@siteed/expo-audio-studio 2.7.0 → 2.8.1

package/build/esm/ExpoAudioStream.web.js.map

@@ -0,0 +1 @@
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src/ExpoAudioStreamModule.web.ts\nimport { LegacyEventEmitter } from 'expo-modules-core'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport {\n    AudioRecording,\n    AudioStreamStatus,\n    BitDepth,\n    ConsoleLike,\n    RecordingConfig,\n    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/esm/ExpoAudioStreamModule.js.map

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

@@ -0,0 +1 @@
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packages/expo-audio-stream/src/WebRecorder.web.ts\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { ConsoleLike, RecordingConfig } from './ExpoAudioStream.types'\nimport {\n    EmitAudioAnalysisFunction,\n    EmitAudioEventFunction,\n} from './ExpoAudioStream.web'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\nimport { writeWavHeader } from './utils/writeWavHeader'\nimport { InlineFeaturesExtractor } from './workers/InlineFeaturesExtractor.web'\nimport { InlineAudioWebWorker } from './workers/inlineAudioWebWorker.web'\n\ninterface AudioWorkletEvent {\n    data: {\n        command: string\n        recordedData?: Float32Array\n        sampleRate?: number\n        position?: number\n        message?: string // For debug messages\n    }\n}\n\ninterface AudioFeaturesEvent {\n    data: {\n        command: string\n        result: AudioAnalysis\n    }\n}\n\nconst DEFAULT_WEB_BITDEPTH = 32\nconst DEFAULT_SEGMENT_DURATION_MS = 100\nconst DEFAULT_WEB_INTERVAL = 500\nconst DEFAULT_WEB_NUMBER_OF_CHANNELS = 1\n\nconst TAG = 'WebRecorder'\n\nexport class WebRecorder {\n    public audioContext: AudioContext\n    private audioWorkletNode!: AudioWorkletNode\n    private featureExtractorWorker?: Worker\n    private source: MediaStreamAudioSourceNode\n    private emitAudioEventCallback: EmitAudioEventFunction\n    private emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n    private config: RecordingConfig\n    private position: number = 0\n    private numberOfChannels: number // Number of audio channels\n    private bitDepth: number // Bit depth of the audio\n    private exportBitDepth: number // Bit depth of the audio\n    private audioAnalysisData: AudioAnalysis // Keep updating the full audio analysis data with latest events\n    private logger?: ConsoleLike\n    private compressedMediaRecorder: MediaRecorder | null = null\n    private compressedChunks: Blob[] = []\n    private compressedSize: number = 0\n    private pendingCompressedChunk: Blob | null = null\n    private dataPointIdCounter: number = 0 // Add this property to track the counter\n    private deviceDisconnectionHandler: (() => void) | null = null\n    private mediaStream: MediaStream | null = null\n    private onInterruptionCallback?: (event: {\n        reason: string\n        isPaused: boolean\n        timestamp: number\n    }) => void\n    private _isDeviceDisconnected: boolean = false\n    private pcmData: Float32Array | null = null // Store original PCM data\n    private totalSampleCount: number = 0\n\n    /**\n     * Flag to indicate whether this is the first audio chunk after a device switch\n     * Used to maintain proper duration counting\n     */\n    public isFirstChunkAfterSwitch: boolean = false\n\n    /**\n     * Gets whether the recording device has been disconnected\n     */\n    get isDeviceDisconnected(): boolean {\n        return this._isDeviceDisconnected\n    }\n\n    /**\n     * Initializes a new WebRecorder instance for audio recording and processing\n     * @param audioContext - The AudioContext to use for recording\n     * @param source - The MediaStreamAudioSourceNode providing the audio input\n     * @param recordingConfig - Configuration options for the recording\n     * @param emitAudioEventCallback - Callback function for audio data events\n     * @param emitAudioAnalysisCallback - Callback function for audio analysis events\n     * @param onInterruption - Callback for recording interruptions\n     * @param logger - Optional logger for debugging information\n     */\n    constructor({\n        audioContext,\n        source,\n        recordingConfig,\n        emitAudioEventCallback,\n        emitAudioAnalysisCallback,\n        onInterruption,\n        logger,\n    }: {\n        audioContext: AudioContext\n        source: MediaStreamAudioSourceNode\n        recordingConfig: RecordingConfig\n        emitAudioEventCallback: EmitAudioEventFunction\n        emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n        onInterruption?: (event: {\n            reason: string\n            isPaused: boolean\n            timestamp: number\n        }) => void\n        logger?: ConsoleLike\n    }) {\n        this.audioContext = audioContext\n        this.source = source\n        this.emitAudioEventCallback = emitAudioEventCallback\n        this.emitAudioAnalysisCallback = emitAudioAnalysisCallback\n        this.config = recordingConfig\n        this.logger = logger\n\n        const audioContextFormat = this.checkAudioContextFormat({\n            sampleRate: this.audioContext.sampleRate,\n        })\n        this.logger?.debug('Initialized WebRecorder with config:', {\n            sampleRate: audioContextFormat.sampleRate,\n            bitDepth: audioContextFormat.bitDepth,\n            numberOfChannels: audioContextFormat.numberOfChannels,\n        })\n\n        this.bitDepth = audioContextFormat.bitDepth\n        this.numberOfChannels =\n            audioContextFormat.numberOfChannels ||\n            DEFAULT_WEB_NUMBER_OF_CHANNELS // Default to 1 if not available\n        this.exportBitDepth =\n            encodingToBitDepth({\n                encoding: recordingConfig.encoding ?? 'pcm_32bit',\n            }) ||\n            audioContextFormat.bitDepth ||\n            DEFAULT_WEB_BITDEPTH\n\n        this.audioAnalysisData = {\n            amplitudeRange: { min: 0, max: 0 },\n            rmsRange: { min: 0, max: 0 },\n            dataPoints: [],\n            durationMs: 0,\n            samples: 0,\n            bitDepth: this.bitDepth,\n            numberOfChannels: this.numberOfChannels,\n            sampleRate: this.config.sampleRate || this.audioContext.sampleRate,\n            segmentDurationMs:\n                this.config.segmentDurationMs ?? DEFAULT_SEGMENT_DURATION_MS, // Default to 100ms segments\n        }\n\n        if (recordingConfig.enableProcessing) {\n            this.initFeatureExtractorWorker()\n        }\n\n        // Initialize compressed recording if enabled\n        if (recordingConfig.compression?.enabled) {\n            this.initializeCompressedRecorder()\n        }\n\n        this.mediaStream = source.mediaStream\n        this.onInterruptionCallback = onInterruption\n\n        // Setup device disconnection detection\n        this.setupDeviceDisconnectionDetection()\n    }\n\n    /**\n     * Initializes the audio worklet using an inline script\n     * Creates and connects the audio processing pipeline\n     */\n    async init() {\n        try {\n            // Create and use inline audio worklet\n            const blob = new Blob([InlineAudioWebWorker], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            await this.audioContext.audioWorklet.addModule(url)\n\n            this.audioWorkletNode = new AudioWorkletNode(\n                this.audioContext,\n                'recorder-processor'\n            )\n\n            this.audioWorkletNode.port.onmessage = async (\n                event: AudioWorkletEvent\n            ) => {\n                const command = event.data.command\n                if (command === 'debug') {\n                    this.logger?.debug(`[AudioWorklet] ${event.data.message}`)\n                    return\n                }\n\n                if (command !