@siteed/expo-audio-studio 2.9.0 → 2.10.0

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
Files changed (64) hide show
  1. package/CHANGELOG.md +9 -1
  2. package/android/build.gradle +9 -0
  3. package/android/src/androidTest/assets/chorus.wav +0 -0
  4. package/android/src/androidTest/assets/jfk.wav +0 -0
  5. package/android/src/androidTest/assets/osr_us_000_0010_8k.wav +0 -0
  6. package/android/src/androidTest/assets/recorder_hello_world.wav +0 -0
  7. package/android/src/androidTest/java/net/siteed/audiostream/AudioProcessorInstrumentedTest.kt +197 -0
  8. package/android/src/androidTest/java/net/siteed/audiostream/AudioRecorderInstrumentedTest.kt +541 -0
  9. package/android/src/androidTest/java/net/siteed/audiostream/integration/BufferDurationIntegrationTest.kt +324 -0
  10. package/android/src/androidTest/java/net/siteed/audiostream/integration/OutputControlIntegrationTest.kt +340 -0
  11. package/android/src/androidTest/java/net/siteed/audiostream/integration/README.md +95 -0
  12. package/android/src/androidTest/java/net/siteed/audiostream/integration/run_integration_tests.sh +28 -0
  13. package/android/src/main/java/net/siteed/audiostream/AudioFormatUtils.kt +264 -13
  14. package/android/src/main/java/net/siteed/audiostream/AudioProcessor.kt +3 -13
  15. package/android/src/main/java/net/siteed/audiostream/AudioRecorderManager.kt +118 -55
  16. package/android/src/main/java/net/siteed/audiostream/LogUtils.kt +32 -4
  17. package/android/src/main/java/net/siteed/audiostream/RecordingConfig.kt +50 -15
  18. package/android/src/test/java/net/siteed/audiostream/AudioFileHandlerTest.kt +279 -0
  19. package/android/src/test/java/net/siteed/audiostream/AudioFormatUtilsTest.kt +273 -0
  20. package/android/src/test/resources/chorus.wav +0 -0
  21. package/android/src/test/resources/generate_test_audio.py +94 -0
  22. package/android/src/test/resources/jfk.wav +0 -0
  23. package/android/src/test/resources/osr_us_000_0010_8k.wav +0 -0
  24. package/android/src/test/resources/recorder_hello_world.wav +0 -0
  25. package/build/cjs/ExpoAudioStream.types.js.map +1 -1
  26. package/build/cjs/ExpoAudioStream.web.js +37 -34
  27. package/build/cjs/ExpoAudioStream.web.js.map +1 -1
  28. package/build/cjs/WebRecorder.web.js +12 -10
  29. package/build/cjs/WebRecorder.web.js.map +1 -1
  30. package/build/esm/ExpoAudioStream.types.js.map +1 -1
  31. package/build/esm/ExpoAudioStream.web.js +37 -34
  32. package/build/esm/ExpoAudioStream.web.js.map +1 -1
  33. package/build/esm/WebRecorder.web.js +12 -10
  34. package/build/esm/WebRecorder.web.js.map +1 -1
  35. package/build/types/ExpoAudioStream.types.d.ts +54 -22
  36. package/build/types/ExpoAudioStream.types.d.ts.map +1 -1
  37. package/build/types/ExpoAudioStream.web.d.ts.map +1 -1
  38. package/build/types/WebRecorder.web.d.ts.map +1 -1
  39. package/ios/AudioNotificationManager.swift +2 -6
  40. package/ios/AudioStreamManager.swift +116 -50
  41. package/ios/ExpoAudioStream.podspec +6 -0
  42. package/ios/ExpoAudioStreamModule.swift +11 -8
  43. package/ios/ExpoAudioStudioTests/AudioFileHandlerTests.swift +338 -0
  44. package/ios/ExpoAudioStudioTests/AudioFormatUtilsTests.swift +331 -0
  45. package/ios/ExpoAudioStudioTests/AudioTestHelpers.swift +130 -0
  46. package/ios/ExpoAudioStudioTests/Info.plist +22 -0
  47. package/ios/ExpoAudioStudioTests/SimpleAudioTest.swift +98 -0
  48. package/ios/ExpoAudioStudioTests/TestAudioGenerator.swift +75 -0
  49. package/ios/RecordingSettings.swift +53 -22
  50. package/ios/tests/integration/buffer_duration_test.swift +185 -0
  51. package/ios/tests/integration/output_control_test.swift +322 -0
  52. package/ios/tests/integration/run_integration_tests.sh +27 -0
  53. package/ios/tests/standalone/audio_processing_test.swift +144 -0
  54. package/ios/tests/standalone/audio_recording_test.swift +277 -0
  55. package/ios/tests/standalone/audio_streaming_test.swift +249 -0
  56. package/ios/tests/standalone/standalone_test.swift +144 -0
  57. package/package.json +140 -133
  58. package/src/ExpoAudioStream.types.ts +66 -22
  59. package/src/ExpoAudioStream.web.ts +43 -38
  60. package/src/WebRecorder.web.ts +13 -10
  61. package/android/src/main/test/java/net/siteed/audiostream/AudioProcessorTest.kt +0 -56
  62. package/ios/siteedexpoaudiostudio.xcodeproj/project.xcworkspace/contents.xcworkspacedata +0 -7
  63. package/ios/siteedexpoaudiostudio.xcodeproj/project.xcworkspace/xcshareddata/IDEWorkspaceChecks.plist +0 -8
  64. /package/plugin/build/{index.d.ts → index.d.cts} +0 -0
@@ -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 { 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 readonly 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 readonly mediaStream: MediaStream | null = null\n private readonly 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] ${event.data.message}`)\n return\n }\n\n if (command !== 'newData') return\n\n const pcmBufferFloat = event.data.recordedData\n if (!pcmBufferFloat) {\n this.logger?.warn('Received empty audio buffer', event)\n return\n }\n\n // Process data in smaller chunks and emit immediately\n const sampleRate =\n event.data.sampleRate ?? this.audioContext.sampleRate\n // Use chunk size from config interval or default to 2 seconds\n const intervalMs = this.config.interval ?? DEFAULT_WEB_INTERVAL\n const chunkSize = Math.floor(sampleRate * (intervalMs / 1000))\n const duration = pcmBufferFloat.length / sampleRate\n\n // Use incoming position if provided by worklet, otherwise use our tracked position\n const incomingPosition =\n typeof event.data.position === 'number'\n ? event.data.position\n : this.position\n\n // Simple position tracking for logging (no duplicate filtering)\n this.logger?.debug(\n `Audio chunk: position=${incomingPosition.toFixed(3)}s, size=${pcmBufferFloat.length}`\n )\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 // 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 - this is for the final WAV file\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 // 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 // Prepare compression data if available\n const 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 ?? 128000,\n }\n : undefined\n\n // Emit chunk immediately - whether compressed or not\n this.emitAudioEventCallback({\n data: chunk,\n position: chunkPosition,\n compression,\n })\n\n // Reset pending compressed chunk after we've used it\n this.pendingCompressedChunk = null\n }\n\n // Update our position based on the worklet's position if provided\n this.position = incomingPosition + duration\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') return\n\n const segmentResult = event.data.result\n const uniqueNewDataPoints = this.filterUniqueDataPoints(\n segmentResult.dataPoints\n )\n\n // Update counter based on the highest ID seen\n this.updateDataPointCounter(uniqueNewDataPoints)\n\n // Update analysis data with the new results\n this.updateAudioAnalysisData(segmentResult, uniqueNewDataPoints)\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 * Filters out data points with duplicate IDs\n */\n private filterUniqueDataPoints(dataPoints: any[]): any[] {\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 uniquePoints = dataPoints.filter((dp) => !existingIds.has(dp.id))\n\n // Log filtered duplicates if any\n if (uniquePoints.length < dataPoints.length && this.logger?.warn) {\n this.logger.warn(\n `Filtered ${dataPoints.length - uniquePoints.length} duplicate datapoints`\n )\n }\n\n return uniquePoints\n }\n\n /**\n * Updates the counter based on the highest ID in datapoints\n */\n private updateDataPointCounter(dataPoints: any[]): void {\n if (dataPoints.length === 0) return\n\n const lastDataPoint = dataPoints[dataPoints.length - 1]\n if (lastDataPoint && typeof lastDataPoint.id === 'number') {\n const nextIdValue = lastDataPoint.id + 1\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 /**\n * Updates audio analysis data with segment results\n */\n private updateAudioAnalysisData(\n segmentResult: AudioAnalysis,\n uniqueDataPoints: any[]\n ): void {\n // Add unique data points to our analysis data\n this.audioAnalysisData.dataPoints.push(...uniqueDataPoints)\n this.audioAnalysisData.durationMs += segmentResult.durationMs\n this.audioAnalysisData.sampleRate = segmentResult.sampleRate\n\n // Update amplitude range if present\n if (segmentResult.amplitudeRange) {\n this.audioAnalysisData.amplitudeRange = this.mergeRange(\n this.audioAnalysisData.amplitudeRange,\n segmentResult.amplitudeRange\n )\n }\n\n // Update RMS range if present\n if (segmentResult.rmsRange) {\n this.audioAnalysisData.rmsRange = this.mergeRange(\n this.audioAnalysisData.rmsRange,\n segmentResult.rmsRange\n )\n }\n }\n\n /**\n * Merges value ranges\n */\n private mergeRange(\n existing: { min: number; max: number } | undefined,\n newRange: { min: number; max: number }\n ): { min: number; max: number } {\n if (!existing) return { ...newRange }\n\n return {\n min: Math.min(existing.min, newRange.min),\n max: Math.max(existing.max, newRange.max),\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 = 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 = 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 this.dataPointIdCounter = 0 // Reset counter\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 this.audioContext.close().catch((e) => {\n // Log closure errors but continue cleanup\n this.logger?.warn('Error closing AudioContext:', e)\n })\n }\n\n // Safely disconnect audioWorkletNode if it exists\n if (this.audioWorkletNode) {\n try {\n this.audioWorkletNode.disconnect()\n } catch (e) {\n // Log disconnection errors but continue cleanup\n this.logger?.warn('Error disconnecting audioWorkletNode:', e)\n }\n }\n\n // Safely disconnect source if it exists\n if (this.source) {\n try {\n this.source.disconnect()\n } catch (e) {\n // Log disconnection errors but continue cleanup\n this.logger?.warn('Error disconnecting source:', e)\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: unknown) {\n this.logger?.error('Error in resume(): ', error)\n // Rethrow the error to inform callers\n throw new Error(\n `Failed to resume recording: ${error instanceof Error ? error.message : 'unknown error'}`\n )\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 // Store the compressed chunk for final blob creation\n this.compressedChunks.push(event.data)\n this.compressedSize += event.data.size\n\n // Store the pending compressed chunk for the next PCM chunk to use\n this.pendingCompressedChunk = event.data\n }\n }\n } catch (error) {\n this.logger?.error(\n 'Failed to initialize compressed recorder:',\n error\n )\n // Setting to null to indicate initialization failed\n this.compressedMediaRecorder = null\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 this.logger?.warn(\n 'Error disconnecting audioWorkletNode:',\n e\n )\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"]}
