npm - @siteed/audio-studio - Versions diffs - 3.0.5 → 3.1.1 - Mend

@siteed/audio-studio 3.0.5 → 3.1.1

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 (63) hide show

package/build/cjs/useAudioRecorder.js.map CHANGED Viewed

	@@ -1 +1 @@
1	- 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src/useAudioRecorder.ts\nimport { EventSubscription, Platform } from 'expo-modules-core'\nimport { useCallback, useEffect, useReducer, useRef, useId } from 'react'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { audioDeviceManager } from './AudioDeviceManager'\nimport {\n AudioDataEvent,\n AudioRecording,\n AudioStreamStatus,\n CompressionInfo,\n ConsoleLike,\n RecordingConfig,\n StartRecordingResult,\n} from './AudioStudio.types'\nimport AudioStudioModule from './AudioStudioModule'\nimport { validateRecordingConfig } from './constants/platformLimitations'\nimport {\n addAudioAnalysisListener,\n addAudioEventListener,\n AudioEventPayload,\n addRecordingInterruptionListener,\n} from './events'\nimport { cleanNativeOptions } from './utils/cleanNativeOptions'\n\nexport interface UseAudioRecorderProps {\n logger?: ConsoleLike\n audioWorkletUrl?: string\n featuresExtratorUrl?: string\n}\n\nexport interface UseAudioRecorderState {\n prepareRecording: (_: RecordingConfig) => Promise<void>\n startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n stopRecording: () => Promise<AudioRecording>\n pauseRecording: () => Promise<void>\n resumeRecording: () => Promise<void>\n isRecording: boolean\n isPaused: boolean\n durationMs: number\n size: number\n compression?: CompressionInfo\n analysisData?: AudioAnalysis\n}\n\ninterface RecorderReducerState {\n isRecording: boolean\n isPaused: boolean\n durationMs: number\n size: number\n compression?: CompressionInfo\n analysisData?: AudioAnalysis\n}\n\ntype RecorderAction =\n \| { type: 'START' \| 'STOP' \| 'PAUSE' \| 'RESUME' }\n \| {\n type: 'UPDATE_RECORDING_STATE'\n payload: {\n isRecording: boolean\n isPaused: boolean\n }\n }\n \| {\n type: 'UPDATE_STATUS'\n payload: {\n durationMs: number\n size: number\n compression?: CompressionInfo\n }\n }\n \| { type: 'UPDATE_ANALYSIS'; payload: AudioAnalysis }\n\nconst defaultAnalysis: AudioAnalysis = {\n segmentDurationMs: 100,\n bitDepth: 32,\n numberOfChannels: 1,\n durationMs: 0,\n sampleRate: 44100,\n samples: 0,\n dataPoints: [],\n rmsRange: {\n min: Number.POSITIVE_INFINITY,\n max: Number.NEGATIVE_INFINITY,\n },\n amplitudeRange: {\n min: Number.POSITIVE_INFINITY,\n max: Number.NEGATIVE_INFINITY,\n },\n extractionTimeMs: 0,\n}\n\nfunction audioRecorderReducer(\n state: RecorderReducerState,\n action: RecorderAction\n): RecorderReducerState {\n switch (action.type) {\n case 'START':\n return {\n ...state,\n isRecording: true,\n isPaused: false,\n durationMs: 0,\n size: 0,\n compression: undefined,\n analysisData: defaultAnalysis,\n }\n case 'STOP':\n return {\n ...state,\n isRecording: false,\n isPaused: false,\n durationMs: 0,\n size: 0,\n compression: undefined,\n analysisData: undefined,\n }\n case 'PAUSE':\n return { ...state, isPaused: true, isRecording: false }\n case 'RESUME':\n return { ...state, isPaused: false, isRecording: true }\n case 'UPDATE_RECORDING_STATE':\n return {\n ...state,\n isPaused: action.payload.isPaused,\n isRecording: action.payload.isRecording,\n }\n case 'UPDATE_STATUS': {\n const newState = {\n ...state,\n durationMs: action.payload.durationMs,\n size: action.payload.size,\n compression: action.payload.compression\n ? {\n size: action.payload.compression.size,\n mimeType: action.payload.compression.mimeType,\n bitrate: action.payload.compression.bitrate,\n format: action.payload.compression.format,\n }\n : undefined,\n }\n return newState\n }\n case 'UPDATE_ANALYSIS':\n return {\n ...state,\n analysisData: action.payload,\n }\n default:\n return state\n }\n}\n\ninterface HandleAudioAnalysisProps {\n analysis: AudioAnalysis\n visualizationDuration: number\n}\n\nexport function useAudioRecorder({\n logger,\n audioWorkletUrl,\n featuresExtratorUrl,\n}: UseAudioRecorderProps = {}): UseAudioRecorderState {\n // Initialize AudioDeviceManager with logger (once)\n if (logger) {\n audioDeviceManager.setLogger(logger)\n }\n const [state, dispatch] = 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[]),\n ...analysis.dataPoints,\n ]\n\n // Calculate the new duration\n // The number of segments is based on how many segments of segmentDurationMs can fit in visualizationDuration\n const numberOfSegments = Math.ceil(\n visualizationDuration / analysis.segmentDurationMs\n )\n // maxDataPoints should be the number of data points, not milliseconds\n const maxDataPoints = numberOfSegments\n\n logger?.debug(\n `[handleAudioAnalysis] Combined data points before trimming: numberOfSegments=${numberOfSegments} visualizationDuration=${visualizationDuration} combinedDataPointsLength=${combinedDataPoints.length} vs maxDataPoints=${maxDataPoints}`\n )\n\n // Trim data points to keep within the maximum number of data points\n if (combinedDataPoints.length > maxDataPoints) {\n combinedDataPoints.splice(\n 0,\n combinedDataPoints.length - maxDataPoints\n )\n }\n\n // Keep the full data points\n fullAnalysisRef.current = {\n ...fullAnalysisRef.current,\n dataPoints: fullCombinedDataPoints,\n }\n fullAnalysisRef.current.durationMs =\n fullCombinedDataPoints.length * analysis.segmentDurationMs\n savedAnalysisData.dataPoints = combinedDataPoints\n savedAnalysisData.bitDepth =\n analysis.bitDepth \|\| savedAnalysisData.bitDepth\n savedAnalysisData.durationMs =\n combinedDataPoints.length * analysis.segmentDurationMs\n\n // Update amplitude range\n const newMin = Math.min(\n savedAnalysisData.amplitudeRange.min,\n analysis.amplitudeRange.min\n )\n const newMax = Math.max(\n savedAnalysisData.amplitudeRange.max,\n analysis.amplitudeRange.max\n )\n\n savedAnalysisData.amplitudeRange = {\n min: newMin,\n max: newMax,\n }\n fullAnalysisRef.current.amplitudeRange = {\n min: newMin,\n max: newMax,\n }\n\n logger?.debug(\n `[handleAudioAnalysis] Updated analysis data: durationMs=${savedAnalysisData.durationMs}`,\n { dataPoints: savedAnalysisData.dataPoints.length }\n )\n\n // Call the onAudioAnalysis callback if it exists in the recording config\n if (recordingConfigRef.current?.onAudioAnalysis) {\n recordingConfigRef.current\n .onAudioAnalysis(analysis)\n .catch((error) => {\n logger?.warn(`Error processing audio analysis:`, error)\n })\n }\n\n // Update the ref\n analysisRef.current = savedAnalysisData\n\n // Dispatch the updated analysis data to state to trigger re-render\n dispatch({\n type: 'UPDATE_ANALYSIS',\n payload: { ...savedAnalysisData },\n })\n },\n [dispatch]\n )\n\n const handleAudioEvent = useCallback(\n async (eventData: AudioEventPayload) => {\n const {\n fileUri,\n deltaSize,\n totalSize,\n lastEmittedSize,\n position,\n streamUuid,\n encoded,\n pcmFloat32,\n mimeType,\n buffer,\n compression,\n } = eventData\n logger?.debug(`[handleAudioEvent] Received audio event:`, {\n fileUri,\n deltaSize,\n totalSize,\n position,\n mimeType,\n lastEmittedSize,\n streamUuid,\n encodedLength: encoded?.length,\n compression,\n })\n if (deltaSize === 0) {\n // Ignore packet with no data\n return\n }\n try {\n // Coming from native ( ios / android ) otherwise buffer is set\n if (Platform.OS !== 'web') {\n const compressionPayload =\n compression && startResultRef.current?.compression\n ? {\n data: compression.data,\n size: compression.totalSize,\n mimeType:\n startResultRef.current.compression\n ?.mimeType,\n bitrate:\n startResultRef.current.compression\n ?.bitrate,\n format: startResultRef.current.compression\n ?.format,\n }\n : undefined\n if (pcmFloat32 != null) {\n // Android new arch delivers Float32Array; iOS delivers number[] — normalize both\n const float32 =\n pcmFloat32 instanceof Float32Array\n ? pcmFloat32\n : new Float32Array(pcmFloat32 as number[])\n onAudioStreamRef.current?.({\n data: float32,\n streamFormat: 'float32',\n position,\n fileUri,\n eventDataSize: deltaSize,\n totalSize,\n compression: compressionPayload,\n })\n } else {\n if (!encoded) {\n logger?.error(`Encoded audio data is missing`)\n throw new Error('Encoded audio data is missing')\n }\n onAudioStreamRef.current?.({\n 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${status.isPaused} isRecording: ${status.isRecording} durationMs: ${status.durationMs} size: ${status.size}`,\n status.compression\n )\n\n // Only dispatch if values actually changed\n if (\n status.isRecording !== stateRef.current.isRecording \|\|\n status.isPaused !== stateRef.current.isPaused\n ) {\n stateRef.current.isRecording = status.isRecording\n stateRef.current.isPaused = status.isPaused\n dispatch({\n type: 'UPDATE_RECORDING_STATE',\n payload: {\n isRecording: status.isRecording,\n isPaused: status.isPaused,\n },\n })\n }\n\n if (\n status.durationMs !== stateRef.current.durationMs \|\|\n status.size !