@siteed/expo-audio-studio 2.5.0 → 2.6.0

package/build/WebRecorder.web.js.map

@@ -1 +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 { 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    }\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\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 !== '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 chunkSize = this.audioContext.sampleRate * 2 // Reduce to 2 seconds chunks\n                const sampleRate =\n                    event.data.sampleRate ?? this.audioContext.sampleRate\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                    // 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            this.logger?.debug(\n                `WebRecorder initialized -- recordSampleRate=${this.audioContext.sampleRate}, startPosition=${this.position}`,\n                this.config\n            )\n            this.audioWorkletNode.port.postMessage({\n                command: 'init',\n                recordSampleRate: this.audioContext.sampleRate,\n                exportSampleRate:\n                    this.config.sampleRate ?? this.audioContext.sampleRate,\n                bitDepth: this.bitDepth,\n                exportBitDepth: this.exportBitDepth,\n                channels: this.numberOfChannels,\n                interval: this.config.interval ?? DEFAULT_WEB_INTERVAL,\n                position: this.position, // Pass the current position to the processor\n                // enableLogging: !!this.logger,\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(`[${TAG}] Failed to initialize WebRecorder`, error)\n        }\n    }\n\n    /**\n     * Initializes the feature extractor worker for audio analysis\n     * Creates an inline worker from a blob for audio feature extraction\n     */\n    initFeatureExtractorWorker() {\n        try {\n            const blob = new Blob([InlineFeaturesExtractor], {\n                type: 'application/javascript',\n            })\n            const url = URL.createObjectURL(blob)\n            this.featureExtractorWorker = new Worker(url)\n            this.featureExtractorWorker.onmessage =\n                this.handleFeatureExtractorMessage.bind(this)\n            this.featureExtractorWorker.onerror = (error) => {\n                console.error(`[${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        } 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     * Stops the audio recording process and returns the recorded data\n     * @param externalAudioChunks Optional array of Float32Array chunks from previous devices\n     * @returns Promise resolving to an object containing PCM data and optional compressed blob\n     */\n    async stop(\n        externalAudioChunks?: Float32Array[]\n    ): Promise<{ pcmData: Float32Array; compressedBlob?: Blob }> {\n        try {\n            // Log what's happening for debugging\n            this.logger?.debug('Stopping recording and collecting final data')\n\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            // Return the compressed blob if available\n            return {\n                pcmData: new Float32Array(), // Return empty array since we're streaming\n                compressedBlob:\n                    this.compressedChunks.length > 0\n                        ? new Blob(this.compressedChunks, {\n                              type: 'audio/webm;codecs=opus',\n                          })\n                        : undefined,\n            }\n        } finally {\n            this.cleanup()\n            // Reset the chunks array\n            this.compressedChunks = []\n            this.compressedSize = 0\n            this.pendingCompressedChunk = null\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"]}
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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/utils/writeWavHeader.d.ts

@@ -10,6 +10,8 @@ export interface WavHeaderOptions {
     numChannels: number;
     /** The bit depth of the audio (e.g., 16, 24, or 32). */
     bitDepth: number;
+    /** Whether the audio data is in float format (only applies to 32-bit) */
+    isFloat?: boolean;
 }
 /**
  * Writes or updates a WAV (RIFF) header based on the provided options.
@@ -27,23 +29,6 @@ export interface WavHeaderOptions {
  * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
  *
  * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
- *
- * @example
- * // Create a standalone WAV header
- * const header = writeWavHeader({
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- *
- * @example
- * // Create a WAV header and combine it with audio data
- * const completeWav = writeWavHeader({
- *   buffer: audioData,
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
  */
-export declare const writeWavHeader: ({ buffer, sampleRate, numChannels, bitDepth, }: WavHeaderOptions) => ArrayBuffer;
+export declare const writeWavHeader: ({ buffer, sampleRate, numChannels, bitDepth, isFloat, }: WavHeaderOptions) => ArrayBuffer;
 //# sourceMappingURL=writeWavHeader.d.ts.map

