@siteed/expo-audio-studio 2.7.0 → 2.8.1

package/build/AudioDeviceManager.js.map

@@ -1 +0,0 @@
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{ EventEmitter } from 'expo-modules-core'\nimport { Platform } from 'react-native'\n\nimport {\n    AudioDevice,\n    AudioDeviceCapabilities,\n    DeviceDisconnectionBehavior,\n    ConsoleLike,\n} from './ExpoAudioStream.types'\nimport ExpoAudioStreamModule from './ExpoAudioStreamModule'\n\n// Default device fallback for web and unsupported platforms\nconst DEFAULT_DEVICE: AudioDevice = {\n    id: 'default',\n    name: 'Default Microphone',\n    type: 'builtin_mic',\n    isDefault: true,\n    isAvailable: true,\n    capabilities: {\n        sampleRates: [16000, 44100, 48000],\n        channelCounts: [1, 2],\n        bitDepths: [16, 24, 32],\n        hasEchoCancellation: true,\n        hasNoiseSuppression: true,\n        hasAutomaticGainControl: true,\n    },\n}\n\n// Helper function to map raw object to AudioDevice interface\n// This handles potential inconsistencies from the native module\nfunction mapRawDeviceToAudioDevice(rawDevice: any): AudioDevice {\n    const capabilities = rawDevice.capabilities || {}\n    return {\n        id: rawDevice.id || 'unknown',\n        name: rawDevice.name || 'Unknown Device',\n        type: rawDevice.type || 'unknown',\n        isDefault: rawDevice.isDefault || false,\n        isAvailable:\n            rawDevice.isAvailable !== undefined ? rawDevice.isAvailable : true, // Default to true if undefined\n        capabilities: {\n            sampleRates: capabilities.sampleRates || [16000, 44100, 48000], // Provide defaults\n            channelCounts: capabilities.channelCounts || [1, 2],\n            bitDepths: capabilities.bitDepths || [16, 24, 32],\n            hasEchoCancellation: capabilities.hasEchoCancellation,\n            hasNoiseSuppression: capabilities.hasNoiseSuppression,\n            hasAutomaticGainControl: capabilities.hasAutomaticGainControl,\n        },\n    }\n}\n\n/**\n * Class that provides a cross-platform API for managing audio input devices\n */\nexport class AudioDeviceManager {\n    private eventEmitter: InstanceType<typeof EventEmitter>\n    private currentDeviceId: string | null = null\n    private availableDevices: AudioDevice[] = []\n    private deviceChangeListeners: Set<(devices: AudioDevice[]) => void> =\n        new Set()\n    private deviceListeners: Set<() => void> = new Set()\n    private lastRefreshTime: number = 0\n    private refreshInProgress: boolean = false\n    private refreshDebounceMs: number = 500 // Minimum 500ms between refreshes\n    private logger?: ConsoleLike\n\n    constructor(options?: { logger?: ConsoleLike }) {\n        this.eventEmitter = new EventEmitter(ExpoAudioStreamModule)\n        this.logger = options?.logger\n\n        // Listen for device change events from native modules if not on web\n        if (Platform.OS !== 'web') {\n            // Store the last event type to avoid duplicates\n            let lastEventType: string | null = null\n            let lastEventTime = 0\n\n            this.eventEmitter.addListener(\n                'deviceChangedEvent',\n                (event: any) => {\n                    // Skip processing duplicate events that occur too close together\n                    const now = Date.now()\n                    const isSimilarEvent =\n                        lastEventType === event.type &&\n                        now - lastEventTime < this.refreshDebounceMs\n\n                    if (isSimilarEvent) {\n                        this.logger?.debug(\n                            `Skipping similar device event (${event.type}) received too soon`\n                        )\n                        return\n                    }\n\n                    // Update the last event tracking\n                    lastEventType = event.type\n                    lastEventTime = now\n\n                    // Only refresh on meaningful events\n                    if (\n                        event.type === 'deviceConnected' ||\n                        event.type === 'deviceDisconnected' ||\n                        event.type === 'routeChanged'\n                    ) {\n                        this.logger?.debug(\n                            `Processing device event: ${event.type}`\n                        )\n                        // Refresh devices and notify listeners regardless of the direct return value\n                        this.refreshDevices()\n                    }\n                }\n            )\n        }\n    }\n\n    /**\n     * Initialize the device manager with a logger\n     * @param logger A logger instance that implements the ConsoleLike interface\n     * @returns The manager instance for chaining\n     */\n    initWithLogger(logger: ConsoleLike): AudioDeviceManager {\n        this.setLogger(logger)\n        return this\n    }\n\n    /**\n     * Set the logger instance\n     * @param logger A logger instance that implements the ConsoleLike interface\n     */\n    setLogger(logger: ConsoleLike) {\n        this.logger = logger\n    }\n\n    /**\n     * Get all available audio input devices\n     * @param options Optional settings to force refresh the device list. Can include a refresh flag.\n     * @returns Promise resolving to an array of audio devices conforming to AudioDevice interface\n     */\n    async getAvailableDevices(options?: {\n        refresh?: boolean\n    }): Promise<AudioDevice[]> {\n        try {\n            if (Platform.OS === 'web') {\n                this.availableDevices = await this.getWebAudioDevices()\n            } else if (ExpoAudioStreamModule.getAvailableInputDevices) {\n                // Expecting an array of raw device objects from native\n                const rawDevices: any[] =\n                    await ExpoAudioStreamModule.getAvailableInputDevices(\n                        options\n                    )\n                // Map raw objects to the AudioDevice interface\n                this.availableDevices = rawDevices.map(\n                    mapRawDeviceToAudioDevice\n                )\n            } else {\n                // Fallback for unsupported platforms\n                this.availableDevices = [DEFAULT_DEVICE]\n            }\n            return this.availableDevices\n        } catch (error) {\n            this.logger?.error('Failed to get available devices:', error)\n            this.availableDevices = [DEFAULT_DEVICE] // Ensure state is updated on error\n            return this.availableDevices\n        }\n    }\n\n    /**\n     * Get the currently selected audio input device\n     * @returns Promise resolving to the current device (conforming to AudioDevice) or null\n     */\n    async getCurrentDevice(): Promise<AudioDevice | null> {\n        try {\n            if (Platform.OS === 'web') {\n                if (!this.currentDeviceId) {\n                    // On web, return the typed default device if nothing is selected\n                    return DEFAULT_DEVICE\n                }\n                // Refresh web devices to ensure the current one is up-to-date\n                const webDevices = await this.getWebAudioDevices()\n                return (\n                    webDevices.find((d) => d.id === this.currentDeviceId) ||\n                    DEFAULT_DEVICE // Fallback to default if current ID not found\n                )\n            } else if (ExpoAudioStreamModule.getCurrentInputDevice) {\n                // Expecting a single raw device object or null from native\n                const rawDevice: any | null =\n                    await ExpoAudioStreamModule.getCurrentInputDevice()\n                // Map to AudioDevice interface if not null\n                return rawDevice ? mapRawDeviceToAudioDevice(rawDevice) : null\n            } else {\n                // Fallback for unsupported platforms\n                return DEFAULT_DEVICE\n            }\n        } catch (error) {\n            this.logger?.error('Failed to get current device:', error)\n            return DEFAULT_DEVICE // Return default on error\n        }\n    }\n\n    /**\n     * Select a specific audio input device for recording\n     * @param deviceId The ID of the device to select\n     * @returns Promise resolving to a boolean indicating success\n     */\n    async selectDevice(deviceId: string): Promise<boolean> {\n        try {\n            let success = false\n            if (Platform.OS === 'web') {\n                // Check if the device exists before setting it\n                const devices = await this.getWebAudioDevices()\n                if (devices.some((d) => d.id === deviceId)) {\n                    this.currentDeviceId = deviceId\n                    success = true\n                } else {\n                    this.logger?.warn(\n                        `Web: Device with ID ${deviceId} not found.`\n                    )\n                    success = false\n                }\n            } else if (ExpoAudioStreamModule.selectInputDevice) {\n                success =\n                    await ExpoAudioStreamModule.selectInputDevice(deviceId)\n                if (success) {\n                    this.currentDeviceId = deviceId\n                }\n            }\n            // Refresh devices after selection attempt to update state\n            await this.refreshDevices()\n            return success\n        } catch (error) {\n            this.logger?.error('Failed to select device:', error)\n            await this.refreshDevices() // Refresh even on error\n            return false\n        }\n    }\n\n    /**\n     * Reset to the default audio input device\n     * @returns Promise resolving to a boolean indicating success\n     */\n    async resetToDefaultDevice(): Promise<boolean> {\n        try {\n            let success = false\n            if (Platform.OS === 'web') {\n                this.currentDeviceId = 'default'\n                success = true\n            } else if (ExpoAudioStreamModule.resetToDefaultDevice) {\n                success = await ExpoAudioStreamModule.resetToDefaultDevice()\n                if (success) {\n                    this.currentDeviceId = null\n                }\n            }\n            // Refresh devices after reset attempt\n            await this.refreshDevices()\n            return success\n        } catch (error) {\n            this.logger?.error('Failed to reset to default device:', error)\n            await this.refreshDevices() // Refresh even on error\n            return false\n        }\n    }\n\n    /**\n     * Register a listener for device changes\n     * @param listener Function to call when devices change (receives AudioDevice[])\n     * @returns Function to remove the listener\n     */\n    addDeviceChangeListener(\n        listener: (devices: AudioDevice[]) => void\n    ): () => void {\n        this.deviceChangeListeners.add(listener)\n\n        // Immediately call listener with current devices if available\n        if (this.availableDevices.length > 0) {\n            listener([...this.availableDevices])\n        }\n\n        // Return a function to remove the listener\n        return () => {\n            this.deviceChangeListeners.delete(listener)\n        }\n    }\n\n    /**\n     * Refresh the list of available devices with debouncing and notify listeners.\n     * @returns Promise resolving to the updated device list (AudioDevice[])\n     */\n    async refreshDevices(): Promise<AudioDevice[]> {\n        const now = Date.now()\n\n        if (this.refreshInProgress) {\n            this.logger?.debug('Refresh already in progress, skipping')\n            return this.availableDevices\n        }\n\n        // Always allow refresh if forced by native event or longer than 2s debounce\n        const timeSinceLastRefresh = now - this.lastRefreshTime\n        const shouldDebounce =\n            timeSinceLastRefresh < this.refreshDebounceMs &&\n            timeSinceLastRefresh < 2000\n\n        if (shouldDebounce) {\n            this.logger?.debug(\n                `Refresh debounced, skipping (last refresh was ${timeSinceLastRefresh}ms ago)`\n            )\n            return this.availableDevices\n        }\n\n        this.logger?.debug('Refreshing devices...')\n        this.refreshInProgress = true\n        try {\n            // Fetch the latest devices; getAvailableDevices handles mapping now\n            const devices = await this.getAvailableDevices({ refresh: true })\n            // availableDevices state is updated within getAvailableDevices\n            this.notifyListeners() // Notify listeners with the updated list\n            this.lastRefreshTime = Date.now()\n            return devices // Return the fetched & mapped list\n        } catch (error) {\n            this.logger?.error('Error during refreshDevices:', error)\n            return this.availableDevices // Return potentially stale list on error\n        } finally {\n            this.refreshInProgress = false\n            this.logger?.debug('Refresh finished.')\n        }\n    }\n\n    /**\n     * Get audio input devices using the Web Audio API\n     * @returns Promise resolving to an array of AudioDevice objects\n     */\n    private async getWebAudioDevices(): Promise<AudioDevice[]> {\n        if (\n            typeof navigator === 'undefined' ||\n            !navigator.mediaDevices ||\n            !navigator.mediaDevices.enumerateDevices\n        ) {\n            return [DEFAULT_DEVICE]\n        }\n\n        try {\n            const permissionStatus = await this.checkMicrophonePermission()\n\n            if (permissionStatus === 'denied') {\n                return [\n                    {\n                        ...DEFAULT_DEVICE,\n                        name: 'Microphone Access Denied',\n                        isAvailable: false,\n                    },\n                ]\n            }\n\n            if (permissionStatus !