1
+ 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packages/expo-audio-stream/src/WebRecorder.web.ts\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { ConsoleLike, RecordingConfig } from './ExpoAudioStream.types'\nimport {\n EmitAudioAnalysisFunction,\n EmitAudioEventFunction,\n} from './ExpoAudioStream.web'\nimport { encodingToBitDepth } from './utils/encodingToBitDepth'\nimport { writeWavHeader } from './utils/writeWavHeader'\nimport { InlineFeaturesExtractor } from './workers/InlineFeaturesExtractor.web'\nimport { InlineAudioWebWorker } from './workers/inlineAudioWebWorker.web'\n\ninterface AudioWorkletEvent {\n data: {\n command: string\n recordedData?: Float32Array\n sampleRate?: number\n position?: number\n message?: string // For debug messages\n }\n}\n\ninterface AudioFeaturesEvent {\n data: {\n command: string\n result: AudioAnalysis\n }\n}\n\nconst DEFAULT_WEB_BITDEPTH = 32\nconst DEFAULT_SEGMENT_DURATION_MS = 100\nconst DEFAULT_WEB_INTERVAL = 500\nconst DEFAULT_WEB_NUMBER_OF_CHANNELS = 1\n\nconst TAG = 'WebRecorder'\n\nexport class WebRecorder {\n public audioContext: AudioContext\n private audioWorkletNode!: AudioWorkletNode\n private featureExtractorWorker?: Worker\n private source: MediaStreamAudioSourceNode\n private emitAudioEventCallback: EmitAudioEventFunction\n private emitAudioAnalysisCallback: EmitAudioAnalysisFunction\n private config: RecordingConfig\n private position: number = 0\n private numberOfChannels: number // Number of audio channels\n private bitDepth: number // Bit depth of the audio\n private exportBitDepth: number // Bit depth of the audio\n private audioAnalysisData: AudioAnalysis // Keep updating the full audio analysis data with latest events\n private readonly 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 readonly mediaStream: MediaStream | null = null\n private readonly 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.output?.compressed?.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] ${event.data.message}`)\n return\n }\n\n if (command !== 'newData') return\n\n const pcmBufferFloat = event.data.recordedData\n if (!pcmBufferFloat) {\n this.logger?.warn('Received empty audio buffer', event)\n return\n }\n\n // Process data in smaller chunks and emit immediately\n const sampleRate =\n event.data.sampleRate ?? this.audioContext.sampleRate\n // Use chunk size from config interval or default to 2 seconds\n const intervalMs = this.config.interval ?? DEFAULT_WEB_INTERVAL\n const chunkSize = Math.floor(sampleRate * (intervalMs / 1000))\n const duration = pcmBufferFloat.length / sampleRate\n\n // Use incoming position if provided by worklet, otherwise use our tracked position\n const incomingPosition =\n typeof event.data.position === 'number'\n ? event.data.position\n : this.position\n\n // Simple position tracking for logging (no duplicate filtering)\n this.logger?.debug(\n `Audio chunk: position=${incomingPosition.toFixed(3)}s, size=${pcmBufferFloat.length}`\n )\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 // Only store PCM data if primary output is enabled\n const shouldStoreUncompressed =\n this.config.output?.primary?.enabled ?? true\n\n // Store PCM chunks when needed - this is for the final WAV file\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 // 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 // Prepare compression data if available\n const compression = this.pendingCompressedChunk\n ? {\n data: this.pendingCompressedChunk,\n size: this.pendingCompressedChunk.size,\n totalSize: this.compressedSize,\n mimeType: 'audio/webm',\n format:\n this.config.output?.compressed?.format ??\n 'opus',\n bitrate:\n this.config.output?.compressed?.bitrate ??\n 128000,\n }\n : undefined\n\n // Emit chunk immediately - whether compressed or not\n this.emitAudioEventCallback({\n data: chunk,\n position: chunkPosition,\n compression,\n })\n\n // Reset pending compressed chunk after we've used it\n this.pendingCompressedChunk = null\n }\n\n // Update our position based on the worklet's position if provided\n this.position = incomingPosition + duration\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.output?.compressed\n ? {\n enabled: this.config.output.compressed.enabled,\n format: this.config.output.compressed.format,\n bitrate: this.config.output.compressed.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') return\n\n const segmentResult = event.data.result\n const uniqueNewDataPoints = this.filterUniqueDataPoints(\n segmentResult.dataPoints\n )\n\n // Update counter based on the highest ID seen\n this.updateDataPointCounter(uniqueNewDataPoints)\n\n // Update analysis data with the new results\n this.updateAudioAnalysisData(segmentResult, uniqueNewDataPoints)\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 * Filters out data points with duplicate IDs\n */\n private filterUniqueDataPoints(dataPoints: any[]): any[] {\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 uniquePoints = dataPoints.filter((dp) => !existingIds.has(dp.id))\n\n // Log filtered duplicates if any\n if (uniquePoints.length < dataPoints.length && this.logger?.warn) {\n this.logger.warn(\n `Filtered ${dataPoints.length - uniquePoints.length} duplicate datapoints`\n )\n }\n\n return uniquePoints\n }\n\n /**\n * Updates the counter based on the highest ID in datapoints\n */\n private updateDataPointCounter(dataPoints: any[]): void {\n if (dataPoints.length === 0) return\n\n const lastDataPoint = dataPoints[dataPoints.length - 1]\n if (lastDataPoint && typeof lastDataPoint.id === 'number') {\n const nextIdValue = lastDataPoint.id + 1\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 /**\n * Updates audio analysis data with segment results\n */\n private updateAudioAnalysisData(\n segmentResult: AudioAnalysis,\n uniqueDataPoints: any[]\n ): void {\n // Add unique data points to our analysis data\n this.audioAnalysisData.dataPoints.push(...uniqueDataPoints)\n this.audioAnalysisData.durationMs += segmentResult.durationMs\n this.audioAnalysisData.sampleRate = segmentResult.sampleRate\n\n // Update amplitude range if present\n if (segmentResult.amplitudeRange) {\n this.audioAnalysisData.amplitudeRange = this.mergeRange(\n this.audioAnalysisData.amplitudeRange,\n segmentResult.amplitudeRange\n )\n }\n\n // Update RMS range if present\n if (segmentResult.rmsRange) {\n this.audioAnalysisData.rmsRange = this.mergeRange(\n this.audioAnalysisData.rmsRange,\n segmentResult.rmsRange\n )\n }\n }\n\n /**\n * Merges value ranges\n */\n private mergeRange(\n existing: { min: number; max: number } | undefined,\n newRange: { min: number; max: number }\n ): { min: number; max: number } {\n if (!existing) return { ...newRange }\n\n return {\n min: Math.min(existing.min, newRange.min),\n max: Math.max(existing.max, newRange.max),\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 = 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 = 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 this.dataPointIdCounter = 0 // Reset counter\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 this.audioContext.close().catch((e) => {\n // Log closure errors but continue cleanup\n this.logger?.warn('Error closing AudioContext:', e)\n })\n }\n\n // Safely disconnect audioWorkletNode if it exists\n if (this.audioWorkletNode) {\n try {\n this.audioWorkletNode.disconnect()\n } catch (e) {\n // Log disconnection errors but continue cleanup\n this.logger?.warn('Error disconnecting audioWorkletNode:', e)\n }\n }\n\n // Safely disconnect source if it exists\n if (this.source) {\n try {\n this.source.disconnect()\n } catch (e) {\n // Log disconnection errors but continue cleanup\n this.logger?.warn('Error disconnecting source:', e)\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: unknown) {\n this.logger?.error('Error in resume(): ', error)\n // Rethrow the error to inform callers\n throw new Error(\n `Failed to resume recording: ${error instanceof Error ? error.message : 'unknown error'}`\n )\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.output?.compressed?.bitrate ?? 128000,\n }\n )\n\n this.compressedMediaRecorder.ondataavailable = (event) => {\n if (event.data.size > 0) {\n // Store the compressed chunk for final blob creation\n this.compressedChunks.push(event.data)\n this.compressedSize += event.data.size\n\n // Store the pending compressed chunk for the next PCM chunk to use\n this.pendingCompressedChunk = event.data\n }\n }\n } catch (error) {\n this.logger?.error(\n 'Failed to initialize compressed recorder:',\n error\n )\n // Setting to null to indicate initialization failed\n this.compressedMediaRecorder = null\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 this.logger?.warn(\n 'Error disconnecting audioWorkletNode:',\n e\n )\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"]}