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

@@ -1 +1 @@
-{"version":3,"file":"writeWavHeader.d.ts","sourceRoot":"","sources":["../../src/utils/writeWavHeader.ts"],"names":[],"mappings":"AAEA;;GAEG;AACH,MAAM,WAAW,gBAAgB;IAC7B,+FAA+F;IAC/F,MAAM,CAAC,EAAE,WAAW,CAAA;IACpB,wDAAwD;IACxD,UAAU,EAAE,MAAM,CAAA;IAClB,qEAAqE;IACrE,WAAW,EAAE,MAAM,CAAA;IACnB,wDAAwD;IACxD,QAAQ,EAAE,MAAM,CAAA;CACnB;AAED;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GAiCG;AACH,eAAO,MAAM,cAAc,mDAKxB,gBAAgB,KAAG,WA0DrB,CAAA"}
+{"version":3,"file":"writeWavHeader.d.ts","sourceRoot":"","sources":["../../src/utils/writeWavHeader.ts"],"names":[],"mappings":"AAEA;;GAEG;AACH,MAAM,WAAW,gBAAgB;IAC7B,+FAA+F;IAC/F,MAAM,CAAC,EAAE,WAAW,CAAA;IACpB,wDAAwD;IACxD,UAAU,EAAE,MAAM,CAAA;IAClB,qEAAqE;IACrE,WAAW,EAAE,MAAM,CAAA;IACnB,wDAAwD;IACxD,QAAQ,EAAE,MAAM,CAAA;IAChB,yEAAyE;IACzE,OAAO,CAAC,EAAE,OAAO,CAAA;CACpB;AAED;;;;;;;;;;;;;;;;GAgBG;AACH,eAAO,MAAM,cAAc,4DAMxB,gBAAgB,KAAG,WAyErB,CAAA"}

package/build/utils/writeWavHeader.js

@@ -15,25 +15,11 @@
  * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.
  *
  * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.
- *
- * @example
- * // Create a standalone WAV header
- * const header = writeWavHeader({
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
- *
- * @example
- * // Create a WAV header and combine it with audio data
- * const completeWav = writeWavHeader({
- *   buffer: audioData,
- *   sampleRate: 44100,
- *   numChannels: 2,
- *   bitDepth: 16
- * });
  */
-export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) => {
+export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, isFloat = bitDepth === 32, // Default to float for 32-bit
+ }) => {
+    // For 32-bit float, we use format 3, otherwise format 1 for PCM
+    const audioFormat = isFloat ? 3 : 1; // 3 = IEEE float, 1 = PCM
     const bytesPerSample = bitDepth / 8;
     const blockAlign = numChannels * bytesPerSample;
     const byteRate = sampleRate * blockAlign;
@@ -45,21 +31,26 @@ export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) =
     };
     // Function to write or update the header
     const writeHeader = (view, dataSize = 0xffffffff) => {
+        // RIFF chunk descriptor
         writeString(view, 0, 'RIFF'); // ChunkID
-        view.setUint32(4, 36 + dataSize, true); // ChunkSize
+        view.setUint32(4, 36 + dataSize, true); // ChunkSize: 4 + (8 + 16) + (8 + dataSize)
         writeString(view, 8, 'WAVE'); // Format
+        // "fmt " sub-chunk
         writeString(view, 12, 'fmt '); // Subchunk1ID
-        view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)
-        view.setUint16(20, bitDepth === 32 ? 3 : 1, true); // AudioFormat (3 for float, 1 for PCM)
+        view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM/Float)
+        view.setUint16(20, audioFormat, true); // AudioFormat (3 for float, 1 for PCM)
         view.setUint16(22, numChannels, true); // NumChannels
         view.setUint32(24, sampleRate, true); // SampleRate
-        view.setUint32(28, byteRate, true); // ByteRate
-        view.setUint16(32, blockAlign, true); // BlockAlign
+        view.setUint32(28, byteRate, true); // ByteRate = SampleRate * NumChannels * BitsPerSample/8
+        view.setUint16(32, blockAlign, true); // BlockAlign = NumChannels * BitsPerSample/8
         view.setUint16(34, bitDepth, true); // BitsPerSample
+        // "data" sub-chunk
         writeString(view, 36, 'data'); // Subchunk2ID
-        view.setUint32(40, dataSize, true); // Subchunk2Size
+        view.setUint32(40, dataSize, true); // Subchunk2Size = NumSamples * NumChannels * BitsPerSample/8
     };
     if (buffer) {
+        // Handle existing buffer
+        // Check for minimum size
         if (buffer.byteLength < 44) {
             throw new Error('Buffer is too small to contain a valid WAV header');
         }
@@ -72,16 +63,18 @@ export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) =
             return buffer;
         }
         else {
-            // Combine the new header with the existing buffer
+            // Create a new buffer with header + data
             const newBuffer = new ArrayBuffer(44 + buffer.byteLength);
             const newView = new DataView(newBuffer);
+            // Write header to new buffer
             writeHeader(newView, buffer.byteLength);
+            // Copy audio data after header
             new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44);
             return newBuffer;
         }
     }
     else {
-        // Create a standalone header
+        // Create standalone header
         const headerBuffer = new ArrayBuffer(44);
         const view = new DataView(headerBuffer);
         writeHeader(view);