== 'granted') {\n                try {\n                    // Requesting stream often reveals device labels\n                    await navigator.mediaDevices.getUserMedia({ audio: true })\n                } catch (error) {\n                    this.logger?.warn(\n                        'Microphone permission request failed:',\n                        error\n                    )\n                    return [\n                        {\n                            ...DEFAULT_DEVICE,\n                            name: 'Microphone Access Required',\n                            isAvailable: false,\n                        },\n                    ]\n                }\n            }\n\n            const devices = await navigator.mediaDevices.enumerateDevices()\n            const audioInputDevices = devices\n                .filter((device) => device.kind === 'audioinput')\n                .map((device) => this.mapWebDeviceToAudioDevice(device))\n\n            const hasUnlabeledDevices = audioInputDevices.some(\n                (device) =>\n                    !device.name || device.name.startsWith('Microphone ')\n            )\n\n            let finalDevices = audioInputDevices\n            if (hasUnlabeledDevices && this.isSafariOrIOS()) {\n                finalDevices = this.enhanceDevicesForSafari(audioInputDevices)\n            }\n\n            if (finalDevices.length === 0) {\n                finalDevices = [DEFAULT_DEVICE]\n            }\n\n            this.setupWebDeviceChangeListener()\n            this.availableDevices = finalDevices // Update internal state\n            return finalDevices\n        } catch (error) {\n            this.logger?.error('Failed to enumerate web audio devices:', error)\n            this.availableDevices = [DEFAULT_DEVICE] // Update state on error\n            return this.availableDevices\n        }\n    }\n\n    /**\n     * Check the current microphone permission status\n     * @returns Permission state ('prompt', 'granted', or 'denied')\n     */\n    private async checkMicrophonePermission(): Promise<PermissionState> {\n        if (!navigator.permissions || !navigator.permissions.query) {\n            return 'prompt'\n        }\n        try {\n            const permissionStatus = await navigator.permissions.query({\n                name: 'microphone' as PermissionName,\n            })\n            permissionStatus.onchange = () => {\n                // Refresh devices when permission changes\n                this.refreshDevices()\n            }\n            return permissionStatus.state\n        } catch (error) {\n            this.logger?.warn('Permission query not supported:', error)\n            return 'prompt'\n        }\n    }\n\n    /**\n     * Setup listener for device changes in web environment\n     */\n    private setupWebDeviceChangeListener() {\n        if (\n            typeof navigator === 'undefined' ||\n            !navigator.mediaDevices ||\n            this.deviceListeners.size > 0 // Avoid adding multiple listeners\n        ) {\n            return\n        }\n\n        const handleDeviceChange = () => {\n            this.logger?.debug('Web device change detected.')\n            // Refresh devices on change\n            this.refreshDevices()\n        }\n\n        navigator.mediaDevices.addEventListener(\n            'devicechange',\n            handleDeviceChange\n        )\n        this.deviceListeners.add(handleDeviceChange)\n        this.logger?.debug('Web device change listener added.')\n    }\n\n    /**\n     * Check if the current browser is Safari or iOS WebKit\n     */\n    private isSafariOrIOS(): boolean {\n        if (typeof navigator === 'undefined') return false\n        const ua = navigator.userAgent\n        return (\n            /^((?!chrome|android).)*safari/i.test(ua) ||\n            /iPad|iPhone|iPod/.test(ua) ||\n            (navigator.platform === 'MacIntel' && navigator.maxTouchPoints > 1)\n        )\n    }\n\n    /**\n     * Create enhanced device information for Safari and privacy-restricted browsers\n     * @param devices Array of AudioDevice objects, potentially unlabeled\n     * @returns Array of enhanced AudioDevice objects\n     */\n    private enhanceDevicesForSafari(devices: AudioDevice[]): AudioDevice[] {\n        const defaultDevice = devices.find((d) => d.isDefault)\n\n        if (devices.length <= 1) {\n            // Return a typed default device\n            return [\n                {\n                    id: defaultDevice?.id || 'default',\n                    name: 'Microphone (Browser Managed)',\n                    type: 'builtin_mic',\n                    isDefault: true,\n                    isAvailable: true,\n                    capabilities:\n                        defaultDevice?.capabilities ||\n                        DEFAULT_DEVICE.capabilities,\n                },\n            ]\n        }\n\n        // Provide more descriptive names for unlabeled devices\n        return devices.map((device, index) => {\n            if (!device.name || device.name.startsWith('Microphone ')) {\n                const deviceTypes = [\n                    'Built-in Microphone',\n                    'External Microphone',\n                    'Headset Microphone',\n                ]\n                const typeName = deviceTypes[index % deviceTypes.length]\n                return {\n                    ...device,\n                    name: device.isDefault ? `${typeName} (Default)` : typeName,\n                }\n            }\n            return device\n        })\n    }\n\n    /**\n     * Map a Web MediaDeviceInfo to our AudioDevice format\n     * @param device The MediaDeviceInfo object from the browser\n     * @returns An object conforming to the AudioDevice interface\n     */\n    private mapWebDeviceToAudioDevice(device: MediaDeviceInfo): AudioDevice {\n        const isDefault = device.deviceId === 'default'\n        const deviceType = this.inferDeviceType(device.label || '')\n\n        // Provide reasonable default capabilities for web devices\n        const defaultWebCapabilities: AudioDeviceCapabilities = {\n            sampleRates: [16000, 44100, 48000],\n            channelCounts: [1, 2],\n            bitDepths: [16, 32], // Web Audio uses float32, common PCM might be 16/32\n            hasEchoCancellation: true, // Often handled by browser\n            hasNoiseSuppression: true, // Often handled by browser\n            hasAutomaticGainControl: true, // Often handled by browser\n        }\n\n        return {\n            id: device.deviceId,\n            name:\n                device.label || `Microphone ${device.deviceId.substring(0, 8)}`,\n            type: deviceType,\n            isDefault,\n            isAvailable: true, // Assume available if enumerated\n            capabilities: defaultWebCapabilities,\n        }\n    }\n\n    /**\n     * Try to infer the device type from its name\n     * @param deviceName The label of the device\n     * @returns A string representing the inferred device type\n     */\n    private inferDeviceType(deviceName: string): string {\n        const name = deviceName.toLowerCase()\n        if (name.includes('bluetooth') || name.includes('airpods'))\n            return 'bluetooth'\n        if (name.includes('usb')) return 'usb'\n        if (name.includes('headphone') || name.includes('headset')) {\n            return name.includes('wired') ? 