== 'newData') return\n\n                const pcmBufferFloat = event.data.recordedData\n                if (!pcmBufferFloat) {\n                    this.logger?.warn('Received empty audio buffer', event)\n                    return\n                }\n\n                // Process data in smaller chunks and emit immediately\n                const sampleRate =\n                    event.data.sampleRate ?? this.audioContext.sampleRate\n                // Use chunk size from config interval or default to 2 seconds\n                const intervalMs = this.config.interval ?? DEFAULT_WEB_INTERVAL\n                const chunkSize = Math.floor(sampleRate * (intervalMs / 1000))\n                const duration = pcmBufferFloat.length / sampleRate\n\n                // Use incoming position if provided by worklet, otherwise use our tracked position\n                const incomingPosition =\n                    typeof event.data.position === 'number'\n                        ? event.data.position\n                        : this.position\n\n                // Calculate bytes per sample based on bit depth\n                const bytesPerSample = this.bitDepth / 8\n\n                // Emit chunks without storing them\n                for (let i = 0; i < pcmBufferFloat.length; i += chunkSize) {\n                    const chunk = pcmBufferFloat.slice(i, i + chunkSize)\n                    const chunkPosition = incomingPosition + i / sampleRate\n\n                    // Calculate byte positions and samples\n                    const startPosition = Math.floor(i * bytesPerSample)\n                    const endPosition = Math.floor(\n                        (i + chunk.length) * bytesPerSample\n                    )\n                    const samples = chunk.length // Number of samples in this chunk\n\n                    // Process features if enabled\n                    if (\n                        this.config.enableProcessing &&\n                        this.featureExtractorWorker\n                    ) {\n                        this.featureExtractorWorker.postMessage({\n                            command: 'process',\n                            channelData: chunk,\n                            sampleRate,\n                            segmentDurationMs:\n                                this.config.segmentDurationMs ??\n                                DEFAULT_SEGMENT_DURATION_MS, // Default to 100ms\n                            bitDepth: this.bitDepth,\n                            fullAudioDurationMs: chunkPosition * 1000,\n                            numberOfChannels: this.numberOfChannels,\n                            features: this.config.features,\n                            intervalAnalysis: this.config.intervalAnalysis,\n                            startPosition,\n                            endPosition,\n                            samples,\n                        })\n                    }\n\n                    // Only store PCM data if web.storeUncompressedAudio is not explicitly false\n                    const shouldStoreUncompressed =\n                        this.config.web?.storeUncompressedAudio !== false\n\n                    // Store PCM chunks when needed\n                    if (shouldStoreUncompressed) {\n                        // Store the original Float32Array data for later WAV creation\n                        this.appendPcmData(chunk)\n                        this.totalSampleCount += chunk.length\n                    }\n\n                    // Emit chunk immediately\n                    this.emitAudioEventCallback({\n                        data: chunk,\n                        position: chunkPosition,\n                        compression: this.pendingCompressedChunk\n                            ? {\n                                  data: this.pendingCompressedChunk,\n                                  size: this.pendingCompressedChunk.size,\n                                  totalSize: this.compressedSize,\n                                  mimeType: 'audio/webm',\n                                  format: 'opus',\n                                  bitrate:\n                                      this.config.compression?.bitrate ??\n                                      128000,\n                              }\n                            : undefined,\n                    })\n                }\n\n                // Update our position based on the worklet's position if provided\n                this.position = incomingPosition + duration\n                this.pendingCompressedChunk = null\n            }\n\n            // Ensure we use all relevant settings from config\n            const recordSampleRate = this.audioContext.sampleRate\n            const exportSampleRate =\n                this.config.sampleRate ?? this.audioContext.sampleRate\n            const channels = this.config.channels ?? this.numberOfChannels\n            const interval = this.config.interval ?? DEFAULT_WEB_INTERVAL\n\n            this.logger?.debug(`WebRecorder initialized with config:`, {\n                recordSampleRate,\n                exportSampleRate,\n                bitDepth: this.bitDepth,\n                exportBitDepth: this.exportBitDepth,\n                channels,\n                interval,\n                position: this.position,\n                deviceId: this.config.deviceId || 'default',\n                compression: this.config.compression\n                    ? {\n                          enabled: this.config.compression.enabled,\n                          format: this.config.compression.format,\n                          bitrate: this.config.compression.bitrate,\n                      }\n                    : 'disabled',\n            })\n\n            // Initialize the worklet with all settings from config\n            this.audioWorkletNode.port.postMessage({\n                command: 'init',\n                recordSampleRate,\n                exportSampleRate,\n                bitDepth: this.bitDepth,\n                exportBitDepth: this.exportBitDepth,\n                channels,\n                interval,\n                position: this.position, // Pass the current position to the processor\n                enableLogging: true,\n            })\n\n            // Connect the source to the AudioWorkletNode and start recording\n            this.source.connect(this.audioWorkletNode)\n            this.audioWorkletNode.connect(this.audioContext.destination)\n        } catch (error) {\n            console.error(`[${TAG}] Failed to initialize WebRecorder`, error)\n        }\n    }\n\n    /**\n     * Append new PCM data to the existing buffer\n     * @param newData New Float32Array data to append\n     */\n    private appendPcmData(newData: Float32Array): void {\n        // Clone the incoming data to ensure it's not modified\n        const dataToAdd = new Float32Array(newData)\n\n        if (!this.pcmData) {\n            // First chunk - create a copy to avoid references to original data\n            this.pcmData = new Float32Array(dataToAdd)\n            return\n        }\n\n        // Create a new buffer with increased size\n        const newBuffer = new Float32Array(\n            this.pcmData.length + dataToAdd.length\n        )\n\n        // Copy existing data\n        newBuffer.set(this.pcmData)\n\n        // Append new data\n        newBuffer.set(dataToAdd, this.pcmData.length)\n\n        // Replace existing buffer\n        this.pcmData = newBuffer\n    }\n\n    /**\n     * Initializes the feature extractor worker for audio analysis\n     * Creates an inline worker from a blob for audio feature extraction\n     */\n    initFeatureExtractorWorker() {\n        try {\n            const blob = new Blob([InlineFeaturesExtractor], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            this.featureExtractorWorker = new Worker(url)\n            this.featureExtractorWorker.onmessage =\n                this.handleFeatureExtractorMessage.bind(this)\n            this.featureExtractorWorker.onerror = (error) => {\n                console.error(`[${TAG}] Feature extractor worker error:`, error)\n            }\n\n            // Initialize worker with counter if needed\n            if (this.dataPointIdCounter > 0) {\n                this.featureExtractorWorker.postMessage({\n                    command: 'resetCounter',\n                    value: this.dataPointIdCounter,\n                })\n                this.logger?.debug(\n                    `Initialized worker with counter value ${this.dataPointIdCounter}`\n                )\n            }\n\n            this.logger?.log(\n                'Feature extractor worker initialized successfully'\n            )\n        } catch (error) {\n            console.error(\n                `[${TAG}] Failed to initialize feature extractor worker`,\n                error\n            )\n        }\n    }\n\n    /**\n     * Processes audio analysis results from the feature extractor worker\n     * Updates the audio analysis data and emits events\n     * @param event - The event containing audio analysis results\n     */\n    handleFeatureExtractorMessage(event: AudioFeaturesEvent) {\n        if (event.data.command === 'features') {\n            const segmentResult = event.data.result\n\n            // Track existing IDs to prevent duplicates\n            const existingIds = new Set(\n                this.audioAnalysisData.dataPoints.map((dp) => dp.id)\n            )\n\n            // Filter out datapoints with duplicate IDs\n            const uniqueNewDataPoints = segmentResult.dataPoints.filter(\n                (dp) => {\n                    return !existingIds.has(dp.id)\n                }\n            )\n\n            // Log filtered duplicates if any\n            if (\n                uniqueNewDataPoints.length < segmentResult.dataPoints.length &&\n                this.logger?.warn\n            ) {\n                this.logger.warn(\n                    `Filtered ${segmentResult.dataPoints.length - uniqueNewDataPoints.length} duplicate datapoints`\n                )\n            }\n\n            // Update counter based on the highest ID seen\n            if (uniqueNewDataPoints.length > 0) {\n                const lastDataPoint =\n                    uniqueNewDataPoints[uniqueNewDataPoints.length - 1]\n\n                if (lastDataPoint && typeof lastDataPoint.id === 'number') {\n                    const nextIdValue = lastDataPoint.id + 1\n\n                    if (nextIdValue > this.dataPointIdCounter) {\n                        this.dataPointIdCounter = nextIdValue\n                        this.logger?.debug(\n                            `Counter updated to ${this.dataPointIdCounter}`\n                        )\n                    }\n                }\n            }\n\n            // Add unique data points to our analysis data\n            this.audioAnalysisData.dataPoints.push(...uniqueNewDataPoints)\n            this.audioAnalysisData.durationMs += segmentResult.durationMs\n            this.audioAnalysisData.sampleRate = segmentResult.sampleRate\n\n            // Merge amplitude ranges\n            if (segmentResult.amplitudeRange) {\n                if (!this.audioAnalysisData.amplitudeRange) {\n                    this.audioAnalysisData.amplitudeRange = {\n                        ...segmentResult.amplitudeRange,\n                    }\n                } else {\n                    this.audioAnalysisData.amplitudeRange = {\n                        min: Math.min(\n                            this.audioAnalysisData.amplitudeRange.min,\n                            segmentResult.amplitudeRange.min\n                        ),\n                        max: Math.max(\n                            this.audioAnalysisData.amplitudeRange.max,\n                            segmentResult.amplitudeRange.max\n                        ),\n                    }\n                }\n            }\n\n            // Merge RMS ranges\n            if (segmentResult.rmsRange) {\n                if (!this.audioAnalysisData.rmsRange) {\n                    this.audioAnalysisData.rmsRange = {\n                        ...segmentResult.rmsRange,\n                    }\n                } else {\n                    this.audioAnalysisData.rmsRange = {\n                        min: Math.min(\n                            this.audioAnalysisData.rmsRange.min,\n                            segmentResult.rmsRange.min\n                        ),\n                        max: Math.max(\n                            this.audioAnalysisData.rmsRange.max,\n                            segmentResult.rmsRange.max\n                        ),\n                    }\n                }\n            }\n\n            // Send filtered result to avoid duplicate IDs\n            const filteredSegmentResult = {\n                ...segmentResult,\n                dataPoints: uniqueNewDataPoints,\n            }\n\n            this.emitAudioAnalysisCallback(filteredSegmentResult)\n        }\n    }\n\n    /**\n     * Reset the data point counter to a specific value or zero\n     * @param startCounterFrom Optional value to start the counter from (for continuing from previous recordings)\n     */\n    resetDataPointCounter(startCounterFrom?: number): void {\n        // Set the counter with the passed value or 0\n        this.dataPointIdCounter =\n            startCounterFrom !== undefined ? startCounterFrom : 0\n        this.logger?.debug(\n            `Reset data point counter to ${this.dataPointIdCounter}`\n        )\n\n        // Update worker counter if available\n        if (this.featureExtractorWorker) {\n            this.featureExtractorWorker.postMessage({\n                command: 'resetCounter',\n                value: this.dataPointIdCounter,\n            })\n        } else {\n            this.logger?.warn(\n                'No feature extractor worker available to update counter'\n            )\n        }\n    }\n\n    /**\n     * Get the current data point counter value\n     * @returns The current value of the data point counter\n     */\n    getDataPointCounter(): number {\n        return this.dataPointIdCounter\n    }\n\n    /**\n     * Prepares the recorder for continuity after device switch\n     * Sets up all necessary state to maintain proper recording continuity\n     */\n    prepareForDeviceSwitch(): void {\n        this.isFirstChunkAfterSwitch = true\n        this.logger?.debug(\n            `Prepared for device switch at position ${this.position}s`\n        )\n    }\n\n    /**\n     * Starts the audio recording process\n     * Connects the audio nodes and begins capturing audio data\n     * @param preserveCounters If true, do not reset the counter (used for device switching)\n     */\n    start(preserveCounters = false) {\n        this.source.connect(this.audioWorkletNode)\n        this.audioWorkletNode.connect(this.audioContext.destination)\n\n        // Only reset the counter when not preserving state (e.g., for a fresh recording)\n        if (!preserveCounters) {\n            this.logger?.debug(\n                'Starting fresh recording, resetting counter to 0'\n            )\n            this.resetDataPointCounter(0) // Explicitly reset to 0 for new recordings\n            this.isFirstChunkAfterSwitch = false\n\n            // Clear PCM data for new recording\n            this.pcmData = null\n            this.totalSampleCount = 0\n        } else {\n            this.logger?.debug(\n                `Preserving counter at ${this.dataPointIdCounter} during device switch`\n            )\n        }\n\n        if (this.compressedMediaRecorder) {\n            this.compressedMediaRecorder.start(this.config.interval ?? 1000)\n        }\n    }\n\n    /**\n     * Creates a WAV file from the stored PCM data\n     */\n    private createWavFromPcmData(): Blob | null {\n        try {\n            // Check if we have PCM data\n            if (!this.pcmData || this.pcmData.length === 0) {\n                this.logger?.warn('No PCM data available to create WAV file')\n                return null\n            }\n\n            const sampleRate =\n                this.config.sampleRate || this.audioContext.sampleRate\n            const channels = this.numberOfChannels || 1\n\n            // Convert float32 PCM data to 16-bit PCM for WAV\n            const bytesPerSample = 2 // 16-bit = 2 bytes\n            const dataLength = this.pcmData.length * bytesPerSample\n            const buffer = new ArrayBuffer(dataLength)\n            const view = new DataView(buffer)\n\n            // Convert Float32Array (-1 to 1) to Int16Array (-32768 to 32767)\n            for (let i = 0; i < this.pcmData.length; i++) {\n                const sample = Math.max(-1, Math.min(1, this.pcmData[i]))\n                const int16Value = Math.round(sample * 32767)\n                view.setInt16(i * 2, int16Value, true)\n            }\n\n            // Use the existing writeWavHeader utility to add a WAV header\n            const wavBuffer = writeWavHeader({\n                buffer,\n                sampleRate,\n                numChannels: channels,\n                bitDepth: 16,\n                isFloat: false,\n            })\n\n            return new Blob([wavBuffer], { type: 'audio/wav' })\n        } catch (error) {\n            this.logger?.error('Error creating WAV file from PCM data:', error)\n            return null\n        }\n    }\n\n    /**\n     * Stops the audio recording process and returns the recorded data\n     * @returns Promise resolving to an object containing compressed and/or uncompressed blobs\n     */\n    async stop(): Promise<{ compressedBlob?: Blob; uncompressedBlob?: Blob }> {\n        try {\n            // Stop any compressed recording first\n            if (\n                this.compressedMediaRecorder &&\n                this.compressedMediaRecorder.state !== 'inactive'\n            ) {\n                this.compressedMediaRecorder.stop()\n            }\n\n            // Wait for any pending compressed chunks to be processed\n            if (this.compressedMediaRecorder) {\n                // Small delay to ensure all data is processed\n                await new Promise((resolve) => setTimeout(resolve, 100))\n            }\n\n            // Create uncompressed WAV file from the PCM data\n            let uncompressedBlob: Blob | undefined\n\n            // Only create WAV if we have PCM data\n            if (this.pcmData && this.pcmData.length > 0) {\n                uncompressedBlob =\n                    (await this.createWavFromPcmData()) || undefined\n            }\n\n            // Return the compressed and/or uncompressed blobs if available\n            return {\n                compressedBlob:\n                    this.compressedChunks.length > 0\n                        ? new Blob(this.compressedChunks, {\n                              type: 'audio/webm;codecs=opus',\n                          })\n                        : undefined,\n                uncompressedBlob,\n            }\n        } finally {\n            this.cleanup()\n            // Reset the chunks array\n            this.compressedChunks = []\n            this.compressedSize = 0\n            this.pendingCompressedChunk = null\n            this.pcmData = null\n            this.totalSampleCount = 0\n        }\n    }\n\n    /**\n     * Cleans up resources when recording is stopped\n     * Closes audio context and disconnects nodes\n     */\n    public cleanup() {\n        // Remove device disconnection handler\n        if (this.deviceDisconnectionHandler) {\n            this.deviceDisconnectionHandler()\n            this.deviceDisconnectionHandler = null\n        }\n\n        // Check if AudioContext is already closed before attempting to close it\n        if (this.audioContext && this.audioContext.state !