package/build/utils/writeWavHeader.js.map

@@ -1 +1 @@
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If provided, it will be combined with the header. */\n    buffer?: ArrayBuffer\n    /** The sample rate of the audio in Hz (e.g., 44100). */\n    sampleRate: number\n    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */\n    numChannels: number\n    /** The bit depth of the audio (e.g., 16, 24, or 32). */\n    bitDepth: number\n}\n\n/**\n * Writes or updates a WAV (RIFF) header based on the provided options.\n *\n * This function can be used in three ways:\n * 1. To create a standalone WAV header (when no buffer is provided).\n * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).\n * 3. To update an existing WAV header in the provided buffer.\n *\n * For streaming audio where the final size is unknown, this function sets the size fields\n * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the\n * `updateWavHeaderSize` function once the final size is known.\n *\n * @param options - The options for creating or updating the WAV header.\n * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.\n *\n * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.\n *\n * @example\n * // Create a standalone WAV header\n * const header = writeWavHeader({\n *   sampleRate: 44100,\n *   numChannels: 2,\n *   bitDepth: 16\n * });\n *\n * @example\n * // Create a WAV header and combine it with audio data\n * const completeWav = writeWavHeader({\n *   buffer: audioData,\n *   sampleRate: 44100,\n *   numChannels: 2,\n *   bitDepth: 16\n * });\n */\nexport const writeWavHeader = ({\n    buffer,\n    sampleRate,\n    numChannels,\n    bitDepth,\n}: WavHeaderOptions): ArrayBuffer => {\n    const bytesPerSample = bitDepth / 8\n    const blockAlign = numChannels * bytesPerSample\n    const byteRate = sampleRate * blockAlign\n\n    // Function to write a string to the DataView\n    const writeString = (view: DataView, offset: number, string: string) => {\n        for (let i = 0; i < string.length; i++) {\n            view.setUint8(offset + i, string.charCodeAt(i))\n        }\n    }\n\n    // Function to write or update the header\n    const writeHeader = (view: DataView, dataSize: number = 0xffffffff) => {\n        writeString(view, 0, 'RIFF') // ChunkID\n        view.setUint32(4, 36 + dataSize, true) // ChunkSize\n        writeString(view, 8, 'WAVE') // Format\n        writeString(view, 12, 'fmt ') // Subchunk1ID\n        view.setUint32(16, 16, true) // Subchunk1Size (16 for PCM)\n        view.setUint16(20, bitDepth === 32 ? 3 : 1, true) // AudioFormat (3 for float, 1 for PCM)\n        view.setUint16(22, numChannels, true) // NumChannels\n        view.setUint32(24, sampleRate, true) // SampleRate\n        view.setUint32(28, byteRate, true) // ByteRate\n        view.setUint16(32, blockAlign, true) // BlockAlign\n        view.setUint16(34, bitDepth, true) // BitsPerSample\n        writeString(view, 36, 'data') // Subchunk2ID\n        view.setUint32(40, dataSize, true) // Subchunk2Size\n    }\n\n    if (buffer) {\n        if (buffer.byteLength < 44) {\n            throw new Error('Buffer is too small to contain a valid WAV header')\n        }\n\n        const view = new DataView(buffer)\n\n        // Check if the buffer already has a WAV header by looking for \"RIFF\" at the start\n        const existingHeader = view.getUint32(0, false) === 0x52494646 // \"RIFF\" in ASCII\n\n        if (existingHeader) {\n            // Update the existing header\n            writeHeader(view, buffer.byteLength - 44)\n            return buffer\n        } else {\n            // Combine the new header with the existing buffer\n            const newBuffer = new ArrayBuffer(44 + buffer.byteLength)\n            const newView = new DataView(newBuffer)\n            writeHeader(newView, buffer.byteLength)\n            new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44)\n            return newBuffer\n        }\n    } else {\n        // Create a standalone header\n        const headerBuffer = new ArrayBuffer(44)\n        const view = new DataView(headerBuffer)\n        writeHeader(view)\n        return headerBuffer\n    }\n}\n"]}