'wired_headset' : 'wired_headphones'\n        }\n        if (name.includes('speaker')) return 'speaker'\n        return 'builtin_mic' // Default assumption\n    }\n\n    /**\n     * Notify all registered listeners about device changes.\n     */\n    private notifyListeners(): void {\n        // Pass a copy of the current devices array to listeners\n        const devicesCopy = [...this.availableDevices]\n        this.logger?.debug(\n            `Notifying ${this.deviceChangeListeners.size} listeners with ${devicesCopy.length} devices.`\n        )\n        this.deviceChangeListeners.forEach((listener) => {\n            try {\n                listener(devicesCopy)\n            } catch (error) {\n                this.logger?.error('Error in device change listener:', error)\n            }\n        })\n    }\n}\n\n// Create and export the singleton instance\nexport const audioDeviceManager = new AudioDeviceManager()\n\nexport { DeviceDisconnectionBehavior }\n"]}

package/build/AudioRecorder.provider.d.ts.map

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package/build/ExpoAudioStream.native.d.ts.map

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package/build/ExpoAudioStream.native.js.map

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package/build/ExpoAudioStream.types.d.ts.map

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package/build/ExpoAudioStream.types.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"ExpoAudioStream.types.js","sourceRoot":"","sources":["../src/ExpoAudioStream.types.ts"],"names":[],"mappings":"AAyRA,4EAA4E;AAC5E,MAAM,CAAC,MAAM,2BAA2B,GAAG;IACvC,8CAA8C;IAC9C,KAAK,EAAE,OAAO;IACd,sDAAsD;IACtD,QAAQ,EAAE,UAAU;CACd,CAAA","sourcesContent":["// packages/expo-audio-stream/src/ExpoAudioStream.types.ts\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DecodingConfig,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport { AudioAnalysisEvent } from './events'\n\nexport interface CompressionInfo {\n    /** Size of the compressed audio data in bytes */\n    size: number\n    /** MIME type of the compressed audio (e.g., 'audio/aac', 'audio/opus') */\n    mimeType: string\n    /** Bitrate of the compressed audio in bits per second */\n    bitrate: number\n    /** Format of the compression (e.g., 'aac', 'opus') */\n    format: string\n    /** URI to the compressed audio file if available */\n    compressedFileUri?: string\n}\n\nexport interface AudioStreamStatus {\n    /** Indicates whether audio recording is currently active */\n    isRecording: boolean\n    /** Indicates whether recording is in a paused state */\n    isPaused: boolean\n    /** Duration of the current recording in milliseconds */\n    durationMs: number\n    /** Size of the recorded audio data in bytes */\n    size: number\n    /** Interval in milliseconds at which recording data is emitted */\n    interval: number\n    /** Interval in milliseconds at which analysis data is emitted */\n    intervalAnalysis: number\n    /** MIME type of the recorded audio (e.g., 'audio/wav') */\n    mimeType: string\n    /** Information about audio compression if enabled */\n    compression?: CompressionInfo\n}\n\nexport interface AudioDataEvent {\n    /** Audio data as base64 string (native) or Float32Array (web) */\n    data: string | Float32Array\n    /** Current position in the audio stream in bytes */\n    position: number\n    /** URI to the file being recorded */\n    fileUri: string\n    /** Size of the current data chunk in bytes */\n    eventDataSize: number\n    /** Total size of the recording so far in bytes */\n    totalSize: number\n    /** Information about compression if enabled, including the compressed data chunk */\n    compression?: CompressionInfo & {\n        /** Base64 (native) or Blob (web) encoded compressed data chunk */\n        data?: string | Blob\n    }\n}\n\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\nexport type SampleRate = 16000 | 44100 | 48000\nexport type BitDepth = 8 | 16 | 32\nexport type PCMFormat = `pcm_${BitDepth}bit`\n\nexport type ConsoleLike = {\n    /** Logs a message with optional arguments */\n    log: (message: string, ...args: unknown[]) => void\n    /** Logs a debug message with optional arguments */\n    debug: (message: string, ...args: unknown[]) => void\n    /** Logs an info message with optional arguments */\n    info: (message: string, ...args: unknown[]) => void\n    /** Logs a warning message with optional arguments */\n    warn: (message: string, ...args: unknown[]) => void\n    /** Logs an error message with optional arguments */\n    error: (message: string, ...args: unknown[]) => void\n}\n\nexport interface Chunk {\n    /** Transcribed text content */\n    text: string\n    /** Start and end timestamp in seconds [start, end] where end can be null if ongoing */\n    timestamp: [number, number | null]\n}\n\nexport interface TranscriberData {\n    /** Unique identifier for the transcription */\n    id: string\n    /** Indicates if the transcriber is currently processing */\n    isBusy: boolean\n    /** Complete transcribed text */\n    text: string\n    /** Start time of the transcription in milliseconds */\n    startTime: number\n    /** End time of the transcription in milliseconds */\n    endTime: number\n    /** Array of transcribed text chunks with timestamps */\n    chunks: Chunk[]\n}\n\nexport interface AudioRecording {\n    /** URI to the recorded audio file */\n    fileUri: string\n    /** Filename of the recorded audio */\n    filename: string\n    /** Duration of the recording in milliseconds */\n    durationMs: number\n    /** Size of the recording in bytes */\n    size: number\n    /** MIME type of the recorded audio */\n    mimeType: string\n    /** Number of audio channels (1 for mono, 2 for stereo) */\n    channels: number\n    /** Bit depth of the audio (8, 16, or 32 bits) */\n    bitDepth: BitDepth\n    /** Sample rate of the audio in Hz */\n    sampleRate: SampleRate\n    /** Timestamp when the recording was created */\n    createdAt?: number\n    /** Array of transcription data if available */\n    transcripts?: TranscriberData[]\n    /** Analysis data for the recording if processing was enabled */\n    analysisData?: AudioAnalysis\n    /** Information about compression if enabled, including the URI to the compressed file */\n    compression?: CompressionInfo & {\n        /** URI to the compressed audio file */\n        compressedFileUri: string\n    }\n}\n\nexport interface StartRecordingResult {\n    /** URI to the file being recorded */\n    fileUri: string\n    /** MIME type of the recording */\n    mimeType: string\n    /** Number of audio channels (1 for mono, 2 for stereo) */\n    channels?: number\n    /** Bit depth of the audio (8, 16, or 32 bits) */\n    bitDepth?: BitDepth\n    /** Sample rate of the audio in Hz */\n    sampleRate?: SampleRate\n    /** Information about compression if enabled, including the URI to the compressed file */\n    compression?: CompressionInfo & {\n        /** URI to the compressed audio file */\n        compressedFileUri: string\n    }\n}\n\nexport interface AudioSessionConfig {\n    /**\n     * Audio session category that defines the audio behavior\n     * - 'Ambient': Audio continues with silent switch, mixes with other audio\n     * - 'SoloAmbient': Audio continues with silent switch, interrupts other audio\n     * - 'Playback': Audio continues in background, interrupts other audio\n     * - 'Record': Optimized for recording, interrupts other audio\n     * - 'PlayAndRecord': Allows simultaneous playback and recording\n     * - 'MultiRoute': Routes audio to multiple outputs simultaneously\n     */\n    category?:\n        | 'Ambient'\n        | 'SoloAmbient'\n        | 'Playback'\n        | 'Record'\n        | 'PlayAndRecord'\n        | 'MultiRoute'\n    /**\n     * Audio session mode that defines the behavior for specific use cases\n     * - 'Default': Standard audio behavior\n     * - 'VoiceChat': Optimized for voice chat applications\n     * - 'VideoChat': Optimized for video chat applications\n     * - 'GameChat': Optimized for in-game chat\n     * - 'VideoRecording': Optimized for video recording\n     * - 'Measurement': Optimized for audio measurement\n     * - 'MoviePlayback': Optimized for movie playback\n     * - 'SpokenAudio': Optimized for spoken audio content\n     */\n    mode?:\n        | 'Default'\n        | 'VoiceChat'\n        | 'VideoChat'\n        | 'GameChat'\n        | 'VideoRecording'\n        | 'Measurement'\n        | 'MoviePlayback'\n        | 'SpokenAudio'\n    /**\n     * Options that modify the behavior of the audio session category\n     * - 'MixWithOthers': Allows mixing with other active audio sessions\n     * - 'DuckOthers': Reduces the volume of other audio sessions\n     * - 'InterruptSpokenAudioAndMixWithOthers': Interrupts spoken audio and mixes with others\n     * - 'AllowBluetooth': Allows audio routing to Bluetooth devices\n     * - 'AllowBluetoothA2DP': Allows audio routing to Bluetooth A2DP devices\n     * - 'AllowAirPlay': Allows audio routing to AirPlay devices\n     * - 'DefaultToSpeaker': Routes audio to the speaker by default\n     */\n    categoryOptions?: (\n        | 'MixWithOthers'\n        | 'DuckOthers'\n        | 'InterruptSpokenAudioAndMixWithOthers'\n        | 'AllowBluetooth'\n        | 'AllowBluetoothA2DP'\n        | 'AllowAirPlay'\n        | 'DefaultToSpeaker'\n    )[]\n}\n\nexport interface IOSConfig {\n    /** Configuration for the iOS audio session */\n    audioSession?: AudioSessionConfig\n}\n\n/** Web platform specific configuration options */\nexport interface WebConfig {\n    /**\n     * Whether to store uncompressed audio data for WAV generation\n     *\n     * When true, all PCM chunks are stored in memory to create a WAV file when compression is disabled\n     * When false, uncompressed audio won't be available, but memory usage will be lower\n     *\n     * Default: true (for backward compatibility)\n     */\n    storeUncompressedAudio?: boolean\n}\n\n// Add new type for interruption reasons\nexport type RecordingInterruptionReason =\n    /** Audio focus was lost to another app */\n    | 'audioFocusLoss'\n    /** Audio focus was regained */\n    | 'audioFocusGain'\n    /** Recording was interrupted by a phone call */\n    | 'phoneCall'\n    /** Phone call that interrupted recording has ended */\n    | 'phoneCallEnded'\n    /** Recording was stopped by the system or another app */\n    | 'recordingStopped'\n    /** Recording device was disconnected */\n    | 'deviceDisconnected'\n    /** Recording switched to default device after disconnection */\n    | 'deviceFallback'\n    /** A new audio device was connected */\n    | 'deviceConnected'\n    /** Device switching failed */\n    | 'deviceSwitchFailed'\n\n// Add new interface for interruption events\nexport interface RecordingInterruptionEvent {\n    /** The reason for the recording interruption */\n    reason: RecordingInterruptionReason\n    /** Indicates whether the recording is paused due to the interruption */\n    isPaused: boolean\n}\n\nexport interface AudioDeviceCapabilities {\n    /** Supported sample rates for the device */\n    sampleRates: number[]\n    /** Supported channel counts for the device */\n    channelCounts: number[]\n    /** Supported bit depths for the device */\n    bitDepths: number[]\n    /** Whether the device supports echo cancellation */\n    hasEchoCancellation?: boolean\n    /** Whether the device supports noise suppression */\n    hasNoiseSuppression?: boolean\n    /** Whether the device supports automatic gain control */\n    hasAutomaticGainControl?: boolean\n}\n\nexport interface AudioDevice {\n    /** Unique identifier for the device */\n    id: string\n    /** Human-readable name of the device */\n    name: string\n    /** Device type (builtin_mic, bluetooth, etc.) */\n    type: string\n    /** Whether this is the system default device */\n    isDefault: boolean\n    /** Audio capabilities for the device */\n    capabilities: AudioDeviceCapabilities\n    /** Whether the device is currently available */\n    isAvailable: boolean\n}\n\n/** Defines how recording should behave when a device becomes unavailable */\nexport const DeviceDisconnectionBehavior = {\n    /** Pause recording when device disconnects */\n    PAUSE: 'pause',\n    /** Switch to default device and continue recording */\n    FALLBACK: 'fallback',\n} as const\n\n/** Type for DeviceDisconnectionBehavior values */\nexport type DeviceDisconnectionBehaviorType =\n    (typeof DeviceDisconnectionBehavior)[keyof typeof DeviceDisconnectionBehavior]\n\nexport interface RecordingConfig {\n    /** Sample rate for recording in Hz (16000, 44100, or 48000) */\n    sampleRate?: SampleRate\n\n    /** Number of audio channels (1 for mono, 2 for stereo) */\n    channels?: 1 | 2\n\n    /** Encoding type for the recording (pcm_32bit, pcm_16bit, pcm_8bit) */\n    encoding?: EncodingType\n\n    /** Interval in milliseconds at which to emit recording data */\n    interval?: number\n\n    /** Interval in milliseconds at which to