                    /\\.[^/.]+$/,\n                    ''\n                )\n            } else {\n                this.streamUuid = Date.now().toString()\n            }\n\n            this.logger?.debug('Recording preparation completed successfully')\n            return true\n        } catch (error) {\n            this.logger?.error('Error preparing recording:', error)\n            return false\n        }\n    }\n\n    // Start recording with options\n    async startRecording(\n        recordingConfig: RecordingConfig = {}\n    ): Promise<StartRecordingResult> {\n        if (this.isRecording) {\n            throw new Error('Recording is already in progress')\n        }\n\n        // If we haven't prepared or have different settings, prepare now\n        if (\n            !this.recordingConfig ||\n            this.recordingConfig.sampleRate !== recordingConfig.sampleRate ||\n            this.recordingConfig.channels !== recordingConfig.channels ||\n            this.recordingConfig.encoding !== recordingConfig.encoding\n        ) {\n            await this.prepareRecording(recordingConfig)\n        } else {\n            this.logger?.debug(\n                'Using previously prepared recording configuration'\n            )\n        }\n\n        // Save recording config for reference\n        this.recordingConfig = recordingConfig\n\n        const audioContext = new (window.AudioContext ||\n            // eslint-disable-next-line @typescript-eslint/ban-ts-comment\n            // @ts-ignore - Allow webkitAudioContext for Safari\n            window.webkitAudioContext)()\n        const stream = await this.getMediaStream()\n\n        const source = audioContext.createMediaStreamSource(stream)\n\n        this.customRecorder = new WebRecorder({\n            logger: this.logger,\n            audioContext,\n            source,\n            recordingConfig,\n            emitAudioEventCallback: this.customRecorderEventCallback.bind(this),\n            emitAudioAnalysisCallback:\n                this.customRecorderAnalysisCallback.bind(this),\n            onInterruption: this.handleRecordingInterruption.bind(this),\n        })\n        await this.customRecorder.init()\n        this.customRecorder.start()\n\n        this.isRecording = true\n        this.recordingStartTime = Date.now()\n        this.pausedTime = 0\n        this.isPaused = false\n        this.lastEmittedSize = 0\n        this.lastEmittedTime = 0\n        this.lastEmittedCompressionSize = 0\n        this.currentInterval = recordingConfig.interval ?? 1000\n        this.currentIntervalAnalysis = recordingConfig.intervalAnalysis ?? 500\n        this.lastEmittedAnalysisTime = Date.now()\n\n        // Use custom filename if provided, otherwise fallback to timestamp\n        if (recordingConfig.filename) {\n            // Remove any existing extension from the filename\n            this.streamUuid = recordingConfig.filename.replace(/\\.[^/.]+$/, '')\n        } else {\n            this.streamUuid = Date.now().toString()\n        }\n\n        const fileUri = `${this.streamUuid}.${this.extension}`\n        const streamConfig: StartRecordingResult = {\n            fileUri,\n            mimeType: `audio/${this.extension}`,\n            bitDepth: this.bitDepth,\n            channels: recordingConfig.channels ?? 1,\n            sampleRate: recordingConfig.sampleRate ?? 44100,\n            compression: recordingConfig.compression\n                ? {\n                      ...recordingConfig.compression,\n                      bitrate: recordingConfig.compression?.bitrate ?? 128000,\n                      size: 0,\n                      mimeType: 'audio/webm',\n                      format: recordingConfig.compression?.format ?? 'opus',\n                      compressedFileUri: '',\n                  }\n                : undefined,\n        }\n        return streamConfig\n    }\n\n    /**\n     * Centralized handler for recording interruptions\n     */\n    private handleRecordingInterruption(event: {\n        reason: RecordingInterruptionReason | string\n        isPaused: boolean\n        timestamp: number\n        message?: string\n    }): void {\n        this.logger?.debug(`Received recording interruption: ${event.reason}`)\n\n        // Update local state if the interruption should pause recording\n        if (event.isPaused) {\n            this.isPaused = true\n\n            // If this is a device disconnection, handle according to behavior setting\n            if (event.reason === 'deviceDisconnected') {\n                this.pausedTime = Date.now()\n\n                // Check if we should try fallback to another device\n                