== stateRef.current.size\n ) {\n stateRef.current.durationMs = status.durationMs\n stateRef.current.size = status.size\n stateRef.current.compression = status.compression\n dispatch({\n type: 'UPDATE_STATUS',\n payload: {\n durationMs: status.durationMs,\n size: status.size,\n compression: status.compression,\n },\n })\n }\n } catch (error) {\n logger?.error(`Error getting status:`, error)\n }\n }, [audioStudio, logger]) // Only depend on audioStudio and logger\n\n // Update ref when state changes\n useEffect(() => {\n stateRef.current = {\n isRecording: state.isRecording,\n isPaused: state.isPaused,\n durationMs: state.durationMs,\n size: state.size,\n compression: state.compression,\n }\n }, [\n state.isRecording,\n state.isPaused,\n state.durationMs,\n state.size,\n state.compression,\n ])\n\n const startRecording = useCallback(\n async (recordingOptions: RecordingConfig) => {\n // Validate the encoding configuration\n const validationResult = validateRecordingConfig({\n encoding: recordingOptions.encoding,\n })\n\n // Log warnings if any\n if (validationResult.warnings.length > 0) {\n validationResult.warnings.forEach((warning) => {\n logger?.warn(warning)\n })\n }\n\n // Update recording options with validated values\n const validatedOptions = {\n ...recordingOptions,\n encoding: validationResult.encoding,\n }\n\n recordingConfigRef.current = validatedOptions\n logger?.debug(\n `start recording with validated config`,\n validatedOptions\n )\n\n analysisRef.current = { ...defaultAnalysis } // Reset analysis data\n fullAnalysisRef.current = { ...defaultAnalysis }\n const {\n onAudioStream,\n onRecordingInterrupted,\n onAudioAnalysis,\n ...options\n } = validatedOptions\n const { enableProcessing } = options\n\n const maxRecentDataDuration = 10000 // TODO compute maxRecentDataDuration based on screen dimensions\n if (typeof onAudioStream === 'function') {\n onAudioStreamRef.current = onAudioStream\n } else {\n logger?.warn(`onAudioStream is not a function`, onAudioStream)\n onAudioStreamRef.current = null\n }\n // Strip undefined values and functions that can't cross the native bridge\n const cleanOptions = cleanNativeOptions(options)\n const startResult: StartRecordingResult =\n await audioStudio.startRecording(cleanOptions)\n dispatch({ type: 'START' })\n\n startResultRef.current = startResult\n\n if (enableProcessing) {\n logger?.debug(`Enabling audio analysis listener`)\n const listener = addAudioAnalysisListener(\n async (analysisData) => {\n try {\n await handleAudioAnalysis({\n analysis: analysisData,\n visualizationDuration: maxRecentDataDuration,\n })\n } catch (error) {\n logger?.warn(\n `Error processing audio analysis:`,\n error\n )\n }\n }\n )\n\n analysisListenerRef.current = listener\n }\n\n return startResult\n },\n [handleAudioAnalysis, dispatch]\n )\n\n const prepareRecording = useCallback(\n async (recordingOptions: RecordingConfig) => {\n recordingConfigRef.current = recordingOptions\n logger?.debug(`preparing recording`, recordingOptions)\n\n analysisRef.current = { ...defaultAnalysis } // Reset analysis data\n fullAnalysisRef.current = { ...defaultAnalysis }\n const {\n onAudioStream,\n onRecordingInterrupted,\n onAudioAnalysis,\n ...options\n } = recordingOptions\n\n // Store onAudioStream for later use when recording starts\n if (typeof onAudioStream === 'function') {\n onAudioStreamRef.current = onAudioStream\n } else {\n logger?.warn(`onAudioStream is not a function`, onAudioStream)\n onAudioStreamRef.current = null\n }\n\n // Strip undefined values and functions that can't cross the native bridge\n const cleanOptions = cleanNativeOptions(options)\n // Call the native prepareRecording method\n await audioStudio.prepareRecording(cleanOptions)\n logger?.debug(`recording prepared successfully`)\n },\n []\n )\n\n const stopRecording = useCallback(async () => {\n logger?.debug(`stoping recording`)\n\n const stopResult: AudioRecording = await audioStudio.stopRecording()\n stopResult.analysisData = fullAnalysisRef.current\n\n if (analysisListenerRef.current) {\n analysisListenerRef.current.remove()\n analysisListenerRef.current = null\n }\n onAudioStreamRef.current = null\n\n // Note: We deliberately DON'T clear recordingConfigRef here to preserve interruption callback\n logger?.debug(`recording stopped`, stopResult)\n dispatch({ type: 'STOP' })\n return stopResult\n }, [dispatch])\n\n const pauseRecording = useCallback(async () => {\n logger?.debug(`pause recording`)\n const pauseResult = await audioStudio.pauseRecording()\n dispatch({ type: 'PAUSE' })\n return pauseResult\n }, [dispatch])\n\n const resumeRecording = useCallback(async () => {\n logger?.debug(`resume recording`)\n const resumeResult = await audioStudio.resumeRecording()\n dispatch({ type: 'RESUME' })\n return resumeResult\n }, [dispatch])\n\n useEffect(() => {\n let intervalId: ReturnType<typeof setInterval> \| undefined\n\n if (state.isRecording \|\| state.isPaused) {\n // Immediately check status when starting\n checkStatus()\n\n // Start interval\n intervalId = setInterval(checkStatus, 1000)\n }\n\n return () => {\n if (intervalId) {\n clearInterval(intervalId)\n intervalId = undefined\n }\n }\n }, [checkStatus, state.isRecording, state.isPaused])\n\n useEffect(() => {\n logger?.debug(`Registering audio event listener`)\n const subscribeAudio = addAudioEventListener(handleAudioEvent)\n\n logger?.debug(\n `Subscribed to audio event listener and analysis listener`,\n {\n subscribeAudio,\n }\n )\n\n return () => {\n logger?.debug(`Removing audio event listener`)\n subscribeAudio.remove()\n }\n }, [handleAudioEvent, handleAudioAnalysis])\n\n useEffect(() => {\n // Add event subscription for recording interruptions\n logger?.debug(\n `Setting up recording interruption listener [${instanceId}]`\n )\n\n const subscription = addRecordingInterruptionListener((event) => {\n logger?.debug(\n `[${instanceId}] Received recording interruption event:`,\n event\n )\n\n // Handle device disconnection for UI updates\n if (event.reason === 'deviceDisconnected') {\n logger?.debug(\n `[${instanceId}] Device disconnected - temporarily hiding last device from UI`\n )\n\n // Get current device list before the native layer updates\n const currentDevices = audioDeviceManager.getRawDevices()\n\n // Wait a moment for native layer to update, then compare\n setTimeout(async () => {\n try {\n // Get updated devices without notifying yet\n const updatedDevices =\n await audioDeviceManager.getAvailableDevices({\n refresh: true,\n })\n\n // Find missing devices by comparing lists\n const missingDevices = currentDevices.filter(\n (oldDevice) =>\n !updatedDevices.some(\n (newDevice) => newDevice.id === oldDevice.id\n )\n )\n\n if (missingDevices.length > 0) {\n // Mark all missing devices as disconnected (silently)\n missingDevices.forEach((missingDevice) => {\n logger?.debug(\n `[${instanceId}] Confirmed disconnected device: ${missingDevice.name} (${missingDevice.id})`\n )\n audioDeviceManager.markDeviceAsDisconnected(\n missingDevice.id,\n false\n )\n })\n }\n\n // Notify listeners once with the final filtered state\n audioDeviceManager.notifyListeners()\n } catch (error) {\n logger?.warn(\n `[${instanceId}] Error in delayed device disconnection handling:`,\n error\n )\n }\n }, 500) // 500ms delay to let native layer update\n } else if (event.reason === 'deviceConnected') {\n // Device reconnected - force refresh to show it immediately\n logger?.debug(\n `[${instanceId}] Device connected, forcing refresh`\n )\n audioDeviceManager.forceRefreshDevices()\n }\n\n // Check if we have a callback configured\n logger?.debug(\n `[${instanceId}] recordingConfigRef.current exists:`,\n !!recordingConfigRef.current\n )\n\n if (recordingConfigRef.current?.onRecordingInterrupted) {\n try {\n logger?.debug(\n `[${instanceId}] Calling recording interruption callback`\n )\n recordingConfigRef.current.onRecordingInterrupted(event)\n } catch (error) {\n logger?.error(\n `[${instanceId}] Error in recording interruption callback:`,\n error\n )\n }\n } else {\n logger?.debug(\n `[${instanceId}] No recording interruption callback configured`\n )\n }\n })\n\n return () => {\n logger?.debug(\n `[${instanceId}] Removing recording interruption listener`\n )\n subscription.remove()\n }\n }, [instanceId, logger]) // Include instanceId and logger in dependencies\n\n return {\n prepareRecording,\n startRecording,\n stopRecording,\n pauseRecording,\n resumeRecording,\n isPaused: state.isPaused,\n isRecording: state.isRecording,\n durationMs: state.durationMs,\n size: state.size,\n compression: state.compression,\n analysisData: state.analysisData,\n }\n}\n"]}