emit analysis data */\n    intervalAnalysis?: number\n\n    /** Keep the device awake while recording (default is false) */\n    keepAwake?: boolean\n\n    /** Show a notification during recording (default is false) */\n    showNotification?: boolean\n\n    /** Show waveform in the notification (Android only, when showNotification is true) */\n    showWaveformInNotification?: boolean\n\n    /** Configuration for the notification */\n    notification?: NotificationConfig\n\n    /** Enable audio processing (default is false) */\n    enableProcessing?: boolean\n\n    /** iOS-specific configuration */\n    ios?: IOSConfig\n\n    /** Web-specific configuration options */\n    web?: WebConfig\n\n    /** Duration of each segment in milliseconds for analysis (default: 100) */\n    segmentDurationMs?: number\n\n    /** Feature options to extract during audio processing */\n    features?: AudioFeaturesOptions\n\n    /** Callback function to handle audio stream data */\n    onAudioStream?: (_: AudioDataEvent) => Promise<void>\n\n    /** Callback function to handle audio features extraction results */\n    onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>\n\n    /** Configuration for audio compression */\n    compression?: {\n        /** Enable audio compression */\n        enabled: boolean\n        /**\n         * Format for compression\n         * - 'aac': Advanced Audio Coding - supported on all platforms\n         * - 'opus': Opus encoding - supported on Android and Web; on iOS will automatically fall back to AAC\n         */\n        format: 'aac' | 'opus'\n        /** Bitrate for compression in bits per second */\n        bitrate?: number\n    }\n\n    /** Whether to automatically resume recording after an interruption (default is false) */\n    autoResumeAfterInterruption?: boolean\n\n    /** Optional callback to handle recording interruptions */\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n\n    /** Optional directory path where output files will be saved */\n    outputDirectory?: string // If not provided, uses default app directory\n    /** Optional filename for the recording (uses UUID if not provided) */\n    filename?: string // If not provided, uses UUID\n\n    /** ID of the device to use for recording (if not specified, uses default) */\n    deviceId?: string\n\n    /** How to handle device disconnection during recording */\n    deviceDisconnectionBehavior?: DeviceDisconnectionBehaviorType\n}\n\nexport interface NotificationConfig {\n    /** Title of the notification */\n    title?: string\n\n    /** Main text content of the notification */\n    text?: string\n\n    /** Icon to be displayed in the notification (resource name or URI) */\n    icon?: string\n\n    /** Android-specific notification configuration */\n    android?: {\n        /** Unique identifier for the notification channel */\n        channelId?: string\n\n        /** User-visible name of the notification channel */\n        channelName?: string\n\n        /** User-visible description of the notification channel */\n        channelDescription?: string\n\n        /** Unique identifier for this notification */\n        notificationId?: number\n\n        /** List of actions that can be performed from the notification */\n        actions?: NotificationAction[]\n\n        /** Configuration for the waveform visualization in the notification */\n        waveform?: WaveformConfig\n\n        /** Color of the notification LED (if device supports it) */\n        lightColor?: string\n\n        /** Priority of the notification (affects how it's displayed) */\n        priority?: 'min' | 'low' | 'default' | 'high' | 'max'\n\n        /** Accent color for the notification (used for the app icon and buttons) */\n        accentColor?: string\n    }\n\n    /** iOS-specific notification configuration */\n    ios?: {\n        /** Identifier for the notification category (used for grouping similar notifications) */\n        categoryIdentifier?: string\n    }\n}\n\nexport interface NotificationAction {\n    /** Display title for the action */\n    title: string\n\n    /** Unique identifier for the action */\n    identifier: string\n\n    /** Icon to be displayed for the action (Android only) */\n    icon?: string\n}\n\nexport interface WaveformConfig {\n    /** The color of the waveform (e.g., \"#FFFFFF\" for white) */\n    color?: string // The color of the waveform (e.g., \"#FFFFFF\" for white)\n    /** Opacity of the waveform (0.0 - 1.0) */\n    opacity?: number // Opacity of the waveform (0.0 - 1.0)\n    /** Width of the waveform line (default: 1.5) */\n    strokeWidth?: number // Width of the waveform line (default: 1.5)\n    /** Drawing style: \"stroke\" for outline, \"fill\" for solid */\n    style?: 'stroke' | 'fill' // Drawing style: \"stroke\" for outline, \"fill\" for solid\n    /** Whether to mirror the waveform (symmetrical display) */\n    mirror?: boolean // Whether to mirror the waveform (symmetrical display)\n    /** Height of the waveform view in dp (default: 64) */\n    height?: number // Height of the waveform view in dp (default: 64)\n}\n\nexport interface ExtractAudioDataOptions {\n    /** URI of the audio file to extract data from */\n    fileUri: string\n    /** Start time in milliseconds (for time-based range) */\n    startTimeMs?: number\n    /** End time in milliseconds (for time-based range) */\n    endTimeMs?: number\n    /** Start position in bytes (for byte-based range) */\n    position?: number\n    /** Length in bytes to extract (for byte-based range) */\n    length?: number\n    /** Include normalized audio data in [-1, 1] range */\n    includeNormalizedData?: boolean\n    /** Include base64 encoded string representation of the audio data */\n    includeBase64Data?: boolean\n    /** Include WAV header in the PCM data (makes it a valid WAV file) */\n    includeWavHeader?: boolean\n    /** Logger for debugging - can pass console directly. */\n    logger?: ConsoleLike\n    /** Compute the checksum of the PCM data */\n    computeChecksum?: boolean\n    /** Target config for the normalized audio (Android and Web) */\n    decodingOptions?: DecodingConfig\n}\n\nexport interface ExtractedAudioData {\n    /** Raw PCM audio data */\n    pcmData: Uint8Array\n    /** Normalized audio data in [-1, 1] range (when includeNormalizedData is true) */\n    normalizedData?: Float32Array\n    /** Base64 encoded string representation of the audio data (when includeBase64Data is true) */\n    base64Data?: string\n    /** Sample rate in Hz (e.g., 44100, 48000) */\n    sampleRate: number\n    /** Number of audio channels (1 for mono, 2 for stereo) */\n    channels: number\n    /** Bits per sample (8, 16, or 32) */\n    bitDepth: BitDepth\n    /** Duration of the audio in milliseconds */\n    durationMs: number\n    /** PCM format identifier (e.g., \"pcm_16bit\") */\n    format: PCMFormat\n    /** Total number of audio samples per channel */\n    samples: number\n    /** Whether the pcmData includes a WAV header */\n    hasWavHeader?: boolean\n    /** CRC32 Checksum of PCM data */\n    checksum?: number\n}\n\nexport interface UseAudioRecorderState {\n    /**\n     * Prepares recording with the specified configuration without starting it.