if (\n                    this.recordingConfig?.deviceDisconnectionBehavior ===\n                    'fallback'\n                ) {\n                    this.logger?.debug(\n                        'Device disconnected with fallback behavior - attempting to switch to default device'\n                    )\n\n                    // Try to restart with default device\n                    this.handleDeviceFallback().catch((error) => {\n                        // If fallback fails, emit warning\n                        this.logger?.error('Device fallback failed:', error)\n                        this.emit('onRecordingInterrupted', {\n                            reason: 'deviceSwitchFailed',\n                            isPaused: true,\n                            timestamp: Date.now(),\n                            message:\n                                'Failed to switch to fallback device. Recording paused.',\n                        })\n                    })\n                } else {\n                    // Just warn about disconnection if fallback not enabled\n                    this.logger?.warn(\n                        'Device disconnected - recording paused automatically'\n                    )\n                    this.emit('onRecordingInterrupted', event)\n                }\n            } else {\n                // For other interruption types, just emit the event\n                this.emit('onRecordingInterrupted', event)\n            }\n        } else {\n            // If not causing a pause, just forward the event\n            this.emit('onRecordingInterrupted', event)\n        }\n    }\n\n    /**\n     * Handler for audio events from the WebRecorder\n     */\n    private customRecorderEventCallback({\n        data,\n        position,\n        compression,\n    }: EmitAudioEventProps): void {\n        // Keep only the latest chunks based on maxBufferSize\n        this.audioChunks.push(new Float32Array(data))\n        if (this.audioChunks.length > this.maxBufferSize) {\n            this.audioChunks.shift() // Remove oldest chunk\n        }\n        this.currentSize += data.byteLength\n        this.emitAudioEvent({ data, position, compression })\n        this.lastEmittedTime = Date.now()\n        this.lastEmittedSize = this.currentSize\n        this.lastEmittedCompressionSize = compression?.size ?? 0\n    }\n\n    /**\n     * Handler for audio analysis events from the WebRecorder\n     */\n    private customRecorderAnalysisCallback(\n        audioAnalysisData: AudioAnalysis\n    ): void {\n        this.emit('AudioAnalysis', audioAnalysisData)\n    }\n\n    // Get recording duration\n    private getRecordingDuration(): number {\n        if (!this.isRecording) {\n            return 0\n        }\n\n        return this.currentDurationMs\n    }\n\n    emitAudioEvent({ data, position, compression }: EmitAudioEventProps) {\n        const fileUri = `${this.streamUuid}.${this.extension}`\n        if (compression?.size) {\n            this.lastEmittedCompressionSize = compression.size\n            this.totalCompressedSize = compression.totalSize\n        }\n\n        // Update latest position for tracking\n        this.latestPosition = position\n\n        // Calculate duration of this chunk in ms\n        const sampleRate = this.recordingConfig?.sampleRate || 44100\n        const chunkDurationMs = (data.length / sampleRate) * 1000\n\n        // Handle duration calculation\n        if (this.customRecorder?.isFirstChunkAfterSwitch) {\n            this.logger?.debug(\n                `Processing first chunk after device switch, duration preserved at ${this.currentDurationMs}ms`\n            )\n            this.customRecorder.isFirstChunkAfterSwitch = false\n        } else {\n            this.currentDurationMs += chunkDurationMs\n        }\n\n        const audioEventPayload: AudioEventPayload = {\n            fileUri,\n            mimeType: `audio/${this.extension}`,\n            lastEmittedSize: this.lastEmittedSize,\n            deltaSize: data.byteLength,\n            position,\n            totalSize: this.currentSize,\n            buffer: data,\n            streamUuid: this.streamUuid ?? '',\n            compression: compression\n                ? {\n                      data: compression?.data,\n                      totalSize: this.totalCompressedSize,\n                      eventDataSize: compression?.size ?? 