== 'closed') {\n            try {\n                this.audioContext.close()\n            } catch (e) {\n                // Ignore closure errors - this happens if already closed\n            }\n        }\n\n        // Safely disconnect audioWorkletNode if it exists\n        if (this.audioWorkletNode) {\n            try {\n                this.audioWorkletNode.disconnect()\n            } catch (e) {\n                // Ignore disconnection errors - node might be already disconnected\n            }\n        }\n\n        // Safely disconnect source if it exists\n        if (this.source) {\n            try {\n                this.source.disconnect()\n            } catch (e) {\n                // Ignore disconnection errors - source might be already disconnected\n            }\n        }\n\n        // Always stop media stream tracks to release hardware resources\n        this.stopMediaStreamTracks()\n\n        // Mark as disconnected to prevent future errors\n        this._isDeviceDisconnected = true\n    }\n\n    /**\n     * Pauses the audio recording process\n     * Disconnects audio nodes and pauses the media recorder\n     */\n    pause() {\n        try {\n            // Note: We're just pausing, not disconnecting the device\n            // Simply disconnect nodes temporarily without marking device as disconnected\n            this.source.disconnect(this.audioWorkletNode)\n            this.audioWorkletNode.disconnect(this.audioContext.destination)\n            this.audioWorkletNode.port.postMessage({ command: 'pause' })\n\n            if (this.compressedMediaRecorder?.state === 'recording') {\n                this.compressedMediaRecorder.pause()\n            }\n\n            this.logger?.debug('Recording paused successfully')\n        } catch (error) {\n            this.logger?.error('Error in pause(): ', error)\n            // Already disconnected, just ignore and continue\n        }\n    }\n\n    /**\n     * Stops all media stream tracks to release hardware resources\n     * Ensures recording indicators (like microphone icon) are turned off\n     */\n    public stopMediaStreamTracks() {\n        // Stop all audio tracks to stop the recording icon\n        if (this.mediaStream) {\n            const tracks = this.mediaStream.getTracks()\n            tracks.forEach((track) => track.stop())\n        } else if (this.source?.mediaStream) {\n            const tracks = this.source.mediaStream.getTracks()\n            tracks.forEach((track) => track.stop())\n        }\n    }\n\n    /**\n     * Determines the audio format capabilities of the current audio context\n     * @param sampleRate - The sample rate to check\n     * @returns Object containing format information (sample rate, bit depth, channels)\n     */\n    private checkAudioContextFormat({ sampleRate }: { sampleRate: number }) {\n        // Create a silent AudioBuffer\n        const frameCount = sampleRate * 1.0 // 1 second buffer\n        const audioBuffer = this.audioContext.createBuffer(\n            1,\n            frameCount,\n            sampleRate\n        )\n\n        // Check the format\n        const channelData = audioBuffer.getChannelData(0)\n        const bitDepth = channelData.BYTES_PER_ELEMENT * 8 // 4 bytes per element means 32-bit\n\n        return {\n            sampleRate: audioBuffer.sampleRate,\n            bitDepth,\n            numberOfChannels: audioBuffer.numberOfChannels,\n        }\n    }\n\n    /**\n     * Resumes a paused recording\n     * Reconnects audio nodes and resumes the media recorder\n     */\n    resume() {\n        // If device was disconnected, we can't resume\n        if (this._isDeviceDisconnected) {\n            this.logger?.warn('Cannot resume recording: device disconnected')\n            return\n        }\n\n        try {\n            this.source.connect(this.audioWorkletNode)\n            this.audioWorkletNode.connect(this.audioContext.destination)\n            this.audioWorkletNode.port.postMessage({ command: 'resume' })\n            this.compressedMediaRecorder?.resume()\n        } catch (error) {\n            this.logger?.error('Error in resume(): ', error)\n        }\n    }\n\n    /**\n     * Initializes the compressed media recorder if compression is enabled\n     * Sets up event handlers for compressed audio data\n     */\n    private initializeCompressedRecorder() {\n        try {\n            const mimeType = 'audio/webm;codecs=opus'\n            if (!MediaRecorder.isTypeSupported(mimeType)) {\n                this.logger?.warn(\n                    'Opus compression not supported in this browser'\n                )\n                return\n            }\n\n            this.compressedMediaRecorder = new MediaRecorder(\n                this.source.mediaStream,\n                {\n                    mimeType,\n                    audioBitsPerSecond:\n                        this.config.compression?.bitrate ?? 128000,\n                }\n            )\n\n            this.compressedMediaRecorder.ondataavailable = (event) => {\n                if (event.data.size > 0) {\n                    this.compressedChunks.push(event.data)\n                    this.compressedSize += event.data.size\n                    this.pendingCompressedChunk = event.data\n                }\n            }\n        } catch (error) {\n            this.logger?.error(\n                'Failed to initialize compressed recorder:',\n                error\n            )\n        }\n    }\n\n    /**\n     * Processes features if enabled\n     */\n    processFeatures(\n        chunk: Float32Array,\n        sampleRate: number,\n        chunkPosition: number,\n        startPosition: number,\n        endPosition: number,\n        samples: number\n    ) {\n        if (this.config.enableProcessing && this.featureExtractorWorker) {\n            this.featureExtractorWorker.postMessage({\n                command: 'process',\n                channelData: chunk,\n                sampleRate,\n                segmentDurationMs:\n                    this.config.segmentDurationMs ??\n                    DEFAULT_SEGMENT_DURATION_MS, // Default to 100ms\n                bitDepth: this.bitDepth,\n                fullAudioDurationMs: chunkPosition * 1000,\n                numberOfChannels: this.numberOfChannels,\n                features: this.config.features,\n                intervalAnalysis: this.config.intervalAnalysis,\n                startPosition,\n                endPosition,\n                samples,\n            })\n        }\n    }\n\n    /**\n     * Sets up detection for device disconnection events\n     */\n    private setupDeviceDisconnectionDetection() {\n        if (!this.mediaStream) return\n\n        // Function to handle track ending (which happens on device disconnection)\n        const handleTrackEnded = () => {\n            this.logger?.warn('Audio track ended - device disconnected')\n            this._isDeviceDisconnected = true\n\n            // Use the callback to notify parent component about device disconnection\n            if (this.onInterruptionCallback) {\n                this.onInterruptionCallback({\n                    reason: 'deviceDisconnected',\n                    isPaused: true,\n                    timestamp: Date.now(),\n                })\n                this.logger?.debug('Notified about device disconnection')\n            }\n\n            // Ensure we disconnect nodes to prevent zombie recordings\n            if (this.audioWorkletNode) {\n                this.audioWorkletNode.port.postMessage({\n                    command: 'deviceDisconnected',\n                })\n\n                try {\n                    this.source.disconnect(this.audioWorkletNode)\n                    this.audioWorkletNode.disconnect()\n                } catch (e) {\n                    // Ignore disconnection errors as the track might already be gone\n                }\n            }\n        }\n\n        // Add listeners