@siteed/audio-studio 3.2.1-beta.1 → 3.2.1-beta.3

package/build/cjs/useAudioRecorder.js.map

@@ -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    MaxDurationReachedEvent,\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    addMaxDurationReachedListener,\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    maxDurationMs?: number\n    maxDurationReached?: boolean\n}\n\ninterface RecorderReducerState {\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number\n    size: number\n    compression?: CompressionInfo\n    analysisData?: AudioAnalysis\n    maxDurationMs?: number\n    maxDurationReached?: boolean\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              maxDurationMs?: number\n              maxDurationReached?: boolean\n          }\n      }\n    | {\n          type: 'MAX_DURATION_REACHED'\n          payload: MaxDurationReachedEvent\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                maxDurationMs: undefined,\n                maxDurationReached: false,\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                // Preserve max-duration state after stop so UI and agentic\n                // validation can explain why recording ended. START resets it.\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                maxDurationMs: action.payload.maxDurationMs,\n                maxDurationReached: action.payload.maxDurationReached,\n            }\n            return newState\n        }\n        case 'MAX_DURATION_REACHED':\n            return {\n                ...state,\n                maxDurationMs: action.payload.maxDurationMs,\n                maxDurationReached: true,\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        maxDurationMs: undefined,\n        maxDurationReached: false,\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        maxDurationMs: undefined as number | undefined,\n        maxDurationReached: false,\n    })\n\n    const recordingConfigRef = useRef<RecordingConfig | null>(null)\n    const maxDurationHandledRef = useRef(false)\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=${savedAnalysisData.dataPoints.length}`\n            )\n\n            // Combine data points\n            const combinedDataPoints = [\n                ...savedAnalysisData.dataPoints,\n                ...analysis.dataPoints,\n            ]\n\n            const keepFullAnalysis = shouldKeepFullAnalysis(\n                recordingConfigRef.current\n            )\n            const fullCombinedDataPoints = keepFullAnalysis\n                ? [\n                      ...(fullAnalysisRef.current?.dataPoints ?? []),\n                      ...analysis.dataPoints,\n                  ]\n                : undefined\n\n            // Calculate the new duration\n            // The number of segments is based on how many segments of segmentDurationMs can fit in visualizationDuration\n            const numberOfSegments = Math.ceil(\n                visualizationDuration / analysis.segmentDurationMs\n            )\n            // maxDataPoints should be the number of data points, not milliseconds\n            const maxDataPoints = numberOfSegments\n\n            logger?.debug(\n                `[handleAudioAnalysis] Combined data points before trimming: numberOfSegments=${numberOfSegments} visualizationDuration=${visualizationDuration} combinedDataPointsLength=${combinedDataPoints.length} vs maxDataPoints=${maxDataPoints}`\n            )\n\n            // Trim data points to keep within the maximum number of data points\n            if (combinedDataPoints.length > maxDataPoints) {\n                combinedDataPoints.splice(\n                    0,\n                    combinedDataPoints.length - maxDataPoints\n                )\n            }\n\n            // Keep the full data points when requested for stopRecording().analysisData.\n            if (keepFullAnalysis && fullCombinedDataPoints) {\n                fullAnalysisRef.current = {\n                    ...fullAnalysisRef.current,\n                    dataPoints: fullCombinedDataPoints,\n                }\n                fullAnalysisRef.current.durationMs =\n                    fullCombinedDataPoints.length * analysis.segmentDurationMs\n            }\n            savedAnalysisData.dataPoints = combinedDataPoints\n            savedAnalysisData.bitDepth =\n                analysis.bitDepth || savedAnalysisData.bitDepth\n            savedAnalysisData.durationMs =\n                combinedDataPoints.length * analysis.segmentDurationMs\n\n            // Update amplitude range\n            const newMin = Math.min(\n                savedAnalysisData.amplitudeRange.min,\n                analysis.amplitudeRange.min\n            )\n            const newMax = Math.max(\n                savedAnalysisData.amplitudeRange.max,\n                analysis.amplitudeRange.max\n            )\n\n            savedAnalysisData.amplitudeRange = {\n                min: newMin,\n                max: newMax,\n            }\n            if (keepFullAnalysis) {\n                fullAnalysisRef.current.amplitudeRange = {\n                    min: newMin,\n                    max: newMax,\n                }\n            }\n\n            logger?.debug(\n                `[handleAudioAnalysis] Updated analysis data: durationMs=${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 handleMaxDurationReached = useCallback(\n        async (event: MaxDurationReachedEvent) => {\n            if (maxDurationHandledRef.current) {\n                return\n            }\n\n            maxDurationHandledRef.current = true\n            const config = recordingConfigRef.current\n            const callbackEvent: MaxDurationReachedEvent = {\n                ...event,\n                autoStopped:\n                    event.autoStopped || !!config?.autoStopOnMaxDuration,\n            }\n\n            stateRef.current.maxDurationMs = callbackEvent.maxDurationMs\n            stateRef.current.maxDurationReached = true\n            dispatch({\n                type: 'MAX_DURATION_REACHED',\n                payload: callbackEvent,\n            })\n\n            try {\n                config?.onMaxDurationReached?.