@@ -1 +1 @@
1
- {"version":3,"file":"ExpoAudioStream.types.js","sourceRoot":"","sources":["../../src/ExpoAudioStream.types.ts"],"names":[],"mappings":"AAyRA,4EAA4E;AAC5E,MAAM,CAAC,MAAM,2BAA2B,GAAG;IACvC,8CAA8C;IAC9C,KAAK,EAAE,OAAO;IACd,sDAAsD;IACtD,QAAQ,EAAE,UAAU;CACd,CAAA","sourcesContent":["// packages/expo-audio-stream/src/ExpoAudioStream.types.ts\nimport {\n AudioAnalysis,\n AudioFeaturesOptions,\n DecodingConfig,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport { AudioAnalysisEvent } from './events'\n\nexport interface CompressionInfo {\n /** Size of the compressed audio data in bytes */\n size: number\n /** MIME type of the compressed audio (e.g., 'audio/aac', 'audio/opus') */\n mimeType: string\n /** Bitrate of the compressed audio in bits per second */\n bitrate: number\n /** Format of the compression (e.g., 'aac', 'opus') */\n format: string\n /** URI to the compressed audio file if available */\n compressedFileUri?: string\n}\n\nexport interface AudioStreamStatus {\n /** Indicates whether audio recording is currently active */\n isRecording: boolean\n /** Indicates whether recording is in a paused state */\n isPaused: boolean\n /** Duration of the current recording in milliseconds */\n durationMs: number\n /** Size of the recorded audio data in bytes */\n size: number\n /** Interval in milliseconds at which recording data is emitted */\n interval: number\n /** Interval in milliseconds at which analysis data is emitted */\n intervalAnalysis: number\n /** MIME type of the recorded audio (e.g., 'audio/wav') */\n mimeType: string\n /** Information about audio compression if enabled */\n compression?: CompressionInfo\n}\n\nexport interface AudioDataEvent {\n /** Audio data as base64 string (native) or Float32Array (web) */\n data: string | Float32Array\n /** Current position in the audio stream in bytes */\n position: number\n /** URI to the file being recorded */\n fileUri: string\n /** Size of the current data chunk in bytes */\n eventDataSize: number\n /** Total size of the recording so far in bytes */\n totalSize: number\n /** Information about compression if enabled, including the compressed data chunk */\n compression?: CompressionInfo & {\n /** Base64 (native) or Blob (web) encoded compressed data chunk */\n data?: string | Blob\n }\n}\n\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\nexport type SampleRate = 16000 | 44100 | 48000\nexport type BitDepth = 8 | 16 | 32\nexport type PCMFormat = `pcm_${BitDepth}bit`\n\nexport type ConsoleLike = {\n /** Logs a message with optional arguments */\n log: (message: string, ...args: unknown[]) => void\n /** Logs a debug message with optional arguments */\n debug: (message: string, ...args: unknown[]) => void\n /** Logs an info message with optional arguments */\n info: (message: string, ...args: unknown[]) => void\n /** Logs a warning message with optional arguments */\n warn: (message: string, ...args: unknown[]) => void\n /** Logs an error message with optional arguments */\n error: (message: string, ...args: unknown[]) => void\n}\n\nexport interface Chunk {\n /** Transcribed text content */\n text: string\n /** Start and end timestamp in seconds [start, end] where end can be null if ongoing */\n timestamp: [number, number | null]\n}\n\nexport interface TranscriberData {\n /** Unique identifier for the transcription */\n id: string\n /** Indicates if the transcriber is currently processing */\n isBusy: boolean\n /** Complete transcribed text */\n text: string\n /** Start time of the transcription in milliseconds */\n startTime: number\n /** End time of the transcription in milliseconds */\n endTime: number\n /** Array of transcribed text chunks with timestamps */\n chunks: Chunk[]\n}\n\nexport interface AudioRecording {\n /** URI to the recorded audio file */\n fileUri: string\n /** Filename of the recorded audio */\n filename: string\n /** Duration of the recording in milliseconds */\n durationMs: number\n /** Size of the recording in bytes */\n size: number\n /** MIME type of the recorded audio */\n mimeType: string\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels: number\n /** Bit depth of the audio (8, 16, or 32 bits) */\n bitDepth: BitDepth\n /** Sample rate of the audio in Hz */\n sampleRate: SampleRate\n /** Timestamp when the recording was created */\n createdAt?: number\n /** Array of transcription data if available */\n transcripts?: TranscriberData[]\n /** Analysis data for the recording if processing was enabled */\n analysisData?: AudioAnalysis\n /** Information about compression if enabled, including the URI to the compressed file */\n compression?: CompressionInfo & {\n /** URI to the compressed audio file */\n compressedFileUri: string\n }\n}\n\nexport interface StartRecordingResult {\n /** URI to the file being recorded */\n fileUri: string\n /** MIME type of the recording */\n mimeType: string\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels?: number\n /** Bit depth of the audio (8, 16, or 32 bits) */\n bitDepth?: BitDepth\n /** Sample rate of the audio in Hz */\n sampleRate?: SampleRate\n /** Information about compression if enabled, including the URI to the compressed file */\n compression?: CompressionInfo & {\n /** URI to the compressed audio file */\n compressedFileUri: string\n }\n}\n\nexport interface AudioSessionConfig {\n /**\n * Audio session category that defines the audio behavior\n * - 'Ambient': Audio continues with silent switch, mixes with other audio\n * - 'SoloAmbient': Audio continues with silent switch, interrupts other audio\n * - 'Playback': Audio continues in background, interrupts other audio\n * - 'Record': Optimized for recording, interrupts other audio\n * - 'PlayAndRecord': Allows simultaneous playback and recording\n * - 'MultiRoute': Routes audio to multiple outputs simultaneously\n */\n category?:\n | 'Ambient'\n | 'SoloAmbient'\n | 'Playback'\n | 'Record'\n | 'PlayAndRecord'\n | 'MultiRoute'\n /**\n * Audio session mode that defines the behavior for specific use cases\n * - 'Default': Standard audio behavior\n * - 'VoiceChat': Optimized for voice chat applications\n * - 'VideoChat': Optimized for video chat applications\n * - 'GameChat': Optimized for in-game chat\n * - 'VideoRecording': Optimized for video recording\n * - 'Measurement': Optimized for audio measurement\n * - 'MoviePlayback': Optimized for movie playback\n * - 'SpokenAudio': Optimized for spoken audio content\n */\n mode?:\n | 'Default'\n | 'VoiceChat'\n | 'VideoChat'\n | 'GameChat'\n | 'VideoRecording'\n | 'Measurement'\n | 'MoviePlayback'\n | 'SpokenAudio'\n /**\n * Options that modify the behavior of the audio session category\n * - 'MixWithOthers': Allows mixing with other active audio sessions\n * - 'DuckOthers': Reduces the volume of other audio sessions\n * - 'InterruptSpokenAudioAndMixWithOthers': Interrupts spoken audio and mixes with others\n * - 'AllowBluetooth': Allows audio routing to Bluetooth devices\n * - 'AllowBluetoothA2DP': Allows audio routing to Bluetooth A2DP devices\n * - 'AllowAirPlay': Allows audio routing to AirPlay devices\n * - 'DefaultToSpeaker': Routes audio to the speaker by default\n */\n categoryOptions?: (\n | 'MixWithOthers'\n | 'DuckOthers'\n | 'InterruptSpokenAudioAndMixWithOthers'\n | 'AllowBluetooth'\n | 'AllowBluetoothA2DP'\n | 'AllowAirPlay'\n | 'DefaultToSpeaker'\n )[]\n}\n\nexport interface IOSConfig {\n /** Configuration for the iOS audio session */\n audioSession?: AudioSessionConfig\n}\n\n/** Web platform specific configuration options */\nexport interface WebConfig {\n /**\n * Whether to store uncompressed audio data for WAV generation\n *\n * When true, all PCM chunks are stored in memory to create a WAV file when compression is disabled\n * When false, uncompressed audio won't be available, but memory usage will be lower\n *\n * Default: true (for backward compatibility)\n */\n storeUncompressedAudio?: boolean\n}\n\n// Add new type for interruption reasons\nexport type RecordingInterruptionReason =\n /** Audio focus was lost to another app */\n | 'audioFocusLoss'\n /** Audio focus was regained */\n | 'audioFocusGain'\n /** Recording was interrupted by a phone call */\n | 'phoneCall'\n /** Phone call that interrupted recording has ended */\n | 'phoneCallEnded'\n /** Recording was stopped by the system or another app */\n | 'recordingStopped'\n /** Recording device was disconnected */\n | 'deviceDisconnected'\n /** Recording switched to default device after disconnection */\n | 'deviceFallback'\n /** A new audio device was connected */\n | 'deviceConnected'\n /** Device switching failed */\n | 'deviceSwitchFailed'\n\n// Add new interface for interruption events\nexport interface RecordingInterruptionEvent {\n /** The reason for the recording interruption */\n reason: RecordingInterruptionReason\n /** Indicates whether the recording is paused due to the interruption */\n isPaused: boolean\n}\n\nexport interface AudioDeviceCapabilities {\n /** Supported sample rates for the device */\n sampleRates: number[]\n /** Supported channel counts for the device */\n channelCounts: number[]\n /** Supported bit depths for the device */\n bitDepths: number[]\n /** Whether the device