to all audio tracks\n        const tracks = this.mediaStream.getAudioTracks()\n        tracks.forEach((track) => {\n            track.addEventListener('ended', handleTrackEnded)\n        })\n\n        // Store the handler for cleanup\n        this.deviceDisconnectionHandler = () => {\n            tracks.forEach((track) => {\n                track.removeEventListener('ended', handleTrackEnded)\n            })\n        }\n    }\n\n    /**\n     * Explicitly set the position for continuous recording across device switches\n     * @param position The position in seconds to continue from\n     */\n    setPosition(position: number): void {\n        if (position >= 0) {\n            this.position = position\n            this.logger?.debug(`Position explicitly set to ${position} seconds`)\n        } else {\n            this.logger?.warn(`Invalid position value: ${position}, ignoring`)\n        }\n    }\n\n    /**\n     * Get the current position in seconds\n     * @returns The current position\n     */\n    getPosition(): number {\n        return this.position\n    }\n\n    /**\n     * Gets the current compressed chunks\n     * @returns Array of current compressed audio chunks\n     */\n    getCompressedChunks(): Blob[] {\n        return [...this.compressedChunks]\n    }\n\n    /**\n     * Sets the compressed chunks from a previous recorder\n     * @param chunks Array of compressed chunks from a previous recorder\n     */\n    setCompressedChunks(chunks: Blob[]): void {\n        if (chunks && chunks.length > 0) {\n            this.logger?.debug(\n                `Adding ${chunks.length} compressed chunks from previous device`\n            )\n            this.compressedChunks = [...chunks, ...this.compressedChunks]\n            // Update size\n            this.compressedSize = this.compressedChunks.reduce(\n                (size, chunk) => size + chunk.size,\n                0\n            )\n        }\n    }\n}\n"]}

package/build/esm/constants.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"constants.js","sourceRoot":"","sources":["../../src/constants.ts"],"names":[],"mappings":"AAAA,8CAA8C;AAC9C,OAAO,EAAE,QAAQ,EAAE,MAAM,cAAc,CAAA;AAIvC,MAAM,CAAC,MAAM,KAAK,GAAG,QAAQ,CAAC,EAAE,KAAK,KAAK,CAAA;AAC1C,MAAM,CAAC,MAAM,eAAe,GAAG,mBAAmB,CAAA;AAElD,iDAAiD;AACjD,MAAM,CAAC,MAAM,WAAW,GAAG,UAAU,CAAA,CAAC,SAAS;AAC/C,MAAM,CAAC,MAAM,WAAW,GAAG,UAAU,CAAA,CAAC,SAAS;AAC/C,MAAM,CAAC,MAAM,YAAY,GAAG,UAAU,CAAA,CAAC,SAAS;AAChD,MAAM,CAAC,MAAM,aAAa,GAAG,UAAU,CAAA,CAAC,SAAS;AACjD,MAAM,CAAC,MAAM,aAAa,GAAG,UAAU,CAAA,CAAC,SAAS;AAEjD,iBAAiB;AACjB,MAAM,CAAC,MAAM,mBAAmB,GAAe,KAAK,CAAA;AACpD,MAAM,CAAC,MAAM,iBAAiB,GAAa,EAAE,CAAA","sourcesContent":["// packages/expo-audio-stream/src/constants.ts\nimport { Platform } from 'react-native'\n\nimport { BitDepth, SampleRate } from './ExpoAudioStream.types'\n\nexport const isWeb = Platform.OS === 'web'\nexport const DEBUG_NAMESPACE = 'expo-audio-stream'\n\n// Constants for identifying chunks in a WAV file\nexport const RIFF_HEADER = 0x52494646 // \"RIFF\"\nexport const WAVE_HEADER = 0x57415645 // \"WAVE\"\nexport const FMT_CHUNK_ID = 0x666d7420 // \"fmt \"\nexport const DATA_CHUNK_ID = 0x64617461 // \"data\"\nexport const INFO_CHUNK_ID = 0x494e464f // \"INFO\"\n\n// Default values\nexport const DEFAULT_SAMPLE_RATE: SampleRate = 16000\nexport const DEFAULT_BIT_DEPTH: BitDepth = 32\n"]}

package/build/esm/events.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"events.js","sourceRoot":"","sources":["../../src/events.ts"],"names":[],"mappings":"AAAA,2CAA2C;AAE3C,OAAO,EAAE,kBAAkB,EAA0B,MAAM,mBAAmB,CAAA;AAI9E,OAAO,qBAAqB,MAAM,yBAAyB,CAAA;AAE3D,MAAM,OAAO,GAAG,IAAI,kBAAkB,CAAC,qBAAqB,CAAC,CAAA;AAqB7D,MAAM,UAAU,qBAAqB,CACjC,QAAqD;IAErD,OAAO,OAAO,CAAC,WAAW,CAAoB,WAAW,EAAE,QAAQ,CAAC,CAAA;AACxE,CAAC;AAKD,MAAM,UAAU,wBAAwB,CACpC,QAAsD;IAEtD,OAAO,OAAO,CAAC,WAAW,CAAqB,eAAe,EAAE,QAAQ,CAAC,CAAA;AAC7E,CAAC;AAED,MAAM,UAAU,gCAAgC,CAC5C,QAAqD;IAErD,oBAAoB;IACpB,OAAO,CAAC,KAAK,CAAC,wCAAwC,CAAC,CAAA;IAEvD,MAAM,YAAY,GAAG,OAAO,CAAC,WAAW,CACpC,wBAAwB,EAAE,+CAA+C;IACzE,CAAC,KAAK,EAAE,EAAE;QACN,OAAO,CAAC,KAAK,CAAC,wCAAwC,EAAE,KAAK,CAAC,CAAA;QAC9D,QAAQ,CAAC,KAAK,CAAC,CAAA;IACnB,CAAC,CACJ,CAAA;IAED,OAAO,YAAY,CAAA;AACvB,CAAC","sourcesContent":["// packages/expo-audio-stream/src/events.ts\n\nimport { LegacyEventEmitter, type EventSubscription } from 'expo-modules-core'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { RecordingInterruptionEvent } from './ExpoAudioStream.types'\nimport ExpoAudioStreamModule from './ExpoAudioStreamModule'\n\nconst emitter = new LegacyEventEmitter(ExpoAudioStreamModule)\n\n// Internal event payload from native module\nexport interface AudioEventPayload {\n    encoded?: string\n    buffer?: Float32Array\n    fileUri: string\n    lastEmittedSize: number\n    position: number\n    deltaSize: number\n    totalSize: number\n    mimeType: string\n    streamUuid: string\n    compression?: {\n        data?: string | Blob // Base64 (native) or Float32Array (web) encoded compressed data chunk\n        position: number\n        eventDataSize: number\n        totalSize: number\n    }\n}\n\nexport function addAudioEventListener(\n    listener: (event: AudioEventPayload) => Promise<void>\n): EventSubscription {\n    return emitter.addListener<AudioEventPayload>('AudioData', listener)\n}\n\n// Only aliasing the AudioAnalysis type for the event payload\nexport interface AudioAnalysisEvent extends AudioAnalysis {}\n\nexport function addAudioAnalysisListener(\n    listener: (event: AudioAnalysisEvent) => Promise<void>\n): EventSubscription {\n    return emitter.addListener<AudioAnalysisEvent>('AudioAnalysis', listener)\n}\n\nexport function addRecordingInterruptionListener(\n    listener: (event: RecordingInterruptionEvent) => void\n): EventSubscription {\n    // Add debug logging\n    console.debug('Adding recording interruption listener')\n\n    const subscription = emitter.addListener<RecordingInterruptionEvent>(\n        'onRecordingInterrupted', // Make sure this matches the native event name\n        (event) => {\n            console.debug('Recording interruption event received:', event)\n            listener(event)\n        }\n    )\n\n    return subscription\n}\n"]}

package/build/esm/hooks/useAudioDevices.js.map

@@ -0,0 +1 @@
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{ useCallback, useEffect, useState } from 'react'\n\nimport { audioDeviceManager } from '../AudioDeviceManager'\nimport { AudioDevice } from '../ExpoAudioStream.types'\n\n/**\n * React hook for managing audio input devices\n */\nexport function useAudioDevices() {\n    const [devices, setDevices] = useState<AudioDevice[]>([])\n    const [currentDevice, setCurrentDevice] = useState<AudioDevice | null>(null)\n    const [loading, setLoading] = useState(true)\n    const [error, setError] = useState<Error | null>(null)\n\n    // Load devices on mount\n    useEffect(() => {\n        let isMounted = true\n\n        const loadDevices = async () => {\n            try {\n                setLoading(true)\n                setError(null)\n\n                // Load available devices\n                const availableDevices =\n                    await audioDeviceManager.getAvailableDevices()\n                if (isMounted) setDevices(availableDevices)\n\n                // Get current device\n                const device = await audioDeviceManager.getCurrentDevice()\n                if (isMounted) setCurrentDevice(device)\n            } catch (err) {\n                console.error('Failed to load audio devices:', err)\n                if (isMounted)\n                    setError(\n                        err instanceof Error\n                            ? err\n                            : new Error('Failed to load audio devices')\n                    )\n            } finally {\n                if (isMounted) setLoading(false)\n            }\n        }\n\n        loadDevices()\n\n        // Set up device change listener\n        const removeListener = audioDeviceManager.addDeviceChangeListener(\n            (updatedDevices: AudioDevice[]) => {\n                if (isMounted) {\n                    setDevices(updatedDevices)\n\n                    // If our current device is no longer available, update it\n                    if (\n                        currentDevice &&\n                        !updatedDevices.some(\n                            (d: AudioDevice) => d.id === currentDevice.id\n                        )\n                    ) {\n                        audioDeviceManager\n                            .getCurrentDevice()\n                            .then((newDevice: AudioDevice | null) => {\n                                if (isMounted) setCurrentDevice(newDevice)\n                            })\n                    }\n                }\n            }\n        )\n\n        return () => {\n            isMounted = false\n            removeListener()\n        }\n    }, [])\n\n    /**\n     * Select a specific audio input device\n     * @param deviceId The ID of the device to select\n     * @returns Promise resolving to a boolean indicating success\n     */\n    const selectDevice = useCallback(\n        async (deviceId: string): Promise<boolean> => {\n            try {\n                setLoading(true)\n                setError(null)\n\n                const success = await audioDeviceManager.selectDevice(deviceId)\n\n                if (success) {\n                    // Get the updated current device after selection\n                    const device = await audioDeviceManager.getCurrentDevice()\n                    setCurrentDevice(device)\n                }\n\n                return success\n            } catch (err) {\n                console.error('Failed to select audio device:', err)\n                setError(\n                    err instanceof Error\n                        ? err\n                        : new Error('Failed to select audio device')\n                )\n                return false\n            } finally {\n                setLoading(false)\n            }\n        },\n        []\n    )\n\n    /**\n     * Reset to the default audio input device\n     * @returns Promise resolving to a boolean indicating success\n     */\n    const resetToDefaultDevice = useCallback(async (): Promise<boolean> => {\n        try {\n            setLoading(true)\n            setError(null)\n\n            const success = await audioDeviceManager.resetToDefaultDevice()\n\n            if (success) {\n                // Get the updated current device after reset\n                const device = await audioDeviceManager.getCurrentDevice()\n                setCurrentDevice(device)\n            }\n\n            return success\n        } catch (err) {\n            console.error('Failed to reset to default audio device:', err)\n            setError(\n                err instanceof Error\n                    ? err\n                    : new Error('Failed to reset to default audio device')\n            )\n            return false\n        } finally {\n            setLoading(false)\n        }\n    }, [])\n\n    /**\n     * Refresh the list of available devices\n     */\n    const refreshDevices = useCallback(async (): Promise<AudioDevice[]> => {\n        try {\n            setLoading(true)\n            setError(null)\n\n            const updatedDevices = await audioDeviceManager.refreshDevices()\n            setDevices(updatedDevices)\n\n            // Also refresh the current device\n            const device = await audioDeviceManager.getCurrentDevice()\n            setCurrentDevice(device)\n\n            return updatedDevices\n        } catch (err) {\n            console.error('Failed to refresh audio devices:', err)\n            setError(\n                err instanceof Error\n                    ? err\n                    : new Error('Failed to refresh audio devices')\n            )\n            return []\n        } finally {\n            setLoading(false)\n        }\n    }, [])\n\n    return {\n        devices,\n        currentDevice,\n        loading,\n        error,\n        selectDevice,\n        resetToDefaultDevice,\n        refreshDevices,\n    }\n}\n"]}

package/build/esm/index.js.map

@@ -0,0 +1 @@
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package/build/esm/trimAudio.js.map

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package/build/esm/useAudioRecorder.js.map

@@ -0,0 +1 @@
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src/useAudioRecorder.ts\nimport { EventSubscription, Platform } from 'expo-modules-core'\nimport { useCallback, useEffect, useReducer, useRef } from 'react'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport {\n    AudioDataEvent,\n    AudioRecording,\n    AudioStreamStatus,\n    CompressionInfo,\n    ConsoleLike,\n    RecordingConfig,\n    StartRecordingResult,\n} from './ExpoAudioStream.types'\nimport ExpoAudioStreamModule from './ExpoAudioStreamModule'\nimport {\n    addAudioAnalysisListener,\n    addAudioEventListener,\n    AudioEventPayload,\n    addRecordingInterruptionListener,\n} from './events'\n\nexport interface UseAudioRecorderProps {\n    logger?: ConsoleLike\n    audioWorkletUrl?: string\n    featuresExtratorUrl?: string\n}\n\nexport interface UseAudioRecorderState {\n    prepareRecording: (_: RecordingConfig) => Promise<void>\n    startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n    stopRecording: () => Promise<AudioRecording>\n    pauseRecording: () => Promise<void>\n    resumeRecording: () => Promise<void>\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number\n    size: number\n    compression?: CompressionInfo\n    analysisData?: AudioAnalysis\n}\n\ninterface RecorderReducerState {\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number\n    size: number\n    compression?: CompressionInfo\n    analysisData?: AudioAnalysis\n}\n\ntype RecorderAction =\n    | { type: 'START' | 'STOP' | 'PAUSE' | 'RESUME' }\n    | {\n          type: 'UPDATE_RECORDING_STATE'\n          payload: {\n              isRecording: boolean\n              isPaused: boolean\n          }\n      }\n    | {\n          type: 'UPDATE_STATUS'\n          payload: {\n              durationMs: number\n              size: number\n              compression?: CompressionInfo\n          }\n      }\n    | { type: 'UPDATE_ANALYSIS'; payload: AudioAnalysis }\n\nconst defaultAnalysis: AudioAnalysis = {\n    segmentDurationMs: 100,\n    bitDepth: 32,\n    numberOfChannels: 1,\n    durationMs: 0,\n    sampleRate: 44100,\n    samples: 0,\n    dataPoints: [],\n    rmsRange: {\n        min: Number.POSITIVE_INFINITY,\n        max: Number.NEGATIVE_INFINITY,\n    },\n    amplitudeRange: {\n        min: Number.POSITIVE_INFINITY,\n        max: Number.NEGATIVE_INFINITY,\n    },\n}\n\nfunction audioRecorderReducer(\n    state: RecorderReducerState,\n    action: RecorderAction\n): RecorderReducerState {\n    switch (action.type) {\n        case 'START':\n            return {\n                ...state,\n                isRecording: true,\n                isPaused: false,\n                durationMs: 0,\n                size: 0,\n                compression: undefined,\n                analysisData: defaultAnalysis,\n            }\n        case 'STOP':\n            return {\n                ...state,\n                isRecording: false,\n                isPaused: false,\n                durationMs: 0,\n                size: 0,\n                compression: undefined,\n                analysisData: undefined,\n            }\n        case 'PAUSE':\n            return { ...state, isPaused: true, isRecording: false }\n        case 'RESUME':\n            return { ...state, isPaused: false, isRecording: true }\n        case 'UPDATE_RECORDING_STATE':\n            return {\n                ...state,\n                isPaused: action.payload.isPaused,\n                isRecording: action.payload.isRecording,\n            }\n        case 'UPDATE_STATUS': {\n            const newState = {\n                ...state,\n                durationMs: action.payload.durationMs,\n                size: action.payload.size,\n                compression: action.payload.compression\n                    ? {\n                          size: action.payload.compression.size,\n                          mimeType: action.payload.compression.mimeType,\n                          bitrate: action.payload.compression.bitrate,\n                          format: action.payload.compression.format,\n                      }\n                    : undefined,\n            }\n            return newState\n        }\n        case 'UPDATE_ANALYSIS':\n            return {\n                ...state,\n                analysisData: action.payload,\n            }\n        default:\n            return state\n    }\n}\n\ninterface HandleAudioAnalysisProps {\n    analysis: AudioAnalysis\n    visualizationDuration: number\n}\n\nexport function useAudioRecorder({\n    logger,\n    audioWorkletUrl,\n    featuresExtratorUrl,\n}: UseAudioRecorderProps = {}): UseAudioRecorderState {\n    const [state, dispatch] = useReducer(audioRecorderReducer, {\n        isRecording: false,\n        isPaused: false,\n        durationMs: 0,\n        size: 0,\n        compression: undefined,\n        analysisData: undefined,\n    })\n\n    const startResultRef = useRef<StartRecordingResult | null>(null)\n\n    const analysisListenerRef = useRef<EventSubscription | null>(null)\n    // analysisRef is the current analysis data (last 10 seconds by default)\n    const analysisRef = useRef<AudioAnalysis>({ ...defaultAnalysis })\n    // fullAnalysisRef is the full analysis data (all data points)\n    const fullAnalysisRef = useRef<AudioAnalysis>({\n        ...defaultAnalysis,\n    })\n\n    // Instantiate the module for web with URLs\n    const ExpoAudioStream =\n        