\n     *\n     * This method eliminates the latency between calling startRecording and the actual recording beginning.\n     * It pre-initializes all audio resources, requests permissions, and sets up audio sessions in advance,\n     * allowing for true zero-latency recording start when startRecording is called later.\n     *\n     * Technical benefits:\n     * - Eliminates audio pipeline initialization delay (50-300ms depending on platform)\n     * - Pre-allocates audio buffers to avoid memory allocation during recording start\n     * - Initializes audio hardware in advance (particularly important on iOS)\n     * - Requests and verifies permissions before the critical recording moment\n     *\n     * Use this method when:\n     * - You need zero-latency recording start (e.g., voice commands, musical applications)\n     * - You're building time-sensitive applications where missing initial audio would be problematic\n     * - You want to prepare resources during app initialization, screen loading, or preceding user interaction\n     * - You need to ensure recording starts reliably and instantly on all platforms\n     *\n     * @param config - The recording configuration, identical to what you would pass to startRecording\n     * @returns A promise that resolves when preparation is complete\n     *\n     * @example\n     * // Prepare during component mounting\n     * useEffect(() => {\n     *   prepareRecording({\n     *     sampleRate: 44100,\n     *     channels: 1,\n     *     encoding: 'pcm_16bit',\n     *   });\n     * }, []);\n     *\n     * // Later when user taps record button, it starts with zero latency\n     * const handleRecordPress = () => startRecording({\n     *   sampleRate: 44100,\n     *   channels: 1,\n     *   encoding: 'pcm_16bit',\n     * });\n     */\n    prepareRecording: (_: RecordingConfig) => Promise<void>\n    /** Starts recording with the specified configuration */\n    startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n    /** Stops the current recording and returns the recording data */\n    stopRecording: () => Promise<AudioRecording | null>\n    /** Pauses the current recording */\n    pauseRecording: () => Promise<void>\n    /** Resumes a paused recording */\n    resumeRecording: () => Promise<void>\n    /** Indicates whether recording is currently active */\n    isRecording: boolean\n    /** Indicates whether recording is in a paused state */\n    isPaused: boolean\n    /** Duration of the current recording in milliseconds */\n    durationMs: number // Duration of the recording\n    /** Size of the recorded audio in bytes */\n    size: number // Size in bytes of the recorded audio\n    /** Information about compression if enabled */\n    compression?: CompressionInfo\n    /** Analysis data for the recording if processing was enabled */\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    /** Optional callback to handle recording interruptions */\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n\n/**\n * Represents an event emitted during the trimming process to report progress.\n */\nexport interface TrimProgressEvent {\n    /**\n     * The percentage of the trimming process that has been completed, ranging from 0 to 100.\n     */\n    progress: number\n\n    /**\n     * The number of bytes that have been processed so far. This is optional and may not be provided in all implementations.\n     */\n    bytesProcessed?: number\n\n    /**\n     * The total number of bytes to process. This is optional and may not be provided in all implementations.\n     */\n    totalBytes?: number\n}\n\n/**\n * Defines a time range in milliseconds for trimming operations.\n */\nexport interface TimeRange {\n    /**\n     * The start time of the range in milliseconds.\n     */\n    startTimeMs: number\n\n    /**\n     * The end time of the range in milliseconds.\n     */\n    endTimeMs: number\n}\n\n/**\n * Options for configuring the audio trimming operation.\n */\nexport interface TrimAudioOptions {\n    /**\n     * The URI of the audio file to trim.\n     */\n    fileUri: string\n\n    /**\n     * The mode of trimming to apply.\n     * - `'single'`: Trims the audio to a single range defined by `startTimeMs` and `endTimeMs`.\n     * - `'keep'`: Keeps the specified `ranges` and removes all other portions of the audio.\n     * - `'remove'`: Removes the specified `ranges` and keeps the remaining portions of the audio.\n     * @default 'single'\n     */\n    mode?: 'single' | 'keep' | 'remove'\n\n    /**\n     * An array of time ranges to keep or remove, depending on the `mode`.\n     * - Required for `'keep'` and `'remove'` modes.\n     * - Ignored when `mode` is `'single'`.\n     */\n    ranges?: TimeRange[]\n\n    /**\n     * The start time in milliseconds for the `'single'` mode.\n     * - If not provided, trimming starts from the beginning of the audio (0 ms).\n     */\n    startTimeMs?: number\n\n    /**\n     * The end time in milliseconds for the `'single'` mode.\n     * - If not provided, trimming extends to the end of the audio.\n     */\n    endTimeMs?: number\n\n    /**\n     * The name of the output file. If not provided, a default name will be generated.\n     */\n    outputFileName?: string\n\n    /**\n     * Configuration for the output audio format.\n     */\n    outputFormat?: {\n        /**\n         * The format of the output audio file.\n         * - `'wav'`: Waveform Audio File Format (uncompressed).\n         * - `'aac'`: Advanced Audio Coding (compressed). Not supported on web platforms.\n         * - `'opus'`: Opus Interactive Audio Codec (compressed).\n         */\n        format: 'wav' | 'aac' | 'opus'\n\n        /**\n         * The sample rate of the output audio in Hertz (Hz).\n         * - If not provided, the input audio's sample rate is used.\n         */\n        sampleRate?: number\n\n        /**\n         * The number of channels in the output audio (e.g., 1 for mono, 2 for stereo).\n         * - If not provided, the input audio's channel count is used.\n         */\n        channels?: number\n\n        /**\n         * The bit depth of the output audio, applicable to PCM formats like `'wav'`.\n         * - If not provided, the input audio's bit depth is used.\n         */\n        bitDepth?: number\n\n        /**\n         * The bitrate of the output audio in bits per second, applicable to compressed formats like `'aac'`.\n         * - If not provided, a default bitrate is used based on the format.