1	+ 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src/useAudioRecorder.ts\nimport { EventSubscription, Platform } from 'expo-modules-core'\nimport { useCallback, useEffect, useReducer, useRef, useId } from 'react'\n\nimport { AudioAnalysis } from './AudioAnalysis/AudioAnalysis.types'\nimport { audioDeviceManager } from './AudioDeviceManager'\nimport {\n AudioDataEvent,\n AudioRecording,\n AudioStreamStatus,\n CompressionInfo,\n ConsoleLike,\n RecordingConfig,\n StartRecordingResult,\n} from './AudioStudio.types'\nimport AudioStudioModule from './AudioStudioModule'\nimport { validateRecordingConfig } from './constants/platformLimitations'\nimport {\n addAudioAnalysisListener,\n addAudioEventListener,\n AudioEventPayload,\n addRecordingInterruptionListener,\n} from './events'\nimport { cleanNativeOptions } from './utils/cleanNativeOptions'\n\nexport interface UseAudioRecorderProps {\n logger?: ConsoleLike\n audioWorkletUrl?: string\n featuresExtratorUrl?: string\n}\n\nexport interface UseAudioRecorderState {\n prepareRecording: (_: RecordingConfig) => Promise<void>\n startRecording: (_: RecordingConfig) => Promise<StartRecordingResult>\n stopRecording: () => Promise<AudioRecording>\n pauseRecording: () => Promise<void>\n resumeRecording: () => Promise<void>\n isRecording: boolean\n isPaused: boolean\n durationMs: number\n size: number\n compression?: CompressionInfo\n analysisData?: AudioAnalysis\n}\n\ninterface RecorderReducerState {\n isRecording: boolean\n isPaused: boolean\n durationMs: number\n size: number\n compression?: CompressionInfo\n analysisData?: AudioAnalysis\n}\n\ntype RecorderAction =\n \| { type: 'START' \| 'STOP' \| 'PAUSE' \| 'RESUME' }\n \| {\n type: 'UPDATE_RECORDING_STATE'\n payload: {\n isRecording: boolean\n isPaused: boolean\n }\n }\n \| {\n type: 'UPDATE_STATUS'\n payload: {\n durationMs: number\n size: number\n compression?: CompressionInfo\n }\n }\n \| { type: 'UPDATE_ANALYSIS'; payload: AudioAnalysis }\n\nconst defaultAnalysis: AudioAnalysis = {\n segmentDurationMs: 100,\n bitDepth: 32,\n numberOfChannels: 1,\n durationMs: 0,\n sampleRate: 44100,\n samples: 0,\n dataPoints: [],\n rmsRange: {\n min: Number.POSITIVE_INFINITY,\n max: Number.NEGATIVE_INFINITY,\n },\n amplitudeRange: {\n min: Number.POSITIVE_INFINITY,\n max: Number.NEGATIVE_INFINITY,\n },\n extractionTimeMs: 0,\n}\n\nfunction audioRecorderReducer(\n state: RecorderReducerState,\n action: RecorderAction\n): RecorderReducerState {\n switch (action.type) {\n case 'START':\n return {\n ...state,\n isRecording: true,\n isPaused: false,\n durationMs: 0,\n size: 0,\n compression: undefined,\n analysisData: defaultAnalysis,\n }\n case 'STOP':\n return {\n ...state,\n isRecording: false,\n isPaused: false,\n durationMs: 0,\n size: 0,\n compression: undefined,\n analysisData: undefined,\n }\n case 'PAUSE':\n return { ...state, isPaused: true, isRecording: false }\n case 'RESUME':\n return { ...state, isPaused: false, isRecording: true }\n case 'UPDATE_RECORDING_STATE':\n return {\n ...state,\n isPaused: action.payload.isPaused,\n isRecording: action.payload.isRecording,\n }\n case 'UPDATE_STATUS': {\n const newState = {\n ...state,\n durationMs: action.payload.durationMs,\n size: action.payload.size,\n compression: action.payload.compression\n ? {\n size: action.payload.compression.size,\n mimeType: action.payload.compression.mimeType,\n bitrate: action.payload.compression.bitrate,\n format: action.payload.compression.format,\n }\n : undefined,\n }\n return newState\n }\n case 'UPDATE_ANALYSIS':\n return {\n ...state,\n analysisData: action.payload,\n }\n default:\n return state\n }\n}\n\ninterface HandleAudioAnalysisProps {\n analysis: AudioAnalysis\n visualizationDuration: number\n}\n\nfunction shouldKeepFullAnalysis(config?: RecordingConfig \| null): boolean {\n return config?.keepFullAnalysis !== false\n}\n\nexport function useAudioRecorder({\n logger,\n audioWorkletUrl,\n featuresExtratorUrl,\n}: UseAudioRecorderProps = {}): UseAudioRecorderState {\n // Initialize AudioDeviceManager with logger (once)\n if (logger) {\n audioDeviceManager.setLogger(logger)\n }\n const [state, dispatch] = useReducer(audioRecorderReducer, {\n isRecording: false,\n isPaused: false,\n durationMs: 0,\n size: 0,\n compression: undefined,\n analysisData: undefined,\n })\n\n const startResultRef = useRef<StartRecordingResult \| null>(null)\n\n const analysisListenerRef = useRef<EventSubscription \| null>(null)\n // analysisRef is the current analysis data (last 10 seconds by default)\n const analysisRef = useRef<AudioAnalysis>({ ...defaultAnalysis })\n // fullAnalysisRef is the full analysis data (all data points)\n const fullAnalysisRef = useRef<AudioAnalysis>({\n ...defaultAnalysis,\n })\n\n // Instantiate the module for web with URLs\n const audioStudio =\n Platform.OS === 'web'\n ? AudioStudioModule({\n audioWorkletUrl,\n featuresExtratorUrl,\n logger,\n })\n : AudioStudioModule\n\n const onAudioStreamRef = useRef<\n ((_: AudioDataEvent) => Promise<void>) \| null\n >(null)\n\n const stateRef = useRef({\n isRecording: false,\n isPaused: false,\n durationMs: 0,\n size: 0,\n compression: undefined as CompressionInfo \| undefined,\n })\n\n const recordingConfigRef = useRef<RecordingConfig \| null>(null)\n\n // Generate unique instance ID for debugging\n const instanceId = useId().replace(/:/g, '').slice(0, 5)\n\n const handleAudioAnalysis = useCallback(\n async ({\n analysis,\n visualizationDuration,\n }: HandleAudioAnalysisProps) => {\n const savedAnalysisData = analysisRef.current \|\| {\n ...defaultAnalysis,\n }\n\n const maxDuration = visualizationDuration\n\n logger?.debug(\n `[handleAudioAnalysis] Received audio analysis: maxDuration=${maxDuration} analysis.dataPoints=${analysis.dataPoints.length} analysisData.dataPoints=${savedAnalysisData.dataPoints.length}`\n )\n\n // Combine data points\n const combinedDataPoints = [\n ...savedAnalysisData.dataPoints,\n ...analysis.dataPoints,\n ]\n\n const keepFullAnalysis = shouldKeepFullAnalysis(\n recordingConfigRef.current\n )\n const fullCombinedDataPoints = keepFullAnalysis\n ? 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fullCombinedDataPoints) {\n fullAnalysisRef.current = {\n ...fullAnalysisRef.current,\n dataPoints: fullCombinedDataPoints,\n }\n fullAnalysisRef.current.durationMs =\n fullCombinedDataPoints.length * analysis.segmentDurationMs\n }\n savedAnalysisData.dataPoints = combinedDataPoints\n savedAnalysisData.bitDepth =\n analysis.bitDepth \|\| savedAnalysisData.bitDepth\n savedAnalysisData.durationMs =\n combinedDataPoints.length * analysis.segmentDurationMs\n\n // Update amplitude range\n const newMin = Math.min(\n savedAnalysisData.amplitudeRange.min,\n analysis.amplitudeRange.min\n )\n const newMax = Math.max(\n savedAnalysisData.amplitudeRange.max,\n analysis.amplitudeRange.max\n )\n\n savedAnalysisData.amplitudeRange = {\n min: newMin,\n max: newMax,\n }\n if (keepFullAnalysis) {\n fullAnalysisRef.current.amplitudeRange = {\n min: newMin,\n max: newMax,\n }\n }\n\n logger?.debug(\n `[handleAudioAnalysis] Updated analysis data: durationMs=${savedAnalysisData.durationMs}`,\n { dataPoints: savedAnalysisData.dataPoints.length }\n )\n\n // Call the onAudioAnalysis callback if it exists in the recording config\n if (recordingConfigRef.current?.onAudioAnalysis) {\n recordingConfigRef.current\n .onAudioAnalysis(analysis)\n .catch((error) => {\n logger?.warn(`Error processing audio analysis:`, error)\n })\n }\n\n // Update the ref\n analysisRef.current = savedAnalysisData\n\n // Dispatch the updated analysis data to state to trigger re-render\n dispatch({\n type: 'UPDATE_ANALYSIS',\n payload: { ...savedAnalysisData },\n })\n },\n [dispatch]\n )\n\n const handleAudioEvent = useCallback(\n async (eventData: AudioEventPayload) => {\n const {\n fileUri,\n deltaSize,\n totalSize,\n lastEmittedSize,\n position,\n streamUuid,\n encoded,\n pcmFloat32,\n mimeType,\n buffer,\n compression,\n } = eventData\n logger?.debug(`[handleAudioEvent] Received audio event:`, {\n fileUri,\n deltaSize,\n totalSize,\n position,\n mimeType,\n lastEmittedSize,\n streamUuid,\n encodedLength: encoded?.length,\n compression,\n })\n if (deltaSize === 0) {\n // Ignore packet with no data\n return\n }\n try {\n // Coming from native ( ios / android ) otherwise buffer is set\n if (Platform.OS !== 'web') {\n const compressionPayload =\n compression && startResultRef.current?.compression\n ? {\n data: compression.data,\n size: compression.totalSize,\n mimeType:\n startResultRef.current.compression\n ?.mimeType,\n bitrate:\n startResultRef.current.compression\n ?.bitrate,\n format: startResultRef.current.compression\n ?.format,\n }\n : undefined\n if (pcmFloat32 != null) {\n // Android new arch delivers Float32Array; iOS delivers number[] — normalize both\n const float32 =\n pcmFloat32 instanceof Float32Array\n ? pcmFloat32\n : new Float32Array(pcmFloat32 as number[])\n onAudioStreamRef.current?.({\n data: float32,\n streamFormat: 'float32',\n position,\n fileUri,\n eventDataSize: deltaSize,\n totalSize,\n compression: compressionPayload,\n })\n } else {\n if (!encoded) {\n logger?.error(`Encoded audio data is missing`)\n throw new Error('Encoded audio data is missing')\n }\n onAudioStreamRef.current?.({\n data: encoded,\n position,\n fileUri,\n eventDataSize: deltaSize,\n totalSize,\n compression: compressionPayload,\n })\n }\n } else if (buffer) {\n // Coming from web\n const webEvent: AudioDataEvent = {\n data: buffer,\n position,\n fileUri,\n eventDataSize: deltaSize,\n totalSize,\n compression:\n compression && startResultRef.current?.compression\n ? {\n data: compression.data,\n size: compression.totalSize,\n mimeType:\n startResultRef.current.compression\n ?.mimeType,\n bitrate:\n startResultRef.current.compression\n ?.bitrate,\n format: startResultRef.current.compression\n ?.format,\n }\n : undefined,\n }\n onAudioStreamRef.current?.(webEvent)\n logger?.debug(\n `[handleAudioEvent] Audio data sent to onAudioStream`,\n webEvent\n )\n }\n } catch (error) {\n logger?.error(`Error processing audio event:`, error)\n }\n },\n []\n )\n\n const checkStatus = useCallback(async () => {\n try {\n const status: AudioStreamStatus = audioStudio.status()\n logger?.debug(\n `Status: paused: ${status.isPaused} isRecording: ${status.isRecording} durationMs: ${status.durationMs} size: ${status.size}`,\n status.compression\n )\n\n // Only dispatch if values actually changed\n if (\n status.isRecording !== stateRef.current.isRecording \|\|\n status.isPaused !