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If provided, it will be combined with the header. */\n    buffer?: ArrayBuffer\n    /** The sample rate of the audio in Hz (e.g., 44100). */\n    sampleRate: number\n    /** The number of audio channels (e.g., 1 for mono, 2 for stereo). */\n    numChannels: number\n    /** The bit depth of the audio (e.g., 16, 24, or 32). */\n    bitDepth: number\n    /** Whether the audio data is in float format (only applies to 32-bit) */\n    isFloat?: boolean\n}\n\n/**\n * Writes or updates a WAV (RIFF) header based on the provided options.\n *\n * This function can be used in three ways:\n * 1. To create a standalone WAV header (when no buffer is provided).\n * 2. To create a WAV header and combine it with existing audio data (when a buffer without a header is provided).\n * 3. To update an existing WAV header in the provided buffer.\n *\n * For streaming audio where the final size is unknown, this function sets the size fields\n * to the maximum 32-bit value (0xFFFFFFFF). These can be updated later using the\n * `updateWavHeaderSize` function once the final size is known.\n *\n * @param options - The options for creating or updating the WAV header.\n * @returns An ArrayBuffer containing the WAV header, or the header combined with the provided audio data.\n *\n * @throws {Error} Throws an error if the provided options are invalid or if the buffer is too small.\n */\nexport const writeWavHeader = ({\n    buffer,\n    sampleRate,\n    numChannels,\n    bitDepth,\n    isFloat = bitDepth === 32, // Default to float for 32-bit\n}: WavHeaderOptions): ArrayBuffer => {\n    // For 32-bit float, we use format 3, otherwise format 1 for PCM\n    const audioFormat = isFloat ? 3 : 1 // 3 = IEEE float, 1 = PCM\n\n    const bytesPerSample = bitDepth / 8\n    const blockAlign = numChannels * bytesPerSample\n    const byteRate = sampleRate * blockAlign\n\n    // Function to write a string to the DataView\n    const writeString = (view: DataView, offset: number, string: string) => {\n        for (let i = 0; i < string.length; i++) {\n            view.setUint8(offset + i, string.charCodeAt(i))\n        }\n    }\n\n    // Function to write or update the header\n    const writeHeader = (view: DataView, dataSize: number = 0xffffffff) => {\n        // RIFF chunk descriptor\n        writeString(view, 0, 'RIFF') // ChunkID\n        view.setUint32(4, 36 + dataSize, true) // ChunkSize: 4 + (8 + 16) + (8 + dataSize)\n        writeString(view, 8, 'WAVE') // Format\n\n        // \"fmt \" sub-chunk\n        writeString(view, 12, 'fmt ') // Subchunk1ID\n        view.setUint32(16, 16, true) // Subchunk1Size (16 for PCM/Float)\n        view.setUint16(20, audioFormat, true) // AudioFormat (3 for float, 1 for PCM)\n        view.setUint16(22, numChannels, true) // NumChannels\n        view.setUint32(24, sampleRate, true) // SampleRate\n        view.setUint32(28, byteRate, true) // ByteRate = SampleRate * NumChannels * BitsPerSample/8\n        view.setUint16(32, blockAlign, true) // BlockAlign = NumChannels * BitsPerSample/8\n        view.setUint16(34, bitDepth, true) // BitsPerSample\n\n        // \"data\" sub-chunk\n        writeString(view, 36, 'data') // Subchunk2ID\n        view.setUint32(40, dataSize, true) // Subchunk2Size = NumSamples * NumChannels * BitsPerSample/8\n    }\n\n    if (buffer) {\n        // Handle existing buffer\n\n        // Check for minimum size\n        if (buffer.byteLength < 44) {\n            throw new Error('Buffer is too small to contain a valid WAV header')\n        }\n\n        const view = new DataView(buffer)\n\n        // Check if the buffer already has a WAV header by looking for \"RIFF\" at the start\n        const existingHeader = view.getUint32(0, false) === 0x52494646 // \"RIFF\" in ASCII\n\n        if (existingHeader) {\n            // Update the existing header\n            writeHeader(view, buffer.byteLength - 44)\n            return buffer\n        } else {\n            // Create a new buffer with header + data\n            const newBuffer = new ArrayBuffer(44 + buffer.byteLength)\n            const newView = new DataView(newBuffer)\n\n            // Write header to new buffer\n            writeHeader(newView, buffer.byteLength)\n\n            // Copy audio data after header\n            new Uint8Array(newBuffer).set(new Uint8Array(buffer), 44)\n            return newBuffer\n        }\n    } else {\n        // Create standalone header\n        const headerBuffer = new ArrayBuffer(44)\n        const view = new DataView(headerBuffer)\n        writeHeader(view)\n        return headerBuffer\n    }\n}\n"]}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@siteed/expo-audio-studio",
-  "version": "2.5.0",
+  "version": "2.6.0",
   "description": "Comprehensive audio processing library for React Native and Expo with recording, analysis, visualization, and streaming capabilities across iOS, Android, and web",
   "license": "MIT",
   "main": "build/index.js",