\n         */\n        bitrate?: number\n    }\n\n    /**\n     * Options for decoding the input audio file.\n     * - See `DecodingConfig` for details.\n     */\n    decodingOptions?: DecodingConfig\n}\n\n/**\n * Result of the audio trimming operation.\n */\nexport interface TrimAudioResult {\n    /**\n     * The URI of the trimmed audio file.\n     */\n    uri: string\n\n    /**\n     * The filename of the trimmed audio file.\n     */\n    filename: string\n\n    /**\n     * The duration of the trimmed audio in milliseconds.\n     */\n    durationMs: number\n\n    /**\n     * The size of the trimmed audio file in bytes.\n     */\n    size: number\n\n    /**\n     * The sample rate of the trimmed audio in Hertz (Hz).\n     */\n    sampleRate: number\n\n    /**\n     * The number of channels in the trimmed audio (e.g., 1 for mono, 2 for stereo).\n     */\n    channels: number\n\n    /**\n     * The bit depth of the trimmed audio, applicable to PCM formats like `'wav'`.\n     */\n    bitDepth: number\n\n    /**\n     * The MIME type of the trimmed audio file (e.g., `'audio/wav'`, `'audio/mpeg'`).\n     */\n    mimeType: string\n\n    /**\n     * Information about compression if the output format is compressed.\n     */\n    compression?: {\n        /**\n         * The format of the compression (e.g., `'aac'`, `'mp3'`, `'opus'`).\n         */\n        format: string\n\n        /**\n         * The bitrate of the compressed audio in bits per second.\n         */\n        bitrate: number\n\n        /**\n         * The size of the compressed audio file in bytes.\n         */\n        size: number\n    }\n\n    /**\n     * Information about the processing time.\n     */\n    processingInfo?: {\n        /**\n         * The time it took to process the audio in milliseconds.\n         */\n        durationMs: number\n    }\n}\n"]}

package/build/ExpoAudioStream.web.d.ts.map

@@ -1 +0,0 @@
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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            this.logger?.debug('Requesting user media (microphone)...')\n\n            // First check if the browser supports the necessary audio APIs\n            if (!navigator?.mediaDevices?.getUserMedia) {\n                this.logger?.error(\n                    'Browser does not support mediaDevices.getUserMedia'\n                )\n                throw new Error('Browser does not support audio recording')\n            }\n\n            // Get media with detailed audio constraints for better diagnostics\n            const constraints = {\n                audio: {\n                    echoCancellation: true,\n                    noiseSuppression: true,\n                    autoGainControl: true,\n                    // Add deviceId constraint if specified\n                    ...(this.recordingConfig?.deviceId\n                        ? {\n                              deviceId: {\n                                  exact: this.recordingConfig.deviceId,\n                              },\n                          }\n                        : {}),\n                },\n            }\n\n            this.logger?.debug('Media constraints:', constraints)\n\n            const stream =\n                await navigator.mediaDevices.getUserMedia(constraints)\n\n            // Get detailed info about the audio track for debugging\n            const audioTracks = stream.getAudioTracks()\n            if (audioTracks.length > 0) {\n                const track = audioTracks[0]\n                const settings = track.getSettings()\n                this.logger?.debug('Audio track obtained:', {\n                    label: track.label,\n                    id: track.id,\n                    enabled: track.enabled,\n                    muted: track.muted,\n                    readyState: track.readyState,\n                    settings,\n                })\n            } else {\n                this.logger?.warn('Stream has no audio tracks!')\n            }\n\n            return stream\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, uncompressedBlob } =\n                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            // Handle both compressed and uncompressed blobs according to configuration\n            const compressionEnabled =\n                this.recordingConfig?.compression?.enabled ?? false\n\n            // Process compressed blob if available\n            if (compressedBlob) {\n                const compressedUri = URL.createObjectURL(compressedBlob)\n                const compressedInfo = {\n                    compressedFileUri: compressedUri,\n                    size: compressedBlob.size,\n                    mimeType: 'audio/webm',\n                    format: 'opus',\n                    bitrate:\n                        this.recordingConfig?.compression?.bitrate ?? 128000,\n                }\n\n                // If compression is enabled, use compressed blob as primary format\n                if (compressionEnabled) {\n                    this.logger?.debug(\n                        'Using compressed audio as primary output'\n                    )\n                    fileUri = compressedUri\n                    mimeType = 'audio/webm'\n\n                    // Store compression info\n                    compression = compressedInfo\n                } else {\n                    // Compression was enabled during recording but not set as primary\n                    // Store as alternate format\n                    compression = compressedInfo\n                }\n            }\n\n            // Process uncompressed WAV if available\n            if (uncompressedBlob) {\n                const wavUri = URL.createObjectURL(uncompressedBlob)\n\n                // If compression is disabled or no compressed blob is available,\n                // use WAV as primary format\n                if (!compressionEnabled || !compressedBlob) {\n                    this.logger?.debug(\n                        'Using uncompressed WAV as primary output'\n                    )\n                    fileUri = wavUri\n                    mimeType = 'audio/wav'\n                }\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            // Reset recording state variables to prepare for next recording\n            this.currentDurationMs = 0\n            this.currentSize = 0\n            this.lastEmittedSize = 0\n            this.totalCompressedSize = 0\n            this.lastEmittedCompressionSize = 0\n            this.audioChunks = []\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 +0,0 @@
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package/build/ExpoAudioStreamModule.js.map

@@ -1 +0,0 @@
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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"]}

package/build/WebRecorder.web.d.ts.map

@@ -1 +0,0 @@
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