Platform.OS === 'web'\n            ? ExpoAudioStreamModule({\n                  audioWorkletUrl,\n                  featuresExtratorUrl,\n                  logger,\n              })\n            : ExpoAudioStreamModule\n\n    const onAudioStreamRef = useRef<\n        ((_: AudioDataEvent) => Promise<void>) | null\n    >(null)\n\n    const stateRef = useRef({\n        isRecording: false,\n        isPaused: false,\n        durationMs: 0,\n        size: 0,\n        compression: undefined as CompressionInfo | undefined,\n    })\n\n    const recordingConfigRef = useRef<RecordingConfig | null>(null)\n\n    const handleAudioAnalysis = useCallback(\n        async ({\n            analysis,\n            visualizationDuration,\n        }: HandleAudioAnalysisProps) => {\n            const savedAnalysisData = analysisRef.current || {\n                ...defaultAnalysis,\n            }\n\n            const maxDuration = visualizationDuration\n\n            logger?.debug(\n                `[handleAudioAnalysis] Received audio analysis: maxDuration=${maxDuration} analysis.dataPoints=${analysis.dataPoints.length} analysisData.dataPoints=${savedAnalysisData.dataPoints.length}`\n            )\n\n            // Combine data points\n            const combinedDataPoints = [\n                ...savedAnalysisData.dataPoints,\n                ...analysis.dataPoints,\n            ]\n\n            const fullCombinedDataPoints = [\n                ...(fullAnalysisRef.current?.dataPoints ?? []),\n                ...analysis.dataPoints,\n            ]\n\n            // Calculate the new duration\n            // The number of segments is based on how many segments of segmentDurationMs can fit in visualizationDuration\n            const numberOfSegments = Math.ceil(\n                visualizationDuration / analysis.segmentDurationMs\n            )\n            // maxDataPoints should be the number of data points, not milliseconds\n            const maxDataPoints = numberOfSegments\n\n            logger?.debug(\n                `[handleAudioAnalysis] Combined data points before trimming: numberOfSegments=${numberOfSegments} visualizationDuration=${visualizationDuration} combinedDataPointsLength=${combinedDataPoints.length} vs maxDataPoints=${maxDataPoints}`\n            )\n\n            // Trim data points to keep within the maximum number of data points\n            if (combinedDataPoints.length > maxDataPoints) {\n                combinedDataPoints.splice(\n                    0,\n                    combinedDataPoints.length - maxDataPoints\n                )\n            }\n\n            // Keep the full data points\n            fullAnalysisRef.current = {\n                ...fullAnalysisRef.current,\n                dataPoints: fullCombinedDataPoints,\n            }\n            fullAnalysisRef.current.durationMs =\n                fullCombinedDataPoints.length * analysis.segmentDurationMs\n            savedAnalysisData.dataPoints = combinedDataPoints\n            savedAnalysisData.bitDepth =\n                analysis.bitDepth || savedAnalysisData.bitDepth\n            savedAnalysisData.durationMs =\n                combinedDataPoints.length * analysis.segmentDurationMs\n\n            // Update amplitude range\n            const newMin = Math.min(\n                savedAnalysisData.amplitudeRange.min,\n                analysis.amplitudeRange.min\n            )\n            const newMax = Math.max(\n                savedAnalysisData.amplitudeRange.max,\n                analysis.amplitudeRange.max\n            )\n\n            savedAnalysisData.amplitudeRange = {\n                min: newMin,\n                max: newMax,\n            }\n            fullAnalysisRef.current.amplitudeRange = {\n                min: newMin,\n                max: newMax,\n            }\n\n            logger?.debug(\n                `[handleAudioAnalysis] Updated analysis data: durationMs=${savedAnalysisData.durationMs}`,\n                { dataPoints: savedAnalysisData.dataPoints.length }\n            )\n\n            // Call the onAudioAnalysis callback if it exists in the recording config\n            if (recordingConfigRef.current?.onAudioAnalysis) {\n                recordingConfigRef.current\n                    .onAudioAnalysis(analysis)\n                    .catch((error) => {\n                        logger?.warn(`Error processing audio analysis:`, error)\n                    })\n            }\n\n            // Update the ref\n            analysisRef.current = savedAnalysisData\n\n            // Dispatch the updated analysis data to state to trigger re-render\n            dispatch({\n                type: 'UPDATE_ANALYSIS',\n                payload: { ...savedAnalysisData },\n            })\n        },\n        [dispatch]\n    )\n\n    const handleAudioEvent = useCallback(\n        async (eventData: AudioEventPayload) => {\n            const {\n                fileUri,\n                deltaSize,\n                totalSize,\n                lastEmittedSize,\n                position,\n                streamUuid,\n                encoded,\n                mimeType,\n                buffer,\n                compression,\n            } = eventData\n            logger?.debug(`[handleAudioEvent] Received audio event:`, {\n                fileUri,\n                deltaSize,\n                totalSize,\n                position,\n                mimeType,\n                lastEmittedSize,\n                streamUuid,\n                encodedLength: encoded?.length,\n                compression,\n            })\n            if (deltaSize === 0) {\n                // Ignore packet with no data\n                return\n            }\n            try {\n                // Coming from native ( ios / android ) otherwise buffer is set\n                if (Platform.OS !== 'web') {\n                    // Read the audio file as a base64 string for comparison\n                    if (!encoded) {\n                        logger?.error(`Encoded audio data is missing`)\n                        throw new Error('Encoded audio data is missing')\n                    }\n                    onAudioStreamRef.current?.({\n                        data: encoded,\n                        position,\n                        fileUri,\n                        eventDataSize: deltaSize,\n                        totalSize,\n                        compression:\n                            compression && startResultRef.current?.compression\n                                ? {\n                                      data: compression.data,\n                                      size: compression.totalSize,\n                                      mimeType:\n                                          startResultRef.current.compression\n                                              ?.mimeType,\n                                      bitrate:\n                                          startResultRef.current.compression\n                                              ?.bitrate,\n                                      format: startResultRef.current.compression\n                                          ?.format,\n                                  }\n                                : undefined,\n                    })\n                } else if (buffer) {\n                    // Coming from web\n                    const webEvent: AudioDataEvent = {\n                        data: buffer,\n                        position,\n                        fileUri,\n                        eventDataSize: deltaSize,\n                        totalSize,\n                        compression:\n                            compression && startResultRef.current?.compression\n                                ? {\n                                      data: compression.data,\n                                      size: compression.totalSize,\n                                      mimeType:\n                                          startResultRef.current.compression\n                                              ?.mimeType,\n                                      bitrate:\n                                          startResultRef.current.compression\n                                              ?.bitrate,\n                                      format: startResultRef.current.compression\n                                          ?.format,\n                                  }\n                                : undefined,\n                    }\n                    onAudioStreamRef.current?.