0,\n                      position,\n                  }\n                : undefined,\n        }\n\n        this.emit('AudioData', audioEventPayload)\n    }\n\n    // Stop recording\n    async stopRecording(): Promise<AudioRecording> {\n        if (!this.customRecorder) {\n            throw new Error('Recorder is not initialized')\n        }\n\n        this.logger?.debug('Starting stop process')\n\n        try {\n            const { compressedBlob } = await this.customRecorder.stop()\n\n            this.isRecording = false\n            this.isPaused = false\n\n            let compression: AudioRecording['compression']\n            let fileUri = `${this.streamUuid}.${this.extension}`\n            let mimeType = `audio/${this.extension}`\n\n            // Process compressed audio if available\n            if (compressedBlob && this.recordingConfig?.compression?.enabled) {\n                const compressedUri = URL.createObjectURL(compressedBlob)\n                compression = {\n                    compressedFileUri: compressedUri,\n                    size: compressedBlob.size,\n                    mimeType: 'audio/webm',\n                    format: 'opus',\n                    bitrate: this.recordingConfig.compression.bitrate ?? 128000,\n                }\n\n                // Use compressed values when compression is enabled\n                fileUri = compressedUri\n                mimeType = 'audio/webm'\n            }\n\n            // Use the stored streamUuid for the final filename\n            const filename = `${this.streamUuid}.${this.extension}`\n            const result: AudioRecording = {\n                fileUri,\n                filename,\n                bitDepth: this.bitDepth,\n                createdAt: this.recordingStartTime,\n                channels: this.recordingConfig?.channels ?? 1,\n                sampleRate: this.recordingConfig?.sampleRate ?? 44100,\n                durationMs: this.currentDurationMs,\n                size: this.currentSize,\n                mimeType,\n                compression,\n            }\n\n            // Reset after creating the result\n            this.streamUuid = null\n\n            return result\n        } catch (error) {\n            this.logger?.error('Error stopping recording:', error)\n            throw error\n        }\n    }\n\n    // Pause recording\n    async pauseRecording() {\n        if (!this.isRecording) {\n            throw new Error('Recording is not active')\n        }\n\n        if (this.isPaused) {\n            this.logger?.debug('Recording already paused, skipping')\n            return\n        }\n\n        try {\n            if (this.customRecorder) {\n                this.customRecorder.pause()\n            }\n            this.isPaused = true\n            this.pausedTime = Date.now()\n        } catch (error) {\n            this.logger?.error('Error in pauseRecording', error)\n            // Even if the pause operation failed, make sure our state is consistent\n            this.isPaused = true\n            this.pausedTime = Date.now()\n        }\n    }\n\n    // Resume recording\n    async resumeRecording() {\n        if (!this.isPaused) {\n            throw new Error('Recording is not paused')\n        }\n\n        this.logger?.debug('Resuming recording', {\n            deviceDisconnectionBehavior:\n                this.recordingConfig?.deviceDisconnectionBehavior,\n            isDeviceDisconnected: this.customRecorder?.isDeviceDisconnected,\n        })\n\n        try {\n            // If we have no recorder, or if the device is disconnected, always attempt fallback\n            if (\n                !this.customRecorder ||\n                this.customRecorder.isDeviceDisconnected\n            ) {\n                this.logger?.debug(\n                    'No recorder exists or device disconnected - attempting fallback on resume'\n                )\n                await this.handleDeviceFallback()\n                // handleDeviceFallback will manage resuming if successful, or emit error if failed.\n                return\n            }\n\n            // Normal resume path - device is still connected\n            this.customRecorder.resume()\n            this.isPaused = false\n\n            // Adjust the recording start time to account for the pause duration\n            const pauseDuration = Date.now() - this.pausedTime\n            this.recordingStartTime += pauseDuration\n            this.pausedTime = 0\n\n            this.emit('onRecordingInterrupted', {\n                reason: 'userResumed',\n                isPaused: false,\n                timestamp: Date.now(),\n            })\n        } catch (error) {\n            this.logger?.error('Resume failed:', error)\n            // Fallback to emitting a general failure if resume fails unexpectedly\n            this.emit('onRecordingInterrupted', {\n                reason: 'resumeFailed', // Use