supports echo cancellation */\n hasEchoCancellation?: boolean\n /** Whether the device supports noise suppression */\n hasNoiseSuppression?: boolean\n /** Whether the device supports automatic gain control */\n hasAutomaticGainControl?: boolean\n}\n\nexport interface AudioDevice {\n /** Unique identifier for the device */\n id: string\n /** Human-readable name of the device */\n name: string\n /** Device type (builtin_mic, bluetooth, etc.) */\n type: string\n /** Whether this is the system default device */\n isDefault: boolean\n /** Audio capabilities for the device */\n capabilities: AudioDeviceCapabilities\n /** Whether the device is currently available */\n isAvailable: boolean\n}\n\n/** Defines how recording should behave when a device becomes unavailable */\nexport const DeviceDisconnectionBehavior = {\n /** Pause recording when device disconnects */\n PAUSE: 'pause',\n /** Switch to default device and continue recording */\n FALLBACK: 'fallback',\n} as const\n\n/** Type for DeviceDisconnectionBehavior values */\nexport type DeviceDisconnectionBehaviorType =\n (typeof DeviceDisconnectionBehavior)[keyof typeof DeviceDisconnectionBehavior]\n\nexport interface RecordingConfig {\n /** Sample rate for recording in Hz (16000, 44100, or 48000) */\n sampleRate?: SampleRate\n\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels?: 1 | 2\n\n /** Encoding type for the recording (pcm_32bit, pcm_16bit, pcm_8bit) */\n encoding?: EncodingType\n\n /** Interval in milliseconds at which to emit recording data */\n interval?: number\n\n /** Interval in milliseconds at which to emit analysis data */\n intervalAnalysis?: number\n\n /** Keep the device awake while recording (default is false) */\n keepAwake?: boolean\n\n /** Show a notification during recording (default is false) */\n showNotification?: boolean\n\n /** Show waveform in the notification (Android only, when showNotification is true) */\n showWaveformInNotification?: boolean\n\n /** Configuration for the notification */\n notification?: NotificationConfig\n\n /** Enable audio processing (default is false) */\n enableProcessing?: boolean\n\n /** iOS-specific configuration */\n ios?: IOSConfig\n\n /** Web-specific configuration options */\n web?: WebConfig\n\n /** Duration of each segment in milliseconds for analysis (default: 100) */\n segmentDurationMs?: number\n\n /** Feature options to extract during audio processing */\n features?: AudioFeaturesOptions\n\n /** Callback function to handle audio stream data */\n onAudioStream?: (_: AudioDataEvent) => Promise<void>\n\n /** Callback function to handle audio features extraction results */\n onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>\n\n /** Configuration for audio compression */\n compression?: {\n /** Enable audio compression */\n enabled: boolean\n /**\n * Format for compression\n * - 'aac': Advanced Audio Coding - supported on all platforms\n * - 'opus': Opus encoding - supported on Android and Web; on iOS will automatically fall back to AAC\n */\n format: 'aac' | 'opus'\n /** Bitrate for compression in bits per second */\n bitrate?: number\n }\n\n /** Whether to automatically resume recording after an interruption (default is false) */\n autoResumeAfterInterruption?: boolean\n\n /** Optional callback to handle recording interruptions */\n onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n\n /** Optional directory path where output files will be saved */\n outputDirectory?: string // If not provided, uses default app directory\n /** Optional filename for the recording (uses UUID if not provided) */\n filename?: string // If not provided, uses UUID\n\n /** ID of the device to use for recording (if not specified, uses default) */\n deviceId?: string\n\n /** How to handle device disconnection during recording */\n deviceDisconnectionBehavior?: DeviceDisconnectionBehaviorType\n}\n\nexport interface NotificationConfig {\n /** Title of the notification */\n title?: string\n\n /** Main text content of the notification */\n text?: string\n\n /** Icon to be displayed in the notification (resource name or URI) */\n icon?: string\n\n /** Android-specific notification configuration */\n android?: {\n /** Unique identifier for the notification channel */\n channelId?: string\n\n /** User-visible name of the notification channel */\n channelName?: string\n\n /** User-visible description of the notification channel */\n channelDescription?: string\n\n /** Unique identifier for this notification */\n notificationId?: number\n\n /** List of actions that can be performed from the notification */\n actions?: NotificationAction[]\n\n /** Configuration for the waveform visualization in the notification */\n waveform?: WaveformConfig\n\n /** Color of the notification LED (if device supports it) */\n lightColor?: string\n\n /** Priority of the notification (affects how it's displayed) */\n priority?: 'min' | 'low' | 'default' | 'high' | 'max'\n\n /** Accent color for the notification (used for the app icon and buttons) */\n accentColor?: string\n }\n\n /** iOS-specific notification configuration */\n ios?: {\n /** Identifier for the notification category (used for grouping similar notifications) */\n categoryIdentifier?: string\n }\n}\n\nexport interface NotificationAction {\n /** Display title for the action */\n title: string\n\n /** Unique identifier for the action */\n identifier: string\n\n /** Icon to be displayed for the action (Android only) */\n icon?: string\n}\n\nexport interface WaveformConfig {\n /** The color of the waveform (e.g., \"#FFFFFF\" for white) */\n color?: string // The color of the waveform (e.g., \"#FFFFFF\" for white)\n /** Opacity of the waveform (0.0 - 1.0) */\n opacity?: number // Opacity of the waveform (0.0 - 1.0)\n /** Width of the waveform line (default: 1.5) */\n strokeWidth?: number // Width of the waveform line (default: 1.5)\n /** Drawing style: \"stroke\" for outline, \"fill\" for solid */\n style?: 'stroke' | 'fill' // Drawing style: \"stroke\" for outline, \"fill\" for solid\n /** Whether to mirror the waveform (symmetrical display) */\n mirror?: boolean // Whether to mirror the waveform (symmetrical display)\n /** Height of the waveform view in dp (default: 64) */\n height?: number // Height of the waveform view in dp (default: 64)\n}\n\nexport interface ExtractAudioDataOptions {\n /** URI of the audio file to extract data from */\n fileUri: string\n /** Start time in milliseconds (for time-based range) */\n startTimeMs?: number\n /** End time in milliseconds (for time-based range) */\n endTimeMs?: number\n /** Start position in bytes (for byte-based range) */\n position?: number\n /** Length in bytes to extract (for byte-based range) */\n length?: number\n /** Include normalized audio data in [-1, 1] range */\n includeNormalizedData?: boolean\n /** Include base64 encoded string representation of the audio data */\n includeBase64Data?: boolean\n /** Include WAV header in the PCM data (makes it a valid WAV file) */\n includeWavHeader?: boolean\n /** Logger for debugging - can pass console directly. */\n logger?: ConsoleLike\n /** Compute the checksum of the PCM data */\n computeChecksum?: boolean\n /** Target config for the normalized audio (Android and Web) */\n decodingOptions?: DecodingConfig\n}\n\nexport interface ExtractedAudioData {\n /** Raw PCM audio data */\n pcmData: Uint8Array\n /** Normalized audio data in [-1, 1] range (when includeNormalizedData is true) */\n normalizedData?: Float32Array\n /** Base64 encoded string representation of the audio data (when includeBase64Data is true) */\n base64Data?: string\n /** Sample rate in Hz (e.g., 44100, 48000) */\n sampleRate: number\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels: number\n /** Bits per sample (8, 16, or 32) */\n bitDepth: BitDepth\n /** Duration of the audio in milliseconds */\n durationMs: number\n /** PCM format identifier (e.g., \"pcm_16bit\") */\n format: PCMFormat\n /** Total number of audio samples per channel */\n samples: number\n /** Whether the pcmData includes a WAV header */\n hasWavHeader?: boolean\n /** CRC32 Checksum of PCM data */\n checksum?: number\n}\n\nexport interface UseAudioRecorderState {\n /**\n * Prepares recording with the specified configuration without starting it.\n *\n * This method eliminates the latency between calling startRecording and the actual recording beginning.\n * It pre-initializes all audio resources, requests permissions, and sets up audio sessions in advance,\n * allowing for true zero-latency recording start when startRecording is called later.