== stateRef.current.isPaused\n ) {\n stateRef.current.isRecording = status.isRecording\n stateRef.current.isPaused = status.isPaused\n dispatch({\n type: 'UPDATE_RECORDING_STATE',\n payload: {\n isRecording: status.isRecording,\n isPaused: status.isPaused,\n },\n })\n }\n\n if (\n status.durationMs !== stateRef.current.durationMs \|\|\n status.size !== stateRef.current.size\n ) {\n stateRef.current.durationMs = status.durationMs\n stateRef.current.size = status.size\n stateRef.current.compression = status.compression\n dispatch({\n type: 'UPDATE_STATUS',\n payload: {\n durationMs: status.durationMs,\n size: status.size,\n compression: status.compression,\n },\n })\n }\n } catch (error) {\n logger?.error(`Error getting status:`, error)\n }\n }, [audioStudio, logger]) // Only depend on audioStudio and logger\n\n // Update ref when state changes\n useEffect(() => {\n stateRef.current = {\n isRecording: state.isRecording,\n isPaused: state.isPaused,\n durationMs: state.durationMs,\n size: state.size,\n compression: state.compression,\n }\n }, [\n state.isRecording,\n state.isPaused,\n state.durationMs,\n state.size,\n state.compression,\n ])\n\n const startRecording = useCallback(\n async (recordingOptions: RecordingConfig) => {\n // Validate the encoding configuration\n const validationResult = validateRecordingConfig({\n encoding: recordingOptions.encoding,\n })\n\n // Log warnings if any\n if (validationResult.warnings.length > 0) {\n validationResult.warnings.forEach((warning) => {\n logger?.warn(warning)\n })\n }\n\n // Update recording options with validated values\n const validatedOptions = {\n ...recordingOptions,\n encoding: validationResult.encoding,\n }\n\n recordingConfigRef.current = validatedOptions\n logger?.debug(\n `start recording with validated config`,\n validatedOptions\n )\n\n analysisRef.current = { ...defaultAnalysis } // Reset analysis data\n fullAnalysisRef.current = { ...defaultAnalysis }\n const {\n onAudioStream,\n onRecordingInterrupted,\n onAudioAnalysis,\n keepFullAnalysis: _keepFullAnalysis,\n ...options\n } = validatedOptions\n const { enableProcessing } = options\n\n const maxRecentDataDuration = 10000 // TODO compute maxRecentDataDuration based on screen dimensions\n if (typeof onAudioStream === 'function') {\n onAudioStreamRef.current = onAudioStream\n } else {\n logger?.warn(`onAudioStream is not a function`, onAudioStream)\n onAudioStreamRef.current = null\n }\n // Strip undefined values and functions that can't cross the native bridge\n const cleanOptions = cleanNativeOptions(options)\n const startResult: StartRecordingResult =\n await audioStudio.startRecording(cleanOptions)\n dispatch({ type: 'START' })\n\n startResultRef.current = startResult\n\n if (enableProcessing) {\n logger?.debug(`Enabling audio analysis listener`)\n const listener = addAudioAnalysisListener(\n async (analysisData) => {\n try {\n await handleAudioAnalysis({\n analysis: analysisData,\n visualizationDuration: maxRecentDataDuration,\n })\n } catch (error) {\n logger?.warn(\n `Error processing audio analysis:`,\n error\n )\n }\n }\n )\n\n analysisListenerRef.current = listener\n }\n\n return startResult\n },\n [handleAudioAnalysis, dispatch]\n )\n\n const prepareRecording = useCallback(\n async (recordingOptions: RecordingConfig) => {\n recordingConfigRef.current = recordingOptions\n logger?.debug(`preparing recording`, recordingOptions)\n\n analysisRef.current = { ...defaultAnalysis } // Reset analysis data\n fullAnalysisRef.current = { ...defaultAnalysis }\n const {\n onAudioStream,\n onRecordingInterrupted,\n onAudioAnalysis,\n keepFullAnalysis: _keepFullAnalysis,\n ...options\n } = recordingOptions\n\n // Store onAudioStream for later use when recording starts\n if (typeof onAudioStream === 'function') {\n onAudioStreamRef.current = onAudioStream\n } else {\n logger?.warn(`onAudioStream is not a function`, onAudioStream)\n onAudioStreamRef.current = null\n }\n\n // Strip undefined values and functions that can't cross the native bridge\n const cleanOptions = cleanNativeOptions(options)\n // Call the native prepareRecording method\n await audioStudio.prepareRecording(cleanOptions)\n logger?.debug(`recording prepared successfully`)\n },\n []\n )\n\n const stopRecording = useCallback(async () => {\n logger?.debug(`stoping recording`)\n\n const stopResult: AudioRecording = await audioStudio.stopRecording()\n if (shouldKeepFullAnalysis(recordingConfigRef.current)) {\n stopResult.analysisData = fullAnalysisRef.current\n } else {\n // `keepFullAnalysis` is a hook-level retention policy. If a platform\n // starts returning native analysisData in the future, keep opt-out\n // semantics explicit and avoid leaking a full history here.\n delete stopResult.analysisData\n }\n\n if (analysisListenerRef.current) {\n analysisListenerRef.current.remove()\n analysisListenerRef.current = null\n }\n onAudioStreamRef.current = null\n\n // Note: We deliberately DON'T clear recordingConfigRef here to preserve interruption callback\n logger?.debug(`recording stopped`, stopResult)\n dispatch({ type: 'STOP' })\n return stopResult\n }, [dispatch])\n\n const pauseRecording = useCallback(async () => {\n logger?.debug(`pause recording`)\n const pauseResult = await audioStudio.pauseRecording()\n dispatch({ type: 'PAUSE' })\n return pauseResult\n }, [dispatch])\n\n const resumeRecording = useCallback(async () => {\n logger?.debug(`resume recording`)\n const resumeResult = await audioStudio.resumeRecording()\n dispatch({ type: 'RESUME' })\n return resumeResult\n }, [dispatch])\n\n useEffect(() => {\n let intervalId: ReturnType<typeof setInterval> \| undefined\n\n if (state.isRecording \|\| state.isPaused) {\n // Immediately check status when starting\n checkStatus()\n\n // Start interval\n intervalId = setInterval(checkStatus, 1000)\n }\n\n return () => {\n if (intervalId) {\n clearInterval(intervalId)\n intervalId = undefined\n }\n }\n }, [checkStatus, state.isRecording, state.isPaused])\n\n useEffect(() => {\n logger?.debug(`Registering audio event listener`)\n const subscribeAudio = addAudioEventListener(handleAudioEvent)\n\n logger?.debug(\n `Subscribed to audio event listener and analysis listener`,\n {\n subscribeAudio,\n }\n )\n\n return () => {\n logger?.debug(`Removing audio event listener`)\n subscribeAudio.remove()\n }\n }, [handleAudioEvent, handleAudioAnalysis])\n\n useEffect(() => {\n // Add event subscription for recording interruptions\n logger?.debug(\n `Setting up recording interruption listener [${instanceId}]`\n )\n\n const subscription = addRecordingInterruptionListener((event) => {\n logger?.debug(\n `[${instanceId}] Received recording interruption event:`,\n event\n )\n\n // Handle device disconnection for UI updates\n if (event.reason === 'deviceDisconnected') {\n logger?.debug(\n `[${instanceId}] Device disconnected - temporarily hiding last device from UI`\n )\n\n // Get current device list before the native layer updates\n const currentDevices = audioDeviceManager.getRawDevices()\n\n // Wait a moment for native layer to update, then compare\n setTimeout(async () => {\n try {\n // Get updated devices without notifying yet\n const updatedDevices =\n await audioDeviceManager.getAvailableDevices({\n refresh: true,\n })\n\n // Find missing devices by comparing lists\n const missingDevices = currentDevices.filter(\n (oldDevice) =>\n !updatedDevices.some(\n (newDevice) => newDevice.id === oldDevice.id\n )\n )\n\n if (missingDevices.length > 0) {\n // Mark all missing devices as disconnected (silently)\n missingDevices.forEach((missingDevice) => {\n logger?.debug(\n `[${instanceId}] Confirmed disconnected device: ${missingDevice.name} (${missingDevice.id})`\n )\n audioDeviceManager.markDeviceAsDisconnected(\n missingDevice.id,\n false\n )\n })\n }\n\n // Notify listeners once with the final filtered state\n audioDeviceManager.notifyListeners()\n } catch (error) {\n logger?.warn(\n `[${instanceId}] Error in delayed device disconnection handling:`,\n error\n )\n }\n }, 500) // 500ms delay to let native layer update\n } else if (event.reason === 'deviceConnected') {\n // Device reconnected - force refresh to show it immediately\n logger?.debug(\n `[${instanceId}] Device connected, forcing refresh`\n )\n audioDeviceManager.forceRefreshDevices()\n }\n\n // Check if we have a callback configured\n logger?.debug(\n `[${instanceId}] recordingConfigRef.current exists:`,\n !!recordingConfigRef.current\n )\n\n if (recordingConfigRef.current?.onRecordingInterrupted) {\n try {\n logger?.debug(\n `[${instanceId}] Calling recording interruption callback`\n )\n recordingConfigRef.current.onRecordingInterrupted(event)\n } catch (error) {\n logger?.error(\n `[${instanceId}] Error in recording interruption callback:`,\n error\n )\n }\n } else {\n logger?.debug(\n `[${instanceId}] No recording interruption callback configured`\n )\n }\n })\n\n return () => {\n logger?.debug(\n `[${instanceId}] Removing recording interruption listener`\n )\n subscription.remove()\n }\n }, [instanceId, logger]) // Include instanceId and logger in dependencies\n\n return {\n prepareRecording,\n startRecording,\n stopRecording,\n pauseRecording,\n resumeRecording,\n isPaused: state.isPaused,\n isRecording: state.isRecording,\n durationMs: state.durationMs,\n size: state.size,\n compression: state.compression,\n analysisData: state.analysisData,\n }\n}\n"]}