a more specific reason\n                isPaused: true, // Remain paused if resume fails\n                timestamp: Date.now(),\n                message:\n                    'Failed to resume recording. Please stop and start again.',\n            })\n        }\n    }\n\n    // Get current status\n    status() {\n        const durationMs = this.getRecordingDuration()\n\n        const status: AudioStreamStatus = {\n            isRecording: this.isRecording,\n            isPaused: this.isPaused,\n            durationMs,\n            size: this.currentSize,\n            interval: this.currentInterval,\n            intervalAnalysis: this.currentIntervalAnalysis,\n            mimeType: `audio/${this.extension}`,\n            compression: this.recordingConfig?.compression?.enabled\n                ? {\n                      size: this.totalCompressedSize,\n                      mimeType: 'audio/webm',\n                      format: this.recordingConfig.compression.format ?? 'opus',\n                      bitrate:\n                          this.recordingConfig.compression.bitrate ?? 128000,\n                      compressedFileUri: `${this.streamUuid}.webm`,\n                  }\n                : undefined,\n        }\n        return status\n    }\n\n    /**\n     * Handles device fallback when the current device is disconnected\n     */\n    private async handleDeviceFallback(): Promise<boolean> {\n        this.logger?.debug('Starting device fallback procedure')\n\n        if (!this.isRecording) {\n            return false\n        }\n\n        try {\n            // Save important state before switching\n            const currentPosition = this.latestPosition\n            const existingAudioChunks = [...this.audioChunks]\n\n            // Save compressed chunks if available\n            let compressedChunks: Blob[] = []\n            if (this.customRecorder) {\n                try {\n                    compressedChunks = this.customRecorder.getCompressedChunks()\n                } catch (err) {\n                    this.logger?.warn('Failed to get compressed chunks:', err)\n                }\n            }\n\n            // Save the current counter value for continuity\n            let currentDataPointCounter = 0\n            if (this.customRecorder) {\n                currentDataPointCounter =\n                    this.customRecorder.getDataPointCounter()\n            }\n\n            // Clean up existing recorder\n            if (this.customRecorder) {\n                try {\n                    this.customRecorder.cleanup()\n                } catch (cleanupError) {\n                    this.logger?.warn('Error during cleanup:', cleanupError)\n                }\n            }\n\n            // Keep recording state true but mark as paused\n            this.isPaused = true\n            this.pausedTime = Date.now()\n\n            // Store current size and other stats\n            const previousTotalSize = this.currentSize\n            const previousLastEmittedSize = this.lastEmittedSize\n            const previousCompressedSize = this.totalCompressedSize\n\n            // Try to get a fallback device\n            const fallbackDeviceInfo = await this.getFallbackDevice()\n            if (!fallbackDeviceInfo) {\n                this.emit('onRecordingInterrupted', {\n                    reason: 'deviceSwitchFailed',\n                    isPaused: true,\n                    timestamp: Date.now(),\n                    message:\n                        'Failed to switch to fallback device. Recording paused.',\n                })\n                return false\n            }\n\n            // Start recording with the new device\n            try {\n                const stream = await this.requestPermissionsAndGetUserMedia(\n                    fallbackDeviceInfo.deviceId\n                )\n                const audioContext = new (window.AudioContext ||\n                    // eslint-disable-next-line @typescript-eslint/ban-ts-comment\n                    // @ts-ignore - Allow webkitAudioContext for Safari\n                    window.webkitAudioContext)()\n\n                const source = audioContext.createMediaStreamSource(stream)\n\n                // Create a new recorder with the fallback device\n                this.customRecorder = new WebRecorder({\n                    logger: this.logger,\n                    audioContext,\n                    source,\n                    recordingConfig: this.recordingConfig || {},\n                    emitAudioEventCallback:\n                        this.customRecorderEventCallback.bind(this),\n                    