\n *\n * Technical benefits:\n * - Eliminates audio pipeline initialization delay (50-300ms depending on platform)\n * - Pre-allocates audio buffers to avoid memory allocation during recording start\n * - Initializes audio hardware in advance (particularly important on iOS)\n * - Requests and verifies permissions before the critical recording moment\n *\n * Use this method when:\n * - You need zero-latency recording start (e.g., voice commands, musical applications)\n * - You're building time-sensitive applications where missing initial audio would be problematic\n * - You want to prepare resources during app initialization, screen loading, or preceding user interaction\n * - You need to ensure recording starts reliably and instantly on all platforms\n *\n * @param config - The recording configuration, identical to what you would pass to startRecording\n * @returns A promise that resolves when preparation is complete\n *\n * @example\n * // Prepare during component mounting\n * useEffect(() => {\n * prepareRecording({\n * sampleRate: 44100,\n * channels: 1,\n * encoding: 'pcm_16bit',\n * });\n * }, []);\n *\n * // Later when user taps record button, it starts with zero latency\n * const handleRecordPress = () => startRecording({\n * sampleRate: 44100,\n * channels: 1,\n * encoding: 'pcm_16bit',\n * });\n */\n prepareRecording: (_: RecordingConfig) => Promise<void>\n /** Starts recording with the specified configuration */\n startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n /** Stops the current recording and returns the recording data */\n stopRecording: () => Promise<AudioRecording | null>\n /** Pauses the current recording */\n pauseRecording: () => Promise<void>\n /** Resumes a paused recording */\n resumeRecording: () => Promise<void>\n /** Indicates whether recording is currently active */\n isRecording: boolean\n /** Indicates whether recording is in a paused state */\n isPaused: boolean\n /** Duration of the current recording in milliseconds */\n durationMs: number // Duration of the recording\n /** Size of the recorded audio in bytes */\n size: number // Size in bytes of the recorded audio\n /** Information about compression if enabled */\n compression?: CompressionInfo\n /** Analysis data for the recording if processing was enabled */\n analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n /** Optional callback to handle recording interruptions */\n onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n\n/**\n * Represents an event emitted during the trimming process to report progress.\n */\nexport interface TrimProgressEvent {\n /**\n * The percentage of the trimming process that has been completed, ranging from 0 to 100.\n */\n progress: number\n\n /**\n * The number of bytes that have been processed so far. This is optional and may not be provided in all implementations.\n */\n bytesProcessed?: number\n\n /**\n * The total number of bytes to process. This is optional and may not be provided in all implementations.\n */\n totalBytes?: number\n}\n\n/**\n * Defines a time range in milliseconds for trimming operations.\n */\nexport interface TimeRange {\n /**\n * The start time of the range in milliseconds.\n */\n startTimeMs: number\n\n /**\n * The end time of the range in milliseconds.\n */\n endTimeMs: number\n}\n\n/**\n * Options for configuring the audio trimming operation.\n */\nexport interface TrimAudioOptions {\n /**\n * The URI of the audio file to trim.\n */\n fileUri: string\n\n /**\n * The mode of trimming to apply.\n * - `'single'`: Trims the audio to a single range defined by `startTimeMs` and `endTimeMs`.\n * - `'keep'`: Keeps the specified `ranges` and removes all other portions of the audio.\n * - `'remove'`: Removes the specified `ranges` and keeps the remaining portions of the audio.\n * @default 'single'\n */\n mode?: 'single' | 'keep' | 'remove'\n\n /**\n * An array of time ranges to keep or remove, depending on the `mode`.\n * - Required for `'keep'` and `'remove'` modes.\n * - Ignored when `mode` is `'single'`.\n */\n ranges?: TimeRange[]\n\n /**\n * The start time in milliseconds for the `'single'` mode.\n * - If not provided, trimming starts from the beginning of the audio (0 ms).\n */\n startTimeMs?: number\n\n /**\n * The end time in milliseconds for the `'single'` mode.\n * - If not provided, trimming extends to the end of the audio.\n */\n endTimeMs?: number\n\n /**\n * The name of the output file. If not provided, a default name will be generated.\n */\n outputFileName?: string\n\n /**\n * Configuration for the output audio format.\n */\n outputFormat?: {\n /**\n * The format of the output audio file.\n * - `'wav'`: Waveform Audio File Format (uncompressed).\n * - `'aac'`: Advanced Audio Coding (compressed). Not supported on web platforms.\n * - `'opus'`: Opus Interactive Audio Codec (compressed).\n */\n format: 'wav' | 'aac' | 'opus'\n\n /**\n * The sample rate of the output audio in Hertz (Hz).\n * - If not provided, the input audio's sample rate is used.\n */\n sampleRate?: number\n\n /**\n * The number of channels in the output audio (e.g., 1 for mono, 2 for stereo).\n * - If not provided, the input audio's channel count is used.\n */\n channels?: number\n\n /**\n * The bit depth of the output audio, applicable to PCM formats like `'wav'`.\n * - If not provided, the input audio's bit depth is used.\n */\n bitDepth?: number\n\n /**\n * The bitrate of the output audio in bits per second, applicable to compressed formats like `'aac'`.\n * - If not provided, a default bitrate is used based on the format.\n */\n bitrate?: number\n }\n\n /**\n * Options for decoding the input audio file.\n * - See `DecodingConfig` for details.\n */\n decodingOptions?: DecodingConfig\n}\n\n/**\n * Result of the audio trimming operation.\n */\nexport interface TrimAudioResult {\n /**\n * The URI of the trimmed audio file.\n */\n uri: string\n\n /**\n * The filename of the trimmed audio file.\n */\n filename: string\n\n /**\n * The duration of the trimmed audio in milliseconds.\n */\n durationMs: number\n\n /**\n * The size of the trimmed audio file in bytes.\n */\n size: number\n\n /**\n * The sample rate of the trimmed audio in Hertz (Hz).\n */\n sampleRate: number\n\n /**\n * The number of channels in the trimmed audio (e.g., 1 for mono, 2 for stereo).\n */\n channels: number\n\n /**\n * The bit depth of the trimmed audio, applicable to PCM formats like `'wav'`.\n */\n bitDepth: number\n\n /**\n * The MIME type of the trimmed audio file (e.g., `'audio/wav'`, `'audio/mpeg'`).\n */\n mimeType: string\n\n /**\n * Information about compression if the output format is compressed.\n */\n compression?: {\n /**\n * The format of the compression (e.g., `'aac'`, `'mp3'`, `'opus'`).\n */\n format: string\n\n /**\n * The bitrate of the compressed audio in bits per second.\n */\n bitrate: number\n\n /**\n * The size of the compressed audio file in bytes.\n */\n size: number\n }\n\n /**\n * Information about the processing time.\n */\n processingInfo?: {\n /**\n * The time it took to process the audio in milliseconds.\n */\n durationMs: number\n }\n}\n"]}
1
+ {"version":3,"file":"ExpoAudioStream.types.js","sourceRoot":"","sources":["../../src/ExpoAudioStream.types.ts"],"names":[],"mappings":"AAiRA,4EAA4E;AAC5E,MAAM,CAAC,MAAM,2BAA2B,GAAG;IACvC,8CAA8C;IAC9C,KAAK,EAAE,OAAO;IACd,sDAAsD;IACtD,QAAQ,EAAE,UAAU;CACd,CAAA","sourcesContent":["// packages/expo-audio-stream/src/ExpoAudioStream.types.ts\nimport {\n AudioAnalysis,\n AudioFeaturesOptions,\n DecodingConfig,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport { AudioAnalysisEvent } from './events'\n\nexport interface CompressionInfo {\n /** Size of the compressed audio data in bytes */\n size: number\n /** MIME type of the compressed audio (e.g., 'audio/aac', 'audio/opus') */\n mimeType: string\n /** Bitrate of the compressed audio in bits per second */\n bitrate: number\n /** Format of the compression (e.g., 'aac', 'opus') */\n format: string\n /** URI to the compressed audio file if available */\n compressedFileUri?: string\n}\n\nexport interface AudioStreamStatus {\n /** Indicates whether audio recording is currently active */\n isRecording: boolean\n /** Indicates whether recording is in a paused state */\n isPaused: boolean\n /** Duration of the current recording in milliseconds */\n durationMs: number\n /** Size of the recorded audio data in bytes */\n size: number\n /** Interval in milliseconds at which recording data is emitted */\n interval: number\n /** Interval in milliseconds at which analysis data is emitted */\n intervalAnalysis: number\n /** MIME type of the recorded audio (e.g., 'audio/wav') */\n mimeType: string\n /** Information about audio compression if enabled */\n compression?: CompressionInfo\n}\n\nexport interface AudioDataEvent {\n /** Audio data as base64 string (native) or Float32Array (web) */\n data: string | Float32Array\n /** Current position in the audio stream in bytes */\n position: number\n /** URI to the file being recorded */\n fileUri: string\n /** Size of the current data chunk in bytes */\n eventDataSize: number\n /** Total size of the recording so far in bytes */\n totalSize: number\n /** Information about compression if enabled, including the compressed data chunk */\n compression?: CompressionInfo & {\n /** Base64 (native) or Blob (web) encoded compressed data chunk */\n data?: string | Blob\n }\n}\n\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\nexport type SampleRate = 16000 | 44100 | 48000\nexport type BitDepth = 8 | 16 | 32\nexport type PCMFormat = `pcm_${BitDepth}bit`\n\nexport type ConsoleLike = {\n /** Logs a message with optional arguments */\n log: (message: string, ...args: unknown[]) => void\n /** Logs a debug message with optional arguments */\n debug: (message: string, ...args: unknown[]) => void\n /** Logs an info message with optional arguments */\n info: (message: string, ...args: unknown[]) => void\n /** Logs a warning message with optional arguments */\n warn: (message: string, ...args: unknown[]) => void\n /** Logs an error message with optional arguments */\n error: (message: string, ...args: unknown[]) => void\n}\n\nexport interface Chunk {\n /** Transcribed text content */\n text: string\n /** Start and end timestamp in seconds [start, end] where end can be null if ongoing */\n timestamp: [number, number | null]\n}\n\nexport interface TranscriberData {\n /** Unique identifier for the transcription */\n id: string\n /** Indicates if the transcriber is currently processing */\n isBusy: boolean\n /** Complete transcribed text */\n text: string\n /** Start time of the transcription in milliseconds */\n startTime: number\n /** End time of the transcription in milliseconds */\n