package/src/AudioDeviceManager.ts

@@ -130,7 +130,7 @@ export class AudioDeviceManager {
 
     /**
      * Get all available audio input devices
-     * @param options Optional settings { refresh: boolean } to force refresh the device list
+     * @param options Optional settings to force refresh the device list. Can include a refresh flag.
      * @returns Promise resolving to an array of audio devices conforming to AudioDevice interface
      */
     async getAvailableDevices(options?: {

package/src/ExpoAudioStream.types.ts

@@ -207,6 +207,19 @@ export interface IOSConfig {
     audioSession?: AudioSessionConfig
 }
 
+/** Web platform specific configuration options */
+export interface WebConfig {
+    /**
+     * Whether to store uncompressed audio data for WAV generation
+     *
+     * When true, all PCM chunks are stored in memory to create a WAV file when compression is disabled
+     * When false, uncompressed audio won't be available, but memory usage will be lower
+     *
+     * Default: true (for backward compatibility)
+     */
+    storeUncompressedAudio?: boolean
+}
+
 // Add new type for interruption reasons
 export type RecordingInterruptionReason =
     /** Audio focus was lost to another app */
@@ -312,6 +325,9 @@ export interface RecordingConfig {
     /** iOS-specific configuration */
     ios?: IOSConfig
 
+    /** Web-specific configuration options */
+    web?: WebConfig
+
     /** Duration of each segment in milliseconds for analysis (default: 100) */
     segmentDurationMs?: number
 