package/build/esm/AudioAnalysis/AudioAnalysis.types.js.map CHANGED Viewed

	@@ -1 +1 @@
1	- {"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,iEAAiE","sourcesContent":["// packages/audio-studio/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../AudioStudio.types'\n\n/*\n Represents the configuration for decoding audio data.\n /\nexport interface DecodingConfig {\n /* Target sample rate for decoded audio (Android and Web) /\n targetSampleRate?: number\n /* Target number of channels (Android and Web) /\n targetChannels?: number\n /* Target bit depth (Android and Web) /\n targetBitDepth?: BitDepth\n /* Whether to normalize audio levels (Android and Web) /\n normalizeAudio?: boolean\n}\n\n/\n Represents speech-related features extracted from audio.\n /\nexport interface SpeechFeatures {\n isActive: boolean // Whether speech is detected in this segment\n speakerId?: number // Optional speaker identification\n // Could add more speech-related features here like:\n // confidence: number\n // language?: string\n // sentiment?: number\n // etc.\n}\n\n/\n Represents various audio features extracted from an audio signal.\n /\nexport interface AudioFeatures {\n energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n minAmplitude?: number // Minimum amplitude value in the audio signal.\n maxAmplitude?: number // Maximum amplitude value in the audio signal.\n zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n tonnetz?: number[] // Tonal network features representing harmonic relationships.\n pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/\n Options to specify which audio features to extract.\n * Note: Advanced features (spectral features, chromagram, pitch, etc.) are experimental,\n * especially during live recording, due to high processing requirements.\n /\nexport interface AudioFeaturesOptions {\n // Basic features - well optimized\n energy?: boolean\n rms?: boolean\n zcr?: boolean\n\n // Advanced features - experimental, may impact performance in live recording\n mfcc?: boolean\n spectralCentroid?: boolean\n spectralFlatness?: boolean\n spectralRolloff?: boolean\n spectralBandwidth?: boolean\n chromagram?: boolean\n tempo?: boolean\n hnr?: boolean\n melSpectrogram?: boolean\n spectralContrast?: boolean\n tonnetz?: boolean\n pitch?: boolean\n\n // Utility\n crc32?: boolean\n}\n\n/\n Represents a single data point in the audio analysis.\n /\nexport interface DataPoint {\n id: number\n amplitude: number // Peak amplitude for the segment\n rms: number // Root mean square value\n dB: number // dBFS (decibels relative to full scale) computed from RMS value\n silent: boolean // Always computed\n features?: AudioFeatures\n speech?: SpeechFeatures\n startTime?: number\n endTime?: number\n // start / end position in bytes\n startPosition?: number\n endPosition?: number\n // number of audio samples for this point (samples size depends on bit depth)\n samples?: number\n}\n\n/\n Represents the complete data from the audio analysis.\n /\nexport interface AudioAnalysis {\n segmentDurationMs: number // Duration of each segment in milliseconds\n durationMs: number // Duration of the audio in milliseconds\n /\n Bit depth used for audio analysis processing.\n \n Important: This represents the internal processing bit depth, which may differ\n * from the recording bit depth. Audio is typically converted to 32-bit float for\n * analysis to ensure precision in calculations, regardless of the original recording format.\n \n Platform behavior:\n * - iOS: Always 32 (float processing)\n * - Android: Always 32 (float processing)\n * - Web: Always 32 (Web Audio API standard)\n \n The actual recorded file will maintain the requested bit depth (8, 16, or 32).\n /\n bitDepth: number\n samples: number // Size of the audio in bytes\n numberOfChannels: number // Number of audio channels\n sampleRate: number // Sample rate of the audio\n dataPoints: DataPoint[] // Array of data points from the analysis.\n amplitudeRange: {\n min: number\n max: number\n }\n rmsRange: {\n min: number\n max: number\n }\n extractionTimeMs: number // Time taken to extract/process the analysis in milliseconds\n // TODO: speaker changes into a broader speech analysis section\n speechAnalysis?: {\n speakerChanges: {\n timestamp: number\n speakerId: number\n }[]\n // Could add more speech analysis data here like:\n // dominantSpeaker?: number\n // totalSpeechDuration?: number\n // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n }\n}\n\n/\n Options for specifying a time range within an audio file.\n /\nexport interface AudioRangeOptions {\n /* Start time in milliseconds /\n startTimeMs?: number\n /* End time in milliseconds /\n endTimeMs?: number\n}\n\n/\n Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n /\nexport interface PreviewOptions extends AudioRangeOptions {\n /* URI of the audio file to analyze /\n fileUri: string\n /\n Total number of points to generate for the preview.\n * @default 100\n /\n numberOfPoints?: number\n /\n Optional logger for debugging.\n /\n logger?: ConsoleLike\n /\n Optional configuration for decoding the audio file.\n * Defaults to:\n * - targetSampleRate: undefined (keep original)\n * - targetChannels: undefined (keep original)\n * - targetBitDepth: 16\n * - normalizeAudio: false\n /\n decodingOptions?: DecodingConfig\n}\n\n/\n Options for mel-spectrogram extraction\n \n @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n /\nexport interface ExtractMelSpectrogramOptions {\n fileUri?: string // Path to audio file\n arrayBuffer?: ArrayBuffer // Raw audio buffer\n windowSizeMs: number // Window size in ms (e.g., 25)\n hopLengthMs: number // Hop length in ms (e.g., 10)\n nMels: number // Number of mel filters (e.g., 60)\n fMin?: number // Min frequency (default: 0)\n fMax?: number // Max frequency (default: sampleRate / 2)\n windowType?: 'hann' \| 'hamming' // Window function (default: 'hann')\n normalize?: boolean // Mean normalization (default: false)\n logScale?: boolean // Log scaling of mel energies (default: true)\n decodingOptions?: DecodingConfig // Audio decoding settings\n /* Optional start time in ms. If neither startTimeMs nor endTimeMs is set, defaults to 0. /\n startTimeMs?: number\n /* Optional end time in ms. Clamped so that the range does not exceed MAX_DURATION_MS (30 s). /\n endTimeMs?: number\n logger?: ConsoleLike\n}\n\n/\n Result type for WASM-based audio feature extraction.\n /\nexport interface AudioFeaturesWasmResult {\n spectralCentroid: number\n spectralFlatness: number\n spectralRolloff: number\n spectralBandwidth: number\n mfcc: number[]\n chromagram: number[]\n}\n\n/\n Return type for mel spectrogram extraction\n \n @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n spectrogram: number[][] // 2D array [time][mel]\n sampleRate: number // Audio sample rate\n nMels: number // Number of mel filters\n timeSteps: number // Number of time frames\n durationMs: number // Audio duration in ms\n}\n"]}
1	+ {"version":3,"file":"AudioAnalysis.types.js","sourceRoot":"","sources":["../../../src/AudioAnalysis/AudioAnalysis.types.ts"],"names":[],"mappings":"AAAA,iEAAiE","sourcesContent":["// packages/audio-studio/src/AudioAnalysis/AudioAnalysis.types.ts\n\nimport { BitDepth, ConsoleLike } from '../AudioStudio.types'\n\n/*\n Represents the configuration for decoding audio data.\n /\nexport interface DecodingConfig {\n /* Target sample rate for decoded audio (Android and Web) /\n targetSampleRate?: number\n /* Target number of channels (Android and Web) /\n targetChannels?: number\n /* Target bit depth (Android and Web) /\n targetBitDepth?: BitDepth\n /* Whether to normalize audio levels (Android and Web) /\n normalizeAudio?: boolean\n /\n RMS threshold below which a segment is flagged silent.\n * Range 0..1. Default 0.01.\n * Currently applied as a JS post-process so the same behavior holds across iOS/Android/Web.\n /\n silenceRmsThreshold?: number\n}\n\n/\n Represents speech-related features extracted from audio.\n /\nexport interface SpeechFeatures {\n isActive: boolean // Whether speech is detected in this segment\n speakerId?: number // Optional speaker identification\n // Could add more speech-related features here like:\n // confidence: number\n // language?: string\n // sentiment?: number\n // etc.\n}\n\n/\n Represents various audio features extracted from an audio signal.\n /\nexport interface AudioFeatures {\n energy?: number // The infinite integral of the squared signal, representing the overall energy of the audio.\n mfcc?: number[] // Mel-frequency cepstral coefficients, describing the short-term power spectrum of a sound.\n rms?: number // Root mean square value, indicating the amplitude of the audio signal.\n minAmplitude?: number // Minimum amplitude value in the audio signal.\n maxAmplitude?: number // Maximum amplitude value in the audio signal.\n zcr?: number // Zero-crossing rate, indicating the rate at which the signal changes sign.\n spectralCentroid?: number // The center of mass of the spectrum, indicating the brightness of the sound.