endTime: number\n /** Array of transcribed text chunks with timestamps */\n chunks: Chunk[]\n}\n\nexport interface AudioRecording {\n /** URI to the recorded audio file */\n fileUri: string\n /** Filename of the recorded audio */\n filename: string\n /** Duration of the recording in milliseconds */\n durationMs: number\n /** Size of the recording in bytes */\n size: number\n /** MIME type of the recorded audio */\n mimeType: string\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels: number\n /** Bit depth of the audio (8, 16, or 32 bits) */\n bitDepth: BitDepth\n /** Sample rate of the audio in Hz */\n sampleRate: SampleRate\n /** Timestamp when the recording was created */\n createdAt?: number\n /** Array of transcription data if available */\n transcripts?: TranscriberData[]\n /** Analysis data for the recording if processing was enabled */\n analysisData?: AudioAnalysis\n /** Information about compression if enabled, including the URI to the compressed file */\n compression?: CompressionInfo & {\n /** URI to the compressed audio file */\n compressedFileUri: string\n }\n}\n\nexport interface StartRecordingResult {\n /** URI to the file being recorded */\n fileUri: string\n /** MIME type of the recording */\n mimeType: string\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels?: number\n /** Bit depth of the audio (8, 16, or 32 bits) */\n bitDepth?: BitDepth\n /** Sample rate of the audio in Hz */\n sampleRate?: SampleRate\n /** Information about compression if enabled, including the URI to the compressed file */\n compression?: CompressionInfo & {\n /** URI to the compressed audio file */\n compressedFileUri: string\n }\n}\n\nexport interface AudioSessionConfig {\n /**\n * Audio session category that defines the audio behavior\n * - 'Ambient': Audio continues with silent switch, mixes with other audio\n * - 'SoloAmbient': Audio continues with silent switch, interrupts other audio\n * - 'Playback': Audio continues in background, interrupts other audio\n * - 'Record': Optimized for recording, interrupts other audio\n * - 'PlayAndRecord': Allows simultaneous playback and recording\n * - 'MultiRoute': Routes audio to multiple outputs simultaneously\n */\n category?:\n | 'Ambient'\n | 'SoloAmbient'\n | 'Playback'\n | 'Record'\n | 'PlayAndRecord'\n | 'MultiRoute'\n /**\n * Audio session mode that defines the behavior for specific use cases\n * - 'Default': Standard audio behavior\n * - 'VoiceChat': Optimized for voice chat applications\n * - 'VideoChat': Optimized for video chat applications\n * - 'GameChat': Optimized for in-game chat\n * - 'VideoRecording': Optimized for video recording\n * - 'Measurement': Optimized for audio measurement\n * - 'MoviePlayback': Optimized for movie playback\n * - 'SpokenAudio': Optimized for spoken audio content\n */\n mode?:\n | 'Default'\n | 'VoiceChat'\n | 'VideoChat'\n | 'GameChat'\n | 'VideoRecording'\n | 'Measurement'\n | 'MoviePlayback'\n | 'SpokenAudio'\n /**\n * Options that modify the behavior of the audio session category\n * - 'MixWithOthers': Allows mixing with other active audio sessions\n * - 'DuckOthers': Reduces the volume of other audio sessions\n * - 'InterruptSpokenAudioAndMixWithOthers': Interrupts spoken audio and mixes with others\n * - 'AllowBluetooth': Allows audio routing to Bluetooth devices\n * - 'AllowBluetoothA2DP': Allows audio routing to Bluetooth A2DP devices\n * - 'AllowAirPlay': Allows audio routing to AirPlay devices\n * - 'DefaultToSpeaker': Routes audio to the speaker by default\n */\n categoryOptions?: (\n | 'MixWithOthers'\n | 'DuckOthers'\n | 'InterruptSpokenAudioAndMixWithOthers'\n | 'AllowBluetooth'\n | 'AllowBluetoothA2DP'\n | 'AllowAirPlay'\n | 'DefaultToSpeaker'\n )[]\n}\n\nexport interface IOSConfig {\n /** Configuration for the iOS audio session */\n audioSession?: AudioSessionConfig\n}\n\n/** Web platform specific configuration options */\nexport interface WebConfig {\n // Reserved for future web-specific options\n}\n\n// Add new type for interruption reasons\nexport type RecordingInterruptionReason =\n /** Audio focus was lost to another app */\n | 'audioFocusLoss'\n /** Audio focus was regained */\n | 'audioFocusGain'\n /** Recording was interrupted by a phone call */\n | 'phoneCall'\n /** Phone call that interrupted recording has ended */\n | 'phoneCallEnded'\n /** Recording was stopped by the system or another app */\n | 'recordingStopped'\n /** Recording device was disconnected */\n | 'deviceDisconnected'\n /** Recording switched to default device after disconnection */\n | 'deviceFallback'\n /** A new audio device was connected */\n | 'deviceConnected'\n /** Device switching failed */\n | 'deviceSwitchFailed'\n\n// Add new interface for interruption events\nexport interface RecordingInterruptionEvent {\n /** The reason for the recording interruption */\n reason: RecordingInterruptionReason\n /** Indicates whether the recording is paused due to the interruption */\n isPaused: boolean\n}\n\nexport interface AudioDeviceCapabilities {\n /** Supported sample rates for the device */\n sampleRates: number[]\n /** Supported channel counts for the device */\n channelCounts: number[]\n /** Supported bit depths for the device */\n bitDepths: number[]\n /** Whether the device supports echo cancellation */\n hasEchoCancellation?: boolean\n /** Whether the device supports noise suppression */\n hasNoiseSuppression?: boolean\n /** Whether the device supports automatic gain control */\n hasAutomaticGainControl?: boolean\n}\n\nexport interface AudioDevice {\n /** Unique identifier for the device */\n id: string\n /** Human-readable name of the device */\n name: string\n /** Device type (builtin_mic, bluetooth, etc.) */\n type: string\n /** Whether this is the system default device */\n isDefault: boolean\n /** Audio capabilities for the device */\n capabilities: AudioDeviceCapabilities\n /** Whether the device is currently available */\n isAvailable: boolean\n}\n\n/** Defines how recording should behave when a device becomes unavailable */\nexport const DeviceDisconnectionBehavior = {\n /** Pause recording when device disconnects */\n PAUSE: 'pause',\n /** Switch to default device and continue recording */\n FALLBACK: 'fallback',\n} as const\n\n/** Type for DeviceDisconnectionBehavior values */\nexport type DeviceDisconnectionBehaviorType =\n (typeof DeviceDisconnectionBehavior)[keyof typeof DeviceDisconnectionBehavior]\n\n/**\n * Configuration for audio output files during recording\n */\nexport interface OutputConfig {\n /**\n * Configuration for the primary (uncompressed) output file\n */\n primary?: {\n /** Whether to create the primary output file (default: true) */\n enabled?: boolean\n /** Format for the primary output (currently only 'wav' is supported) */\n format?: 'wav'\n }\n\n /**\n * Configuration for the compressed output file\n */\n compressed?: {\n /** Whether to create a compressed output file (default: false) */\n enabled?: boolean\n /**\n * Format for compression\n * - 'aac': Advanced Audio Coding - supported on all platforms\n * - 'opus': Opus encoding - supported on Android and Web; on iOS will automatically fall back to AAC\n */\n format?: 'aac' | 'opus'\n /** Bitrate for compression in bits per second (default: 128000) */\n bitrate?: number\n }\n\n // Future enhancement: Post-processing pipeline\n // postProcessing?: {\n // normalize?: boolean\n // trimSilence?: boolean\n // noiseReduction?: boolean\n // customProcessors?: AudioProcessor[]\n // }\n}\n\nexport interface RecordingConfig {\n /** Sample rate for recording in Hz (16000, 44100, or 48000) */\n sampleRate?: SampleRate\n\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels?: 1 | 2\n\n /** Encoding type for the recording (pcm_32bit, pcm_16bit, pcm_8bit) */\n encoding?: EncodingType\n\n /** Interval in milliseconds at which to emit recording data */\n interval?: number\n\n /** Interval in milliseconds at which to emit analysis data */\n intervalAnalysis?: number\n\n /** Keep the device awake while recording (default is false) */\n keepAwake?: boolean\n\n /** Show a notification during recording (default is false) */\n showNotification?: boolean\n\n /** Show waveform in the notification (Android only, when showNotification is true) */\n showWaveformInNotification?: boolean\n\n /** Configuration for the notification */\n notification?: NotificationConfig\n\n /** Enable audio processing (default is false) */\n enableProcessing?: boolean\n\n /** iOS-specific configuration */\n ios?: IOSConfig\n\n /** Web-specific configuration options */\n web?: WebConfig\n\n /** Duration of each segment in milliseconds for analysis (default: 100) */\n segmentDurationMs?: number\n\n /** Feature options to extract during audio processing */\n features?: AudioFeaturesOptions\n\n /** Callback function to handle audio stream data */\n onAudioStream?: (_: AudioDataEvent) => Promise<void>\n\n /** Callback function to handle audio features extraction results */\n onAudioAnalysis?: (_: AudioAnalysisEvent) => Promise<void>\n\n /**\n * Configuration for audio output files\n *\n * Examples:\n * - Primary only (default): `{ primary: { enabled: true } }`\n * - Compressed only: `{ primary: { enabled: false }, compressed: { enabled: true, format: 'aac' } }`\n * - Both outputs: `{ compressed: { enabled: true } }`\n * - Streaming only: `{ primary: { enabled: false } }`\n */\n output?: OutputConfig\n\n /** Whether to automatically resume recording after an interruption (default is false) */\n autoResumeAfterInterruption?: boolean\n\n /** Optional callback to handle recording interruptions */\n onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n\n /** Optional directory path where output files will be saved */\n outputDirectory?: string // If not provided, uses default app directory\n /** Optional filename for the recording (uses UUID if not provided) */\n filename?: string // If not provided, uses UUID\n\n /** ID of the device to use for recording (if not specified, uses default) */\n deviceId?: string\n\n /** How to handle device disconnection during recording */\n deviceDisconnectionBehavior?: DeviceDisconnectionBehaviorType\n\n /**\n * Buffer duration in seconds. Controls the size of audio buffers\n * used during recording. Smaller values reduce latency but increase\n * CPU usage. Larger values improve efficiency but increase latency.