emitAudioAnalysisCallback:\n                        this.customRecorderAnalysisCallback.bind(this),\n                    onInterruption: this.handleRecordingInterruption.bind(this),\n                })\n\n                await this.customRecorder.init()\n\n                // Set the initial position to continue from the previous device\n                this.customRecorder.setPosition(currentPosition)\n\n                // Reset the data point counter to continue from where the previous device left off\n                if (currentDataPointCounter > 0) {\n                    this.customRecorder.resetDataPointCounter(\n                        currentDataPointCounter\n                    )\n                }\n\n                // Prepare the recorder to handle the device switch properly\n                this.customRecorder.prepareForDeviceSwitch()\n\n                // Restore the existing audio chunks\n                if (existingAudioChunks.length > 0) {\n                    this.audioChunks = existingAudioChunks\n                }\n\n                // Restore compressed chunks if available\n                if (compressedChunks.length > 0) {\n                    this.customRecorder.setCompressedChunks(compressedChunks)\n                }\n\n                // Start the new recorder while preserving counters\n                this.customRecorder.start(true)\n\n                // Update recording state\n                this.isPaused = false\n                this.recordingStartTime = Date.now()\n\n                // Restore size counters to maintain continuity\n                this.currentSize = previousTotalSize\n                this.lastEmittedSize = previousLastEmittedSize\n                this.totalCompressedSize = previousCompressedSize\n\n                // Notify that we switched to a fallback device\n                if (this.eventCallback) {\n                    this.eventCallback({\n                        type: 'deviceFallback',\n                        device: fallbackDeviceInfo.deviceId,\n                        timestamp: new Date(),\n                    })\n                }\n                return true\n            } catch (error) {\n                this.logger?.error(\n                    'Failed to start recording with fallback device',\n                    error\n                )\n                this.isPaused = true\n                this.emit('onRecordingInterrupted', {\n                    reason: 'deviceSwitchFailed',\n                    isPaused: true,\n                    timestamp: Date.now(),\n                    message:\n                        'Failed to switch to fallback device. Recording paused.',\n                })\n                return false\n            }\n        } catch (error) {\n            this.logger?.error('Failed to use fallback device', error)\n            this.isPaused = true\n            this.emit('onRecordingInterrupted', {\n                reason: 'deviceSwitchFailed',\n                isPaused: true,\n                timestamp: Date.now(),\n                message:\n                    'Failed to switch to fallback device. Recording paused.',\n            })\n            return false\n        }\n    }\n\n    /**\n     * Attempts to get a fallback audio device\n     */\n    private async getFallbackDevice(): Promise<MediaDeviceInfo | null> {\n        try {\n            // Get list of available audio input devices\n            const devices = await navigator.mediaDevices.enumerateDevices()\n            const audioInputDevices = devices.filter(\n                (device) => device.kind === 'audioinput'\n            )\n\n            if (audioInputDevices.length === 0) {\n                return null\n            }\n\n            // Try to find a device that's not the current one\n            if (this.customRecorder) {\n                try {\n                    // Use mediaDevices.enumerateDevices to find the current active device\n                    const tracks = navigator.mediaDevices\n                        .getUserMedia({ audio: true })\n                        .then((stream) => {\n                            const track = stream.getAudioTracks()[0]\n                            return track ? track.label : ''\n                        })\n                        .catch(() => '')\n\n                    const currentTrackLabel = await tracks\n\n                    if (currentTrackLabel) {\n                        // Find a device with a different label\n                        const differentDevice = audioInputDevices.find(\n                            (device) =>\n                                device.label &&\n                                device.label !