\n spectralFlatness?: number // Measure of the flatness of the spectrum, indicating how noise-like the signal is.\n spectralRolloff?: number // The frequency below which a specified percentage (usually 85%) of the total spectral energy lies.\n spectralBandwidth?: number // The width of the spectrum, indicating the range of frequencies present.\n chromagram?: number[] // Chromagram, representing the 12 different pitch classes of the audio.\n tempo?: number // Estimated tempo of the audio signal, measured in beats per minute (BPM).\n hnr?: number // Harmonics-to-noise ratio, indicating the proportion of harmonics to noise in the audio signal.\n melSpectrogram?: number[] // Mel-scaled spectrogram representation of the audio.\n spectralContrast?: number[] // Spectral contrast features representing the difference between peaks and valleys.\n tonnetz?: number[] // Tonal network features representing harmonic relationships.\n pitch?: number // Pitch of the audio signal, measured in Hertz (Hz).\n crc32?: number // crc32 checksum of the audio signal, used to verify the integrity of the audio.\n}\n\n/\n Options to specify which audio features to extract.\n * Note: Advanced features (spectral features, chromagram, pitch, etc.) are experimental,\n * especially during live recording, due to high processing requirements.\n /\nexport interface AudioFeaturesOptions {\n // Basic features - well optimized\n energy?: boolean\n rms?: boolean\n zcr?: boolean\n\n // Advanced features - experimental, may impact performance in live recording\n mfcc?: boolean\n spectralCentroid?: boolean\n spectralFlatness?: boolean\n spectralRolloff?: boolean\n spectralBandwidth?: boolean\n chromagram?: boolean\n tempo?: boolean\n hnr?: boolean\n melSpectrogram?: boolean\n spectralContrast?: boolean\n tonnetz?: boolean\n pitch?: boolean\n\n // Utility\n crc32?: boolean\n}\n\n/\n Represents a single data point in the audio analysis.\n /\nexport interface DataPoint {\n id: number\n amplitude: number // Peak amplitude for the segment\n rms: number // Root mean square value\n dB: number // dBFS (decibels relative to full scale) computed from RMS value\n silent: boolean // Always computed\n features?: AudioFeatures\n speech?: SpeechFeatures\n startTime?: number\n endTime?: number\n // start / end position in bytes\n startPosition?: number\n endPosition?: number\n // number of audio samples for this point (samples size depends on bit depth)\n samples?: number\n}\n\n/\n Represents the complete data from the audio analysis.\n /\nexport interface AudioAnalysis {\n segmentDurationMs: number // Duration of each segment in milliseconds\n durationMs: number // Duration of the audio in milliseconds\n /\n Bit depth used for audio analysis processing.\n \n Important: This represents the internal processing bit depth, which may differ\n * from the recording bit depth. Audio is typically converted to 32-bit float for\n * analysis to ensure precision in calculations, regardless of the original recording format.\n \n Platform behavior:\n * - iOS: Always 32 (float processing)\n * - Android: Always 32 (float processing)\n * - Web: Always 32 (Web Audio API standard)\n \n The actual recorded file will maintain the requested bit depth (8, 16, or 32).\n /\n bitDepth: number\n samples: number // Size of the audio in bytes\n numberOfChannels: number // Number of audio channels\n sampleRate: number // Sample rate of the audio\n dataPoints: DataPoint[] // Array of data points from the analysis.\n amplitudeRange: {\n min: number\n max: number\n }\n rmsRange: {\n min: number\n max: number\n }\n extractionTimeMs: number // Time taken to extract/process the analysis in milliseconds\n // TODO: speaker changes into a broader speech analysis section\n speechAnalysis?: {\n speakerChanges: {\n timestamp: number\n speakerId: number\n }[]\n // Could add more speech analysis data here like:\n // dominantSpeaker?: number\n // totalSpeechDuration?: number\n // speakerStats?: { [speakerId: number]: { duration: number, segments: number } }\n }\n}\n\n/\n Options for specifying a time range within an audio file.\n /\nexport interface AudioRangeOptions {\n /* Start time in milliseconds /\n startTimeMs?: number\n /* End time in milliseconds /\n endTimeMs?: number\n}\n\n/\n Options for generating a quick preview of audio waveform.\n * This is optimized for UI rendering with a specified number of points.\n /\nexport interface PreviewBar {\n /* Stable zero-based bar identifier. /\n id: number\n /* Peak amplitude for this bar, normalized to 0..1. /\n amplitude: number\n /* Root mean square amplitude for this bar, normalized to 0..1. /\n rms: number\n /* Whether this bar is below the configured silence RMS threshold. /\n silent: boolean\n /* Bar start time in milliseconds from the extracted range start. /\n startTimeMs: number\n /* Bar end time in milliseconds from the extracted range start. /\n endTimeMs: number\n}\n\n/\n Compact preview-bars result for UI waveform rendering.\n * Unlike `AudioAnalysis`, this intentionally omits full `DataPoint` feature data.\n /\nexport interface PreviewBarsResult {\n bars: PreviewBar[]\n durationMs: number\n sampleRate: number\n numberOfChannels: number\n bitDepth: number\n samples: number\n /* Requested bar count before native/platform clamping. /\n requestedNumberOfBars: number\n /* Approximate duration represented by each bar. /\n barDurationMs: number\n amplitudeRange: {\n min: number\n max: number\n }\n rmsRange: {\n min: number\n max: number\n }\n extractionTimeMs: number\n}\n\n/\n Options for extracting compact waveform preview bars for UI rendering.\n /\nexport interface PreviewBarsOptions extends AudioRangeOptions {\n /* URI of the audio file to analyze /\n fileUri: string\n /\n Total number of bars to generate for the preview.\n * @default 100\n /\n numberOfBars?: number\n /* Optional logger for debugging. /\n logger?: ConsoleLike\n /* Optional configuration for decoding the audio file. /\n decodingOptions?: DecodingConfig\n /\n Optional callback fired once per compact bar after extraction resolves.\n * Native progressive streaming is not implied by this callback.\n /\n onBarReady?: (bar: PreviewBar, index: number, total: number) => void\n}\n\nexport interface PreviewOptions extends AudioRangeOptions {\n /* URI of the audio file to analyze /\n fileUri: string\n /\n Total number of points to generate for the preview.\n * @default 100\n /\n numberOfPoints?: number\n /\n Optional logger for debugging.\n /\n logger?: ConsoleLike\n /\n Optional configuration for decoding the audio file.\n * Defaults to:\n * - targetSampleRate: undefined (keep original)\n * - targetChannels: undefined (keep original)\n * - targetBitDepth: 16\n * - normalizeAudio: false\n /\n decodingOptions?: DecodingConfig\n /\n Optional callback fired once per data point as the preview becomes available.\n * Today the native module returns the full analysis in one shot; the points are then\n * micro-batched on the JS side so consumers can render bars incrementally.\n * Native progressive streaming is a future enhancement.\n /\n onPointReady?: (point: DataPoint, index: number, total: number) => void\n /\n Optional cancellation signal for JS-side progressive point emission.\n * Aborting does not cancel native extraction after it has started, but it\n * stops any queued `onPointReady` callbacks from an older request.\n /\n signal?: AbortSignal\n}\n\n/\n Options for mel-spectrogram extraction\n \n @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n /\nexport interface ExtractMelSpectrogramOptions {\n fileUri?: string // Path to audio file\n arrayBuffer?: ArrayBuffer // Raw audio buffer\n windowSizeMs: number // Window size in ms (e.g., 25)\n hopLengthMs: number // Hop length in ms (e.g., 10)\n nMels: number // Number of mel filters (e.g., 60)\n fMin?: number // Min frequency (default: 0)\n fMax?: number // Max frequency (default: sampleRate / 2)\n windowType?: 'hann' \| 'hamming' // Window function (default: 'hann')\n normalize?: boolean // Mean normalization (default: false)\n logScale?: boolean // Log scaling of mel energies (default: true)\n decodingOptions?: DecodingConfig // Audio decoding settings\n /* Optional start time in ms. If neither startTimeMs nor endTimeMs is set, defaults to 0. /\n startTimeMs?: number\n /* Optional end time in ms. Clamped so that the range does not exceed MAX_DURATION_MS (30 s). /\n endTimeMs?: number\n logger?: ConsoleLike\n}\n\n/\n Result type for WASM-based audio feature extraction.\n /\nexport interface AudioFeaturesWasmResult {\n spectralCentroid: number\n spectralFlatness: number\n spectralRolloff: number\n spectralBandwidth: number\n mfcc: number[]\n chromagram: number[]\n}\n\n/\n Return type for mel spectrogram extraction\n \n @experimental This feature is experimental and currently only available on Android.\n * The API may change in future versions.\n */\nexport interface MelSpectrogram {\n spectrogram: number[][] // 2D array [time][mel]\n sampleRate: number // Audio sample rate\n nMels: number // Number of mel filters\n timeSteps: number // Number of time frames\n durationMs: number // Audio duration in ms\n}\n"]}