\n *\n * Platform Notes:\n * - iOS/macOS: Minimum effective 0.1s, uses accumulation below\n * - Android: Respects all sizes within hardware limits\n * - Web: Fully configurable\n *\n * Default: undefined (uses platform default ~23ms at 44.1kHz)\n * Recommended: 0.01 - 0.5 seconds\n * Optimal iOS: >= 0.1 seconds\n */\n bufferDurationSeconds?: number\n}\n\nexport interface NotificationConfig {\n /** Title of the notification */\n title?: string\n\n /** Main text content of the notification */\n text?: string\n\n /** Icon to be displayed in the notification (resource name or URI) */\n icon?: string\n\n /** Android-specific notification configuration */\n android?: {\n /** Unique identifier for the notification channel */\n channelId?: string\n\n /** User-visible name of the notification channel */\n channelName?: string\n\n /** User-visible description of the notification channel */\n channelDescription?: string\n\n /** Unique identifier for this notification */\n notificationId?: number\n\n /** List of actions that can be performed from the notification */\n actions?: NotificationAction[]\n\n /** Configuration for the waveform visualization in the notification */\n waveform?: WaveformConfig\n\n /** Color of the notification LED (if device supports it) */\n lightColor?: string\n\n /** Priority of the notification (affects how it's displayed) */\n priority?: 'min' | 'low' | 'default' | 'high' | 'max'\n\n /** Accent color for the notification (used for the app icon and buttons) */\n accentColor?: string\n }\n\n /** iOS-specific notification configuration */\n ios?: {\n /** Identifier for the notification category (used for grouping similar notifications) */\n categoryIdentifier?: string\n }\n}\n\nexport interface NotificationAction {\n /** Display title for the action */\n title: string\n\n /** Unique identifier for the action */\n identifier: string\n\n /** Icon to be displayed for the action (Android only) */\n icon?: string\n}\n\nexport interface WaveformConfig {\n /** The color of the waveform (e.g., \"#FFFFFF\" for white) */\n color?: string // The color of the waveform (e.g., \"#FFFFFF\" for white)\n /** Opacity of the waveform (0.0 - 1.0) */\n opacity?: number // Opacity of the waveform (0.0 - 1.0)\n /** Width of the waveform line (default: 1.5) */\n strokeWidth?: number // Width of the waveform line (default: 1.5)\n /** Drawing style: \"stroke\" for outline, \"fill\" for solid */\n style?: 'stroke' | 'fill' // Drawing style: \"stroke\" for outline, \"fill\" for solid\n /** Whether to mirror the waveform (symmetrical display) */\n mirror?: boolean // Whether to mirror the waveform (symmetrical display)\n /** Height of the waveform view in dp (default: 64) */\n height?: number // Height of the waveform view in dp (default: 64)\n}\n\nexport interface ExtractAudioDataOptions {\n /** URI of the audio file to extract data from */\n fileUri: string\n /** Start time in milliseconds (for time-based range) */\n startTimeMs?: number\n /** End time in milliseconds (for time-based range) */\n endTimeMs?: number\n /** Start position in bytes (for byte-based range) */\n position?: number\n /** Length in bytes to extract (for byte-based range) */\n length?: number\n /** Include normalized audio data in [-1, 1] range */\n includeNormalizedData?: boolean\n /** Include base64 encoded string representation of the audio data */\n includeBase64Data?: boolean\n /** Include WAV header in the PCM data (makes it a valid WAV file) */\n includeWavHeader?: boolean\n /** Logger for debugging - can pass console directly. */\n logger?: ConsoleLike\n /** Compute the checksum of the PCM data */\n computeChecksum?: boolean\n /** Target config for the normalized audio (Android and Web) */\n decodingOptions?: DecodingConfig\n}\n\nexport interface ExtractedAudioData {\n /** Raw PCM audio data */\n pcmData: Uint8Array\n /** Normalized audio data in [-1, 1] range (when includeNormalizedData is true) */\n normalizedData?: Float32Array\n /** Base64 encoded string representation of the audio data (when includeBase64Data is true) */\n base64Data?: string\n /** Sample rate in Hz (e.g., 44100, 48000) */\n sampleRate: number\n /** Number of audio channels (1 for mono, 2 for stereo) */\n channels: number\n /** Bits per sample (8, 16, or 32) */\n bitDepth: BitDepth\n /** Duration of the audio in milliseconds */\n durationMs: number\n /** PCM format identifier (e.g., \"pcm_16bit\") */\n format: PCMFormat\n /** Total number of audio samples per channel */\n samples: number\n /** Whether the pcmData includes a WAV header */\n hasWavHeader?: boolean\n /** CRC32 Checksum of PCM data */\n checksum?: number\n}\n\nexport interface UseAudioRecorderState {\n /**\n * Prepares recording with the specified configuration without starting it.\n *\n * This method eliminates the latency between calling startRecording and the actual recording beginning.\n * It pre-initializes all audio resources, requests permissions, and sets up audio sessions in advance,\n * allowing for true zero-latency recording start when startRecording is called later.\n *\n * Technical benefits:\n * - Eliminates audio pipeline initialization delay (50-300ms depending on platform)\n * - Pre-allocates audio buffers to avoid memory allocation during recording start\n * - Initializes audio hardware in advance (particularly important on iOS)\n * - Requests and verifies permissions before the critical recording moment\n *\n * Use this method when:\n * - You need zero-latency recording start (e.g., voice commands, musical applications)\n * - You're building time-sensitive applications where missing initial audio would be problematic\n * - You want to prepare resources during app initialization, screen loading, or preceding user interaction\n * - You need to ensure recording starts reliably and instantly on all platforms\n *\n * @param config - The recording configuration, identical to what you would pass to startRecording\n * @returns A promise that resolves when preparation is complete\n *\n * @example\n * // Prepare during component mounting\n * useEffect(() => {\n * prepareRecording({\n * sampleRate: 44100,\n * channels: 1,\n * encoding: 'pcm_16bit',\n * });\n * }, []);\n *\n * // Later when user taps record button, it starts with zero latency\n * const handleRecordPress = () => startRecording({\n * sampleRate: 44100,\n * channels: 1,\n * encoding: 'pcm_16bit',\n * });\n */\n prepareRecording: (_: RecordingConfig) => Promise<void>\n /** Starts recording with the specified configuration */\n startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n /** Stops the current recording and returns the recording data */\n stopRecording: () => Promise<AudioRecording | null>\n /** Pauses the current recording */\n pauseRecording: () => Promise<void>\n /** Resumes a paused recording */\n resumeRecording: () => Promise<void>\n /** Indicates whether recording is currently active */\n isRecording: boolean\n /** Indicates whether recording is in a paused state */\n isPaused: boolean\n /** Duration of the current recording in milliseconds */\n durationMs: number // Duration of the recording\n /** Size of the recorded audio in bytes */\n size: number // Size in bytes of the recorded audio\n /** Information about compression if enabled */\n compression?: CompressionInfo\n /** Analysis data for the recording if processing was enabled */\n analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n /** Optional callback to handle recording interruptions */\n onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n\n/**\n * Represents an event emitted during the trimming process to report progress.\n */\nexport interface TrimProgressEvent {\n /**\n * The percentage of the trimming process that has been completed, ranging from 0 to 100.\n */\n progress: number\n\n /**\n * The number of bytes that have been processed so far. This is optional and may not be provided in all implementations.\n */\n bytesProcessed?: number\n\n /**\n * The total number of bytes to process. This is optional and may not be provided in all implementations.\n */\n totalBytes?: number\n}\n\n/**\n * Defines a time range in milliseconds for trimming operations.\n */\nexport interface TimeRange {\n /**\n * The start time of the range in milliseconds.\n */\n startTimeMs: number\n\n /**\n * The end time of the range in milliseconds.\n */\n endTimeMs: number\n}\n\n/**\n * Options for configuring the audio trimming operation.\n */\nexport interface TrimAudioOptions {\n /**\n * The URI of the audio file to trim.\n */\n fileUri: string\n\n /**\n * The mode of trimming to apply.\n * - `'single'`: Trims the audio to a single range defined by `startTimeMs` and `endTimeMs`.\n * - `'keep'`: Keeps the specified `ranges` and removes all other portions of the audio.\n * - `'remove'`: Removes the specified `ranges` and keeps the remaining portions of the audio.