(webEvent)\n                    logger?.debug(\n                        `[handleAudioEvent] Audio data sent to onAudioStream`,\n                        webEvent\n                    )\n                }\n            } catch (error) {\n                logger?.error(`Error processing audio event:`, error)\n            }\n        },\n        []\n    )\n\n    const checkStatus = useCallback(async () => {\n        try {\n            const status: AudioStreamStatus = ExpoAudioStream.status()\n            logger?.debug(\n                `Status: paused: ${status.isPaused} isRecording: ${status.isRecording} durationMs: ${status.durationMs} size: ${status.size}`,\n                status.compression\n            )\n\n            // Only dispatch if values actually changed\n            if (\n                status.isRecording !== stateRef.current.isRecording ||\n                status.isPaused !== stateRef.current.isPaused\n            ) {\n                stateRef.current.isRecording = status.isRecording\n                stateRef.current.isPaused = status.isPaused\n                dispatch({\n                    type: 'UPDATE_RECORDING_STATE',\n                    payload: {\n                        isRecording: status.isRecording,\n                        isPaused: status.isPaused,\n                    },\n                })\n            }\n\n            if (\n                status.durationMs !== stateRef.current.durationMs ||\n                status.size !== stateRef.current.size\n            ) {\n                stateRef.current.durationMs = status.durationMs\n                stateRef.current.size = status.size\n                stateRef.current.compression = status.compression\n                dispatch({\n                    type: 'UPDATE_STATUS',\n                    payload: {\n                        durationMs: status.durationMs,\n                        size: status.size,\n                        compression: status.compression,\n                    },\n                })\n            }\n        } catch (error) {\n            logger?.error(`Error getting status:`, error)\n        }\n    }, [ExpoAudioStream, logger]) // Only depend on ExpoAudioStream and logger\n\n    // Update ref when state changes\n    useEffect(() => {\n        stateRef.current = {\n            isRecording: state.isRecording,\n            isPaused: state.isPaused,\n            durationMs: state.durationMs,\n            size: state.size,\n            compression: state.compression,\n        }\n    }, [\n        state.isRecording,\n        state.isPaused,\n        state.durationMs,\n        state.size,\n        state.compression,\n    ])\n\n    const startRecording = useCallback(\n        async (recordingOptions: RecordingConfig) => {\n            recordingConfigRef.current = recordingOptions\n            logger?.debug(`start recoding`, recordingOptions)\n\n            analysisRef.current = { ...defaultAnalysis } // Reset analysis data\n            fullAnalysisRef.current = { ...defaultAnalysis }\n            const { onAudioStream, ...options } = recordingOptions\n            const { enableProcessing } = options\n\n            const maxRecentDataDuration = 10000 // TODO compute maxRecentDataDuration based on screen dimensions\n            if (typeof onAudioStream === 'function') {\n                onAudioStreamRef.current = onAudioStream\n            } else {\n                logger?.warn(`onAudioStream is not a function`, onAudioStream)\n                onAudioStreamRef.current = null\n            }\n            const startResult: StartRecordingResult =\n                await ExpoAudioStream.startRecording(options)\n            dispatch({ type: 'START' })\n\n            startResultRef.current = startResult\n\n            if (enableProcessing) {\n                logger?.debug(`Enabling audio analysis listener`)\n                const listener = addAudioAnalysisListener(\n                    async (analysisData) => {\n                        try {\n                            await handleAudioAnalysis({\n                                analysis: analysisData,\n                                visualizationDuration: maxRecentDataDuration,\n                            })\n                        } catch (error) {\n                            logger?.warn(\n                                `Error processing audio analysis:`,\n                                error\n                            )\n                        }\n                    }\n                )\n\n                analysisListenerRef.current = listener\n            }\n\n            return startResult\n        },\n        [handleAudioAnalysis, dispatch]\n    )\n\n    const prepareRecording = useCallback(\n        async (recordingOptions: RecordingConfig) => {\n            recordingConfigRef.current = recordingOptions\n            logger?.debug(`preparing recording`, recordingOptions)\n\n            analysisRef.current = { ...defaultAnalysis } // Reset analysis data\n            fullAnalysisRef.current = { ...defaultAnalysis }\n            const { onAudioStream, ...options } = recordingOptions\n\n            // Store onAudioStream for later use when recording starts\n            if (typeof onAudioStream === 'function') {\n                onAudioStreamRef.current = onAudioStream\n            } else {\n                logger?.warn(`onAudioStream is not a function`, onAudioStream)\n                onAudioStreamRef.current = null\n            }\n\n            // Call the native prepareRecording method\n            await ExpoAudioStream.prepareRecording(options)\n            logger?.debug(`recording prepared successfully`)\n        },\n        []\n    )\n\n    const stopRecording = useCallback(async () => {\n        logger?.debug(`stoping recording`)\n\n        const stopResult: AudioRecording = await ExpoAudioStream.stopRecording()\n        stopResult.analysisData = fullAnalysisRef.current\n\n        if (analysisListenerRef.current) {\n            analysisListenerRef.current.remove()\n            analysisListenerRef.current = null\n        }\n        onAudioStreamRef.current = null\n        logger?.debug(`recording stopped`, stopResult)\n        dispatch({ type: 'STOP' })\n        return stopResult\n    }, [dispatch])\n\n    const pauseRecording = useCallback(async () => {\n        logger?.debug(`pause recording`)\n        const pauseResult = await ExpoAudioStream.pauseRecording()\n        dispatch({ type: 'PAUSE' })\n        return pauseResult\n    }, [dispatch])\n\n    const resumeRecording = useCallback(async () => {\n        logger?.debug(`resume recording`)\n        const resumeResult = await ExpoAudioStream.resumeRecording()\n        dispatch({ type: 'RESUME' })\n        return resumeResult\n    }, [dispatch])\n\n    useEffect(() => {\n        let intervalId: NodeJS.Timeout | undefined\n\n        if (state.isRecording || state.isPaused) {\n            // Immediately check status when starting\n            checkStatus()\n\n            // Start interval\n            intervalId = setInterval(checkStatus, 1000)\n        }\n\n        return () => {\n            if (intervalId) {\n                clearInterval(intervalId)\n                intervalId = undefined\n            }\n        }\n    }, [checkStatus, state.isRecording, state.isPaused])\n\n    useEffect(() => {\n        logger?.debug(`Registering audio event listener`)\n        const subscribeAudio = addAudioEventListener(handleAudioEvent)\n\n        logger?.debug(\n            `Subscribed to audio event listener and analysis listener`,\n            {\n                subscribeAudio,\n            }\n        )\n\n        return () => {\n            logger?.debug(`Removing audio event listener`)\n            subscribeAudio.remove()\n        }\n    }, [handleAudioEvent, handleAudioAnalysis])\n\n    useEffect(() => {\n        // Add event subscription for recording interruptions\n        logger?.debug('Setting up recording interruption listener')\n\n        const subscription = addRecordingInterruptionListener((event) => {\n            logger?.debug('Received recording interruption event:', event)\n\n            // Check if we have a callback configured\n            if (recordingConfigRef.current?.onRecordingInterrupted) {\n                try {\n                    recordingConfigRef.current.onRecordingInterrupted(event)\n                } catch (error) {\n                    logger?.error(\n                        'Error in recording interruption callback:',\n                        error\n                    )\n                }\n            } else {\n                logger?.warn('No recording interruption callback configured')\n            }\n        })\n\n        return () => {\n            logger?.debug('Removing recording interruption listener')\n            subscription.remove()\n        }\n    }, []) // Empty dependency array since we want this to run once\n\n    return {\n        prepareRecording,\n        startRecording,\n        stopRecording,\n        pauseRecording,\n        resumeRecording,\n        isPaused: state.isPaused,\n        isRecording: state.isRecording,\n        durationMs: state.durationMs,\n        size: state.size,\n        compression: state.compression,\n        analysisData: state.analysisData,\n    }\n}\n"]}