package/build/esm/AudioAnalysis/extractPreview.js CHANGED Viewed

@@ -1,25 +1,102 @@
+import { mapExtractionError } from '../errors/AudioExtractionError';
 import { extractAudioAnalysis } from './extractAudioAnalysis';
+const DEFAULT_SILENCE_THRESHOLD = 0.01;
+/**
+ * Apply a silence threshold to the data points by recomputing the `silent` flag from rms.
+ * Returns a new array (does not mutate the source).
+ */
+function applySilenceThreshold(dataPoints, threshold) {
+    return dataPoints.map((p) => ({
+        ...p,
+        silent: p.rms < threshold,
+    }));
+}
+const SMALL_TOTAL_INSTANT_THRESHOLD = 50;
+const PROGRESSIVE_BATCH_DELAY_MS = 30;
+const PROGRESSIVE_BATCH_COUNT = 8;
+/**
+ * Schedule progressive emission of points after the native one-shot resolve.
+ * Native progressive streaming is a future enhancement; today the points are
+ * micro-batched on the JS side so consumers (and the agentic recipe runner)
+ * can observe an in-flight `pointsReceived < totalPoints` window.
+ */
+function emitPointsProgressively(dataPoints, onPointReady, signal, logger) {
+    const total = dataPoints.length;
+    if (total === 0)
+        return;
+    const safeEmit = (point, index) => {
+        if (signal?.aborted)
+            return;
+        try {
+            onPointReady(point, index, total);
+        }
+        catch (err) {
+            // Swallow callback errors so a buggy consumer cannot break extraction.
+            logger?.warn?.('extractPreview onPointReady callback failed', err);
+        }
+    };
+    if (signal?.aborted)
+        return;
+    if (total <= SMALL_TOTAL_INSTANT_THRESHOLD) {
+        for (let i = 0; i < total; i++)
+            safeEmit(dataPoints[i], i);
+        return;
+    }
+    // First quarter flushes immediately so the UI shows something within a frame.
+    const firstFlushCount = Math.max(1, Math.floor(total / 4));
+    for (let i = 0; i < firstFlushCount; i++)
+        safeEmit(dataPoints[i], i);
+    if (firstFlushCount >= total)
+        return;
+    const remaining = total - firstFlushCount;
+    const batchSize = Math.max(1, Math.ceil(remaining / PROGRESSIVE_BATCH_COUNT));
+    let cursor = firstFlushCount;
+    const pump = () => {
+        if (signal?.aborted)
+            return;
+        const end = Math.min(total, cursor + batchSize);
+        for (let i = cursor; i < end; i++)
+            safeEmit(dataPoints[i], i);
+        cursor = end;
+        if (cursor < total) {
+            setTimeout(pump, PROGRESSIVE_BATCH_DELAY_MS);
+        }
+    };
+    setTimeout(pump, PROGRESSIVE_BATCH_DELAY_MS);
+}
 /**
  * Generates a simplified preview of the audio waveform for quick visualization.
  * Ideal for UI rendering with a specified number of points.
  *
  * @param options - The options for the preview, including file URI and time range.
  * @returns A promise that resolves to the audio preview data.
+ * @throws {AudioExtractionError} when the underlying extraction fails.
  */
-export async function extractPreview({ fileUri, numberOfPoints = 100, startTimeMs = 0, endTimeMs = 30000, // First 30 seconds
-decodingOptions, logger, }) {
-    const durationMs = endTimeMs - startTimeMs;
-    const segmentDurationMs = Math.floor(durationMs / numberOfPoints);
-    // Call extractAudioAnalysis with calculated parameters
-    const analysis = await extractAudioAnalysis({
-        fileUri,
-        startTimeMs,
-        endTimeMs,
-        logger,
-        segmentDurationMs,
-        decodingOptions,
-    });
-    // Transform the result into AudioPreview format
-    return analysis;
+export async function extractPreview({ fileUri, numberOfPoints = 100, startTimeMs = 0, endTimeMs = 30000, decodingOptions, logger, onPointReady, signal, }) {
+    const durationMs = Math.max(1, endTimeMs - startTimeMs);
+    const segmentDurationMs = Math.max(1, Math.floor(durationMs / numberOfPoints));
+    let analysis;
+    try {
+        analysis = await extractAudioAnalysis({
+            fileUri,
+            startTimeMs,
+            endTimeMs,
+            logger,
+            segmentDurationMs,
+            decodingOptions,
+        });
+    }
+    catch (err) {
+        throw mapExtractionError(err, fileUri);
+    }
+    const threshold = decodingOptions?.silenceRmsThreshold ?? DEFAULT_SILENCE_THRESHOLD;
+    const adjusted = {
+        ...analysis,
+        dataPoints: applySilenceThreshold(analysis.dataPoints, threshold),
+    };
+    if (onPointReady) {
+        emitPointsProgressively(adjusted.dataPoints, onPointReady, signal, logger);
+    }
+    return adjusted;
 }
 //# sourceMappingURL=extractPreview.js.map

package/build/esm/AudioAnalysis/extractPreview.js.map CHANGED Viewed

	@@ -1 +1 @@
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today the points are\n * micro-batched on the JS side so consumers (and the agentic recipe runner)\n * can observe an in-flight `pointsReceived < totalPoints` window.\n /\nfunction emitPointsProgressively(\n dataPoints: DataPoint[],\n onPointReady: NonNullable<PreviewOptions['onPointReady']>,\n signal?: PreviewOptions['signal'],\n logger?: PreviewOptions['logger']\n): void {\n const total = dataPoints.length\n if (total === 0) return\n\n const safeEmit = (point: DataPoint, index: number) => {\n if (signal?.aborted) return\n try {\n onPointReady(point, index, total)\n } catch (err) {\n // Swallow callback errors so a buggy consumer cannot break extraction.\n logger?.warn?.('extractPreview onPointReady callback failed', err)\n }\n }\n\n if (signal?.aborted) return\n if (total <= SMALL_TOTAL_INSTANT_THRESHOLD) {\n for (let i = 0; i < total; i++) safeEmit(dataPoints[i], i)\n return\n }\n\n // First quarter flushes immediately so the UI shows something within a frame.\n const firstFlushCount = Math.max(1, Math.floor(total / 4))\n for (let i = 0; i < firstFlushCount; i++) safeEmit(dataPoints[i], i)\n\n if (firstFlushCount >= total) return\n\n const remaining = total - firstFlushCount\n const batchSize = Math.max(\n 1,\n Math.ceil(remaining / PROGRESSIVE_BATCH_COUNT)\n )\n let cursor = firstFlushCount\n const pump = () => {\n if (signal?.aborted) return\n const end = Math.min(total, cursor + batchSize)\n for (let i = cursor; i < end; i++) safeEmit(dataPoints[i], i)\n cursor = end\n if (cursor < total) {\n setTimeout(pump, PROGRESSIVE_BATCH_DELAY_MS)\n }\n }\n setTimeout(pump, PROGRESSIVE_BATCH_DELAY_MS)\n}\n\n/\n Generates a simplified preview of the audio waveform for quick visualization.\n * Ideal for UI rendering with a specified number of points.\n \n @param options - The options for the preview, including file URI and time range.\n * @returns A promise that resolves to the audio preview data.\n * @throws {AudioExtractionError} when the underlying extraction fails.\n */\nexport async function extractPreview({\n fileUri,\n numberOfPoints = 100,\n startTimeMs = 0,\n endTimeMs = 30000,\n decodingOptions,\n logger,\n onPointReady,\n signal,\n}: PreviewOptions): Promise<AudioAnalysis> {\n const durationMs = Math.max(1, endTimeMs - startTimeMs)\n const segmentDurationMs = Math.max(\n 1,\n Math.floor(durationMs / numberOfPoints)\n )\n\n let analysis: AudioAnalysis\n try {\n analysis = await extractAudioAnalysis({\n fileUri,\n startTimeMs,\n endTimeMs,\n logger,\n segmentDurationMs,\n decodingOptions,\n })\n } catch (err) {\n throw mapExtractionError(err, fileUri)\n }\n\n const threshold =\n decodingOptions?.silenceRmsThreshold ?? DEFAULT_SILENCE_THRESHOLD\n const adjusted: AudioAnalysis = {\n ...analysis,\n dataPoints: applySilenceThreshold(analysis.dataPoints, threshold),\n }\n\n if (onPointReady) {\n emitPointsProgressively(\n adjusted.dataPoints,\n onPointReady,\n signal,\n logger\n )\n }\n\n return adjusted\n}\n"]}