== currentTrackLabel\n                        )\n\n                        if (differentDevice) {\n                            return differentDevice\n                        }\n                    }\n                } catch (err) {\n                    this.logger?.warn(\n                        'Error determining current device, using default'\n                    )\n                }\n            }\n\n            // Return the first available device (default device)\n            return audioInputDevices[0]\n        } catch (error) {\n            this.logger?.error('Error finding fallback device:', error)\n            return null\n        }\n    }\n\n    /**\n     * Gets user media with specific device ID\n     */\n    private async requestPermissionsAndGetUserMedia(\n        deviceId: string\n    ): Promise<MediaStream> {\n        try {\n            // Request the specific device\n            return await navigator.mediaDevices.getUserMedia({\n                audio: {\n                    deviceId: { exact: deviceId },\n                },\n            })\n        } catch (error) {\n            this.logger?.error(\n                `Failed to get media for device ${deviceId}`,\n                error\n            )\n            // Try with default constraints as fallback\n            return await navigator.mediaDevices.getUserMedia({ audio: true })\n        }\n    }\n\n    init(options?: ExpoAudioStreamOptions): Promise<void> {\n        try {\n            this.logger = options?.logger\n            this.eventCallback = options?.eventCallback\n            return Promise.resolve()\n        } catch (error) {\n            this.logger?.error('Error initializing ExpoAudioStream', error)\n            return Promise.reject(error)\n        }\n    }\n}\n"]}

package/build/ExpoAudioStreamModule.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"ExpoAudioStreamModule.d.ts","sourceRoot":"","sources":["../src/ExpoAudioStreamModule.ts"],"names":[],"mappings":"AAmBA,QAAA,IAAI,qBAAqB,EAAE,GAAG,CAAA;AAm7B9B,eAAe,qBAAqB,CAAA"}
+{"version":3,"file":"ExpoAudioStreamModule.d.ts","sourceRoot":"","sources":["../src/ExpoAudioStreamModule.ts"],"names":[],"mappings":"AAmBA,QAAA,IAAI,qBAAqB,EAAE,GAAG,CAAA;AA08B9B,eAAe,qBAAqB,CAAA"}

package/build/ExpoAudioStreamModule.js

@@ -573,6 +573,26 @@ if (Platform.OS === 'web') {
             delete ExpoAudioStreamModule.listeners[eventName];
         }
     };
+    ExpoAudioStreamModule.prepareRecording = async (options) => {
+        // For web platform, we'll implement a simplified version that just checks permissions
+        // and does minimal setup. The actual recording setup will still happen in startRecording.
+        try {
+            // Check for microphone permissions
+            const permissionsResult = await ExpoAudioStreamModule.getPermissionsAsync();
+            if (!permissionsResult.granted) {
+                throw new Error('Microphone permission not granted');
+            }
+            // If using a web instance, call its prepareRecording method
+            if (instance) {
+                return await instance.prepareRecording(options);
+            }
+            return true;
+        }
+        catch (error) {
+            console.error('Error preparing recording:', error);
+            throw error;
+        }
+    };
 }
 // Move the encodeCompressedAudio function outside the if block to fix the ESLint error
 async function encodeCompressedAudio(buffer, format, bitrate) {

package/build/ExpoAudioStreamModule.js.map

@@ -1 +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"]}
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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    ExpoAudioStreamModule.prepareRecording = async (options: any) => {\n        // For web platform, we'll implement a simplified version that just checks permissions\n        // and does minimal setup. The actual recording setup will still happen in startRecording.\n        try {\n            // Check for microphone permissions\n            const permissionsResult =\n                await ExpoAudioStreamModule.getPermissionsAsync()\n            if (!permissionsResult.granted) {\n                throw new Error('Microphone permission not granted')\n            }\n\n            // If using a web instance, call its prepareRecording method\n            if (instance) {\n                return await instance.prepareRecording(options)\n            }\n\n            return true\n        } catch (error) {\n            console.error('Error preparing recording:', error)\n            throw error\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"]}