\n * @default 'single'\n */\n mode?: 'single' | 'keep' | 'remove'\n\n /**\n * An array of time ranges to keep or remove, depending on the `mode`.\n * - Required for `'keep'` and `'remove'` modes.\n * - Ignored when `mode` is `'single'`.\n */\n ranges?: TimeRange[]\n\n /**\n * The start time in milliseconds for the `'single'` mode.\n * - If not provided, trimming starts from the beginning of the audio (0 ms).\n */\n startTimeMs?: number\n\n /**\n * The end time in milliseconds for the `'single'` mode.\n * - If not provided, trimming extends to the end of the audio.\n */\n endTimeMs?: number\n\n /**\n * The name of the output file. If not provided, a default name will be generated.\n */\n outputFileName?: string\n\n /**\n * Configuration for the output audio format.\n */\n outputFormat?: {\n /**\n * The format of the output audio file.\n * - `'wav'`: Waveform Audio File Format (uncompressed).\n * - `'aac'`: Advanced Audio Coding (compressed). Not supported on web platforms.\n * - `'opus'`: Opus Interactive Audio Codec (compressed).\n */\n format: 'wav' | 'aac' | 'opus'\n\n /**\n * The sample rate of the output audio in Hertz (Hz).\n * - If not provided, the input audio's sample rate is used.\n */\n sampleRate?: number\n\n /**\n * The number of channels in the output audio (e.g., 1 for mono, 2 for stereo).\n * - If not provided, the input audio's channel count is used.\n */\n channels?: number\n\n /**\n * The bit depth of the output audio, applicable to PCM formats like `'wav'`.\n * - If not provided, the input audio's bit depth is used.\n */\n bitDepth?: number\n\n /**\n * The bitrate of the output audio in bits per second, applicable to compressed formats like `'aac'`.\n * - If not provided, a default bitrate is used based on the format.\n */\n bitrate?: number\n }\n\n /**\n * Options for decoding the input audio file.\n * - See `DecodingConfig` for details.\n */\n decodingOptions?: DecodingConfig\n}\n\n/**\n * Result of the audio trimming operation.\n */\nexport interface TrimAudioResult {\n /**\n * The URI of the trimmed audio file.\n */\n uri: string\n\n /**\n * The filename of the trimmed audio file.\n */\n filename: string\n\n /**\n * The duration of the trimmed audio in milliseconds.\n */\n durationMs: number\n\n /**\n * The size of the trimmed audio file in bytes.\n */\n size: number\n\n /**\n * The sample rate of the trimmed audio in Hertz (Hz).\n */\n sampleRate: number\n\n /**\n * The number of channels in the trimmed audio (e.g., 1 for mono, 2 for stereo).\n */\n channels: number\n\n /**\n * The bit depth of the trimmed audio, applicable to PCM formats like `'wav'`.\n */\n bitDepth: number\n\n /**\n * The MIME type of the trimmed audio file (e.g., `'audio/wav'`, `'audio/mpeg'`).\n */\n mimeType: string\n\n /**\n * Information about compression if the output format is compressed.\n */\n compression?: {\n /**\n * The format of the compression (e.g., `'aac'`, `'mp3'`, `'opus'`).\n */\n format: string\n\n /**\n * The bitrate of the compressed audio in bits per second.\n */\n bitrate: number\n\n /**\n * The size of the compressed audio file in bytes.\n */\n size: number\n }\n\n /**\n * Information about the processing time.\n */\n processingInfo?: {\n /**\n * The time it took to process the audio in milliseconds.\n */\n durationMs: number\n }\n}\n"]}
@@ -200,13 +200,13 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
200
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  bitDepth: this.bitDepth,
201
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  channels: recordingConfig.channels ?? 1,
202
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  sampleRate: recordingConfig.sampleRate ?? 44100,
203
- compression: recordingConfig.compression
203
+ compression: recordingConfig.output?.compressed?.enabled
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  ? {
205
- ...recordingConfig.compression,
206
- bitrate: recordingConfig.compression?.bitrate ?? 128000,
205
+ ...recordingConfig.output.compressed,
206
+ bitrate: recordingConfig.output.compressed.bitrate ?? 128000,
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  size: 0,
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  mimeType: 'audio/webm',
209
- format: recordingConfig.compression?.format ?? 'opus',
209
+ format: recordingConfig.output.compressed.format ?? 'opus',
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  compressedFileUri: '',
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  }
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  : undefined,
@@ -336,45 +336,46 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
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  let compression;
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  let fileUri = `${this.streamUuid}.${this.extension}`;
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  let mimeType = `audio/${this.extension}`;
339
- // Handle both compressed and uncompressed blobs according to configuration
340
- const compressionEnabled = this.recordingConfig?.compression?.enabled ?? false;
341
- // Process compressed blob if available
342
- if (compressedBlob) {
339
+ // Handle both compressed and uncompressed blobs according to new output configuration
340
+ const primaryEnabled = this.recordingConfig?.output?.primary?.enabled ?? true;
341
+ const compressedEnabled = this.recordingConfig?.output?.compressed?.enabled ?? false;
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+ // Process compressed blob if available and enabled
343
+ if (compressedBlob && compressedEnabled) {
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  const compressedUri = URL.createObjectURL(compressedBlob);
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  const compressedInfo = {
345
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  compressedFileUri: compressedUri,
346
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  size: compressedBlob.size,
347
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  mimeType: 'audio/webm',
348
- format: 'opus',
349
- bitrate: this.recordingConfig?.compression?.bitrate ?? 128000,
349
+ format: this.recordingConfig?.output?.compressed?.format ??
350
+ 'opus',
351
+ bitrate: this.recordingConfig?.output?.compressed?.bitrate ??
352
+ 128000,
350
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  };
351
- // If compression is enabled, use compressed blob as primary format
352
- if (compressionEnabled) {
353
- this.logger?.debug('Using compressed audio as primary output');
354
+ // Store compression info
355
+ compression = compressedInfo;
356
+ // If primary is disabled, use compressed as main file
357
+ if (!primaryEnabled) {
358
+ this.logger?.debug('Using compressed audio as primary output (primary disabled)');
354
359
  fileUri = compressedUri;
355
360
  mimeType = 'audio/webm';
356
- // Store compression info
357
- compression = compressedInfo;
358
- }
359
- else {
360
- // Compression was enabled during recording but not set as primary
361
- // Store as alternate format
362
- compression = compressedInfo;
363
361
  }
364
362
  }
365
- // Process uncompressed WAV if available
366
- if (uncompressedBlob) {
363
+ // Process uncompressed WAV if available and primary is enabled
364
+ if (uncompressedBlob && primaryEnabled) {
367
365
  const wavUri = URL.createObjectURL(uncompressedBlob);
368
- // If compression is disabled or no compressed blob is available,
369
- // use WAV as primary format
370
- if (!compressionEnabled || !compressedBlob) {
371
- this.logger?.debug('Using uncompressed WAV as primary output');
372
- fileUri = wavUri;
373
- mimeType = 'audio/wav';
374
- }
366
+ fileUri = wavUri;
367
+ mimeType = 'audio/wav';
368
+ }
369
+ else if (!primaryEnabled && !compressedEnabled) {
370
+ // No outputs enabled - streaming only mode
371
+ this.logger?.debug('No outputs enabled - streaming only mode');
372
+ fileUri = '';
373
+ mimeType = 'audio/wav';
375
374
  }
376
375
  // Use the stored streamUuid for the final filename
377
- const filename = `${this.streamUuid}.${this.extension}`;
376
+ const filename = fileUri
377
+ ? `${this.streamUuid}.${this.extension}`
378
+ : 'stream-only';
378
379
  const result = {
379
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  fileUri,
380
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  filename,
@@ -383,7 +384,7 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
383
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  channels: this.recordingConfig?.channels ?? 1,
384
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  sampleRate: this.recordingConfig?.sampleRate ?? 44100,
385
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  durationMs: this.currentDurationMs,
386
- size: this.currentSize,
387
+ size: primaryEnabled ? this.currentSize : 0,
387
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  mimeType,
388
389
  compression,
389
390
  };
@@ -479,12 +480,14 @@ export class ExpoAudioStreamWeb extends LegacyEventEmitter {
479
480
  interval: this.currentInterval,
480
481
  intervalAnalysis: this.currentIntervalAnalysis,
481
482
  mimeType: `audio/${this.extension}`,
482
- compression: this.recordingConfig?.compression?.enabled
483
+ compression: this.recordingConfig?.output?.compressed?.enabled
483
484
  ? {
484
485
  size: this.totalCompressedSize,
485
486
  mimeType: 'audio/webm',
486
- format: this.recordingConfig.compression.format ?? 'opus',
487
- bitrate: this.recordingConfig.compression.bitrate ?? 128000,
487
+ format: this.recordingConfig.output.compressed.format ??
488
+ 'opus',
489
+ bitrate: this.recordingConfig.output.compressed.bitrate ??
490
+ 128000,
488
491
  compressedFileUri: `${this.streamUuid}.webm`,
489
492
  }
490
493
  : undefined,