package/build/esm/AudioAnalysis/extractPreviewBars.js ADDED Viewed

@@ -0,0 +1,128 @@
+import { extractPreview } from './extractPreview';
+import AudioStudioModule from '../AudioStudioModule';
+import { mapExtractionError } from '../errors/AudioExtractionError';
+import { cleanNativeOptions } from '../utils/cleanNativeOptions';
+const DEFAULT_PREVIEW_BARS = 100;
+const DEFAULT_PREVIEW_END_TIME_MS = 30000;
+const DEFAULT_SILENCE_THRESHOLD = 0.01;
+function hasNativePreviewBars(module) {
+    return (module !== null &&
+        (typeof module === 'object' || typeof module === 'function') &&
+        typeof module.extractPreviewBars ===
+            'function');
+}
+function clamp01(value) {
+    if (!Number.isFinite(value))
+        return 0;
+    return Math.max(0, Math.min(1, value));
+}
+function getPointTimeMs(value, fallbackMs) {
+    return typeof value === 'number' && Number.isFinite(value)
+        ? Math.round(value)
+        : fallbackMs;
+}
+function pointToPreviewBar(point, index, fallbackBarDurationMs, silenceRmsThreshold) {
+    const fallbackStartTimeMs = Math.round(index * fallbackBarDurationMs);
+    const fallbackEndTimeMs = Math.round((index + 1) * fallbackBarDurationMs);
+    const startTimeMs = getPointTimeMs(point.startTime, fallbackStartTimeMs);
+    const endTimeMs = getPointTimeMs(point.endTime, fallbackEndTimeMs);
+    const rms = clamp01(point.rms);
+    return {
+        id: point.id ?? index,
+        amplitude: clamp01(point.amplitude),
+        rms,
+        silent: point.silent ?? rms < silenceRmsThreshold,
+        startTimeMs,
+        endTimeMs: Math.max(startTimeMs, endTimeMs),
+    };
+}
+function calculateRange(values) {
+    if (values.length === 0) {
+        return { min: 0, max: 0 };
+    }
+    let min = Number.POSITIVE_INFINITY;
+    let max = Number.NEGATIVE_INFINITY;
+    for (const value of values) {
+        const safeValue = clamp01(value);
+        min = Math.min(min, safeValue);
+        max = Math.max(max, safeValue);
+    }
+    return { min, max };
+}
+function fromAudioAnalysis(analysis, requestedNumberOfBars, silenceRmsThreshold) {
+    const barDurationMs = analysis.segmentDurationMs ||
+        Math.max(1, analysis.durationMs / Math.max(1, analysis.dataPoints.length));
+    const bars = analysis.dataPoints.map((point, index) => pointToPreviewBar(point, index, barDurationMs, silenceRmsThreshold));
+    return {
+        bars,
+        durationMs: analysis.durationMs,
+        sampleRate: analysis.sampleRate,
+        numberOfChannels: analysis.numberOfChannels,
+        bitDepth: analysis.bitDepth,
+        samples: analysis.samples,
+        requestedNumberOfBars,
+        barDurationMs,
+        amplitudeRange: calculateRange(bars.map((bar) => bar.amplitude)),
+        rmsRange: calculateRange(bars.map((bar) => bar.rms)),
+        extractionTimeMs: analysis.extractionTimeMs,
+    };
+}
+function emitBarsProgressively(bars, onBarReady) {
+    const total = bars.length;
+    for (let index = 0; index < total; index++) {
+        onBarReady(bars[index], index, total);
+    }
+}
+/**
+ * Extracts compact waveform preview bars for UI rendering.
+ *
+ * Native platforms may provide a compact `extractPreviewBars` bridge. Until that
+ * bridge is available, this safely falls back to the existing `extractPreview`
+ * compatibility path and adapts `DataPoint` objects into compact bars.
+ *
+ * @throws {AudioExtractionError} when the underlying extraction fails.
+ */
+export async function extractPreviewBars({ fileUri, numberOfBars = DEFAULT_PREVIEW_BARS, startTimeMs = 0, endTimeMs = DEFAULT_PREVIEW_END_TIME_MS, decodingOptions, logger, onBarReady, }) {
+    const requestedNumberOfBars = Math.max(1, Math.floor(numberOfBars));
+    const nativeOptions = {
+        fileUri,
+        numberOfBars: requestedNumberOfBars,
+        startTimeMs,
+        endTimeMs,
+        decodingOptions,
+    };
+    const nativeModule = AudioStudioModule;
+    if (hasNativePreviewBars(nativeModule)) {
+        let result;
+        try {
+            result = await nativeModule.extractPreviewBars(cleanNativeOptions(nativeOptions));
+        }
+        catch (err) {
+            throw mapExtractionError(err, fileUri);
+        }
+        if (onBarReady) {
+            emitBarsProgressively(result.bars, onBarReady);
+        }
+        return result;
+    }
+    let analysis;
+    try {
+        analysis = await extractPreview({
+            fileUri,
+            numberOfPoints: requestedNumberOfBars,
+            startTimeMs,
+            endTimeMs,
+            decodingOptions,
+            logger,
+        });
+    }
+    catch (err) {
+        throw mapExtractionError(err, fileUri);
+    }
+    const result = fromAudioAnalysis(analysis, requestedNumberOfBars, decodingOptions?.silenceRmsThreshold ?? DEFAULT_SILENCE_THRESHOLD);
+    if (onBarReady) {
+        emitBarsProgressively(result.bars, onBarReady);
+    }
+    return result;
+}
+//# sourceMappingURL=extractPreviewBars.js.map

package/build/esm/AudioAnalysis/extractPreviewBars.js.map ADDED Viewed

@@ -0,0 +1 @@

+ 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{\n AudioAnalysis,\n DataPoint,\n PreviewBar,\n PreviewBarsOptions,\n PreviewBarsResult,\n} from './AudioAnalysis.types'\nimport { extractPreview } from './extractPreview'\nimport AudioStudioModule from '../AudioStudioModule'\nimport { mapExtractionError } from '../errors/AudioExtractionError'\nimport { cleanNativeOptions } from '../utils/cleanNativeOptions'\n\nconst DEFAULT_PREVIEW_BARS = 100\nconst DEFAULT_PREVIEW_END_TIME_MS = 30000\nconst DEFAULT_SILENCE_THRESHOLD = 0.01\n\ninterface NativePreviewBarsModule {\n extractPreviewBars: (\n options: Record<string, unknown>\n ) => Promise<PreviewBarsResult>\n}\n\nfunction hasNativePreviewBars(\n module: unknown\n): module is NativePreviewBarsModule {\n return (\n module !== null &&\n (typeof module === 'object' || typeof module === 'function') &&\n typeof (module as NativePreviewBarsModule).extractPreviewBars ===\n 'function'\n )\n}\n\nfunction clamp01(value: number): number {\n if (!Number.isFinite(value)) return 0\n return Math.max(0, Math.min(1, value))\n}\n\nfunction getPointTimeMs(value: number | undefined, fallbackMs: number): number {\n return typeof value === 'number' && Number.isFinite(value)\n ? Math.round(value)\n : fallbackMs\n}\n\nfunction pointToPreviewBar(\n point: DataPoint,\n index: number,\n fallbackBarDurationMs: number,\n silenceRmsThreshold: number\n): PreviewBar {\n const fallbackStartTimeMs = Math.round(index * fallbackBarDurationMs)\n const fallbackEndTimeMs = Math.round((index + 1) * fallbackBarDurationMs)\n const startTimeMs = getPointTimeMs(point.startTime, fallbackStartTimeMs)\n const endTimeMs = getPointTimeMs(point.endTime, fallbackEndTimeMs)\n const rms = clamp01(point.rms)\n\n return {\n id: point.id ?? index,\n amplitude: clamp01(point.amplitude),\n rms,\n silent: point.silent ?? rms < silenceRmsThreshold,\n startTimeMs,\n endTimeMs: Math.max(startTimeMs, endTimeMs),\n }\n}\n\nfunction calculateRange(values: number[]): { min: number; max: number } {\n if (values.length === 0) {\n return { min: 0, max: 0 }\n }\n\n let min = Number.POSITIVE_INFINITY\n let max = Number.NEGATIVE_INFINITY\n\n for (const value of values) {\n const safeValue = clamp01(value)\n min = Math.min(min, safeValue)\n max = Math.max(max, safeValue)\n }\n\n return { min, max }\n}\n\nfunction fromAudioAnalysis(\n analysis: AudioAnalysis,\n requestedNumberOfBars: number,\n silenceRmsThreshold: number\n): PreviewBarsResult {\n const barDurationMs =\n analysis.segmentDurationMs ||\n Math.max(\n 1,\n analysis.durationMs / Math.max(1, analysis.dataPoints.length)\n )\n const bars = analysis.dataPoints.map((point, index) =>\n pointToPreviewBar(point, index, barDurationMs, silenceRmsThreshold)\n )\n\n return {\n bars,\n durationMs: analysis.durationMs,\n sampleRate: analysis.sampleRate,\n numberOfChannels: analysis.numberOfChannels,\n bitDepth: analysis.bitDepth,\n samples: analysis.samples,\n requestedNumberOfBars,\n barDurationMs,\n amplitudeRange: calculateRange(bars.map((bar) => bar.amplitude)),\n rmsRange: calculateRange(bars.map((bar) => bar.rms)),\n extractionTimeMs: analysis.extractionTimeMs,\n }\n}\n\nfunction emitBarsProgressively(\n bars: PreviewBar[],\n onBarReady: NonNullable<PreviewBarsOptions['onBarReady']>\n): void {\n const total = bars.length\n for (let index = 0; index < total; index++) {\n onBarReady(bars[index], index, total)\n }\n}\n\n/**\n * Extracts compact waveform preview bars for UI rendering.\n *\n * Native platforms may provide a compact `extractPreviewBars` bridge. Until that\n * bridge is available, this safely falls back to the existing `extractPreview`\n * compatibility path and adapts `DataPoint` objects into compact bars.\n *\n * @throws {AudioExtractionError} when the underlying extraction fails.\n */\nexport async function extractPreviewBars({\n fileUri,\n numberOfBars = DEFAULT_PREVIEW_BARS,\n startTimeMs = 0,\n endTimeMs = DEFAULT_PREVIEW_END_TIME_MS,\n decodingOptions,\n logger,\n onBarReady,\n}: PreviewBarsOptions): Promise<PreviewBarsResult> {\n const requestedNumberOfBars = Math.max(1, Math.floor(numberOfBars))\n const nativeOptions = {\n fileUri,\n numberOfBars: requestedNumberOfBars,\n startTimeMs,\n endTimeMs,\n decodingOptions,\n }\n\n const nativeModule = AudioStudioModule as unknown\n if (hasNativePreviewBars(nativeModule)) {\n let result: PreviewBarsResult\n try {\n result = await nativeModule.extractPreviewBars(\n cleanNativeOptions(nativeOptions)\n )\n } catch (err) {\n throw mapExtractionError(err, fileUri)\n }\n\n if (onBarReady) {\n emitBarsProgressively(result.bars, onBarReady)\n }\n return result\n }\n\n let analysis: AudioAnalysis\n try {\n analysis = await extractPreview({\n fileUri,\n numberOfPoints: requestedNumberOfBars,\n startTimeMs,\n endTimeMs,\n decodingOptions,\n logger,\n })\n } catch (err) {\n throw mapExtractionError(err, fileUri)\n }\n\n const result = fromAudioAnalysis(\n analysis,\n requestedNumberOfBars,\n decodingOptions?.silenceRmsThreshold ?? DEFAULT_SILENCE_THRESHOLD\n )\n\n if (onBarReady) {\n emitBarsProgressively(result.bars, onBarReady)\n }\n\n return result\n}\n"]}