package/src/ExpoAudioStream.web.ts

@@ -113,7 +113,56 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
     // Utility to handle user media stream
     async getMediaStream() {
         try {
-            return await navigator.mediaDevices.getUserMedia({ audio: true })
+            this.logger?.debug('Requesting user media (microphone)...')
+
+            // First check if the browser supports the necessary audio APIs
+            if (!navigator?.mediaDevices?.getUserMedia) {
+                this.logger?.error(
+                    'Browser does not support mediaDevices.getUserMedia'
+                )
+                throw new Error('Browser does not support audio recording')
+            }
+
+            // Get media with detailed audio constraints for better diagnostics
+            const constraints = {
+                audio: {
+                    echoCancellation: true,
+                    noiseSuppression: true,
+                    autoGainControl: true,
+                    // Add deviceId constraint if specified
+                    ...(this.recordingConfig?.deviceId
+                        ? {
+                              deviceId: {
+                                  exact: this.recordingConfig.deviceId,
+                              },
+                          }
+                        : {}),
+                },
+            }
+
+            this.logger?.debug('Media constraints:', constraints)
+
+            const stream =
+                await navigator.mediaDevices.getUserMedia(constraints)
+
+            // Get detailed info about the audio track for debugging
+            const audioTracks = stream.getAudioTracks()
+            if (audioTracks.length > 0) {
+                const track = audioTracks[0]
+                const settings = track.getSettings()
+                this.logger?.debug('Audio track obtained:', {
+                    label: track.label,
+                    id: track.id,
+                    enabled: track.enabled,
+                    muted: track.muted,
+                    readyState: track.readyState,
+                    settings,
+                })
+            } else {
+                this.logger?.warn('Stream has no audio tracks!')
+            }
+
+            return stream
         } catch (error) {
             this.logger?.error('Failed to get media stream:', error)
             throw error
@@ -400,7 +449,8 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
         this.logger?.debug('Starting stop process')
 
         try {
-            const { compressedBlob } = await this.customRecorder.stop()
+            const { compressedBlob, uncompressedBlob } =
+                await this.customRecorder.stop()
 
             this.isRecording = false
             this.isPaused = false
@@ -409,20 +459,52 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
             let fileUri = `${this.streamUuid}.${this.extension}`
             let mimeType = `audio/${this.extension}`
 
-            // Process compressed audio if available
-            if (compressedBlob && this.recordingConfig?.compression?.enabled) {
+            // Handle both compressed and uncompressed blobs according to configuration
+            const compressionEnabled =
+                this.recordingConfig?.compression?.enabled ?? false
+
+            // Process compressed blob if available
+            if (compressedBlob) {
                 const compressedUri = URL.createObjectURL(compressedBlob)
-                compression = {
+                const compressedInfo = {
                     compressedFileUri: compressedUri,
                     size: compressedBlob.size,
                     mimeType: 'audio/webm',
                     format: 'opus',
-                    bitrate: this.recordingConfig.compression.bitrate ?? 128000,
+                    bitrate:
+                        this.recordingConfig?.compression?.bitrate ?? 128000,
                 }
 
-                // Use compressed values when compression is enabled
-                fileUri = compressedUri
-                mimeType = 'audio/webm'
+                // If compression is enabled, use compressed blob as primary format
+                if (compressionEnabled) {
+                    this.logger?.debug(
+                        'Using compressed audio as primary output'
+                    )
+                    fileUri = compressedUri
+                    mimeType = 'audio/webm'
+
+                    // Store compression info
+                    compression = compressedInfo
+                } else {
+                    // Compression was enabled during recording but not set as primary
+                    // Store as alternate format
+                    compression = compressedInfo
+                }
+            }
+
+            // Process uncompressed WAV if available
+            if (uncompressedBlob) {
+                const wavUri = URL.createObjectURL(uncompressedBlob)
+
+                // If compression is disabled or no compressed blob is available,
+                // use WAV as primary format
+                if (!compressionEnabled || !compressedBlob) {
+                    this.logger?.debug(
+                        'Using uncompressed WAV as primary output'
+                    )
+                    fileUri = wavUri
+                    mimeType = 'audio/wav'
+                }
             }
 
             // Use the stored streamUuid for the final filename
@@ -443,6 +525,14 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
             // Reset after creating the result
             this.streamUuid = null
 
+            // Reset recording state variables to prepare for next recording
+            this.currentDurationMs = 0
+            this.currentSize = 0
+            this.lastEmittedSize = 0
+            this.totalCompressedSize = 0
+            this.lastEmittedCompressionSize = 0
+            this.audioChunks = []
+
             return result
         } catch (error) {
             this.logger?.error('Error stopping recording:', error)