(callbackEvent)\n            } catch (error) {\n                logger?.error(`Error in max duration callback:`, error)\n            }\n\n            const finishStoppedState = () => {\n                if (analysisListenerRef.current) {\n                    analysisListenerRef.current.remove()\n                    analysisListenerRef.current = null\n                }\n                onAudioStreamRef.current = null\n                stateRef.current.isRecording = false\n                stateRef.current.isPaused = false\n                dispatch({ type: 'STOP' })\n            }\n\n            const waitForPlatformAutoStop = async () => {\n                const timeoutMs = 3000\n                const startedAt = Date.now()\n                let lastStatus: AudioStreamStatus | undefined\n\n                while (Date.now() - startedAt < timeoutMs) {\n                    await new Promise((resolve) => setTimeout(resolve, 50))\n                    try {\n                        const currentStatus: AudioStreamStatus =\n                            audioStudio.status()\n                        lastStatus = currentStatus\n                        if (\n                            !currentStatus.isRecording &&\n                            !currentStatus.isPaused\n                        ) {\n                            finishStoppedState()\n                            return\n                        }\n                    } catch (error) {\n                        logger?.warn(\n                            `Error checking status after max duration auto-stop:`,\n                            error\n                        )\n                        break\n                    }\n                }\n\n                if (\n                    lastStatus &&\n                    (lastStatus.isRecording || lastStatus.isPaused)\n                ) {\n                    try {\n                        await audioStudio.stopRecording()\n                    } catch (error) {\n                        logger?.warn(\n                            `Error completing max duration auto-stop fallback:`,\n                            error\n                        )\n                    }\n                }\n                // At this point platform-owned auto-stop did not settle cleanly.\n                // Clear hook state so the UI does not stay stuck as recording.\n                finishStoppedState()\n            }\n\n            // Only the original event tells us whether the platform already\n            // owns auto-stop. Keep stream callbacks alive until status confirms\n            // stop completion so native final audio flushes can still reach JS.\n            if (event.autoStopped && stateRef.current.isRecording) {\n                await waitForPlatformAutoStop()\n                return\n            }\n\n            if (\n                config?.autoStopOnMaxDuration &&\n                !event.autoStopped &&\n                stateRef.current.isRecording\n            ) {\n                try {\n                    await audioStudio.stopRecording()\n                    finishStoppedState()\n                } catch (error) {\n                    logger?.error(`Error auto-stopping on max duration:`, error)\n                }\n            }\n        },\n        [audioStudio, dispatch, logger]\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            if (\n                status.maxDurationReached === true &&\n                status.maxDurationMs != null &&\n                !stateRef.current.maxDurationReached\n            ) {\n                await handleMaxDurationReached({\n                    durationMs: status.durationMs,\n                    maxDurationMs: status.maxDurationMs,\n                    overrunMs: Math.max(\n                        0,\n                        status.durationMs - status.maxDurationMs\n                    ),\n                    autoStopped: false,\n                })\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                status.maxDurationMs !== stateRef.current.maxDurationMs ||\n                status.maxDurationReached !==\n                    stateRef.current.maxDurationReached\n            ) {\n                stateRef.current.durationMs = status.durationMs\n                stateRef.current.size = status.size\n                stateRef.current.compression = status.compression\n                stateRef.current.maxDurationMs = status.maxDurationMs\n                stateRef.current.maxDurationReached =\n                    status.maxDurationReached ?? false\n                dispatch({\n                    type: 'UPDATE_STATUS',\n                    payload: {\n                        durationMs: status.durationMs,\n                        size: status.size,\n                        compression: status.compression,\n                        maxDurationMs: status.maxDurationMs,\n                        maxDurationReached: status.maxDurationReached,\n                    },\n                })\n            }\n        } catch (error) {\n            logger?.error(`Error getting status:`, error)\n        }\n    }, [audioStudio, handleMaxDurationReached, 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            maxDurationMs: state.maxDurationMs,\n            maxDurationReached: state.maxDurationReached ?? false,\n        }\n    }, [\n        state.isRecording,\n        state.isPaused,\n        state.durationMs,\n        state.size,\n        state.compression,\n        state.maxDurationMs,\n        state.maxDurationReached,\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            maxDurationHandledRef.current = false\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                onMaxDurationReached,\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                onMaxDurationReached,\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        maxDurationHandledRef.current = false\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        const subscription = addMaxDurationReachedListener(async (event) => {\n            await handleMaxDurationReached(event)\n        })\n\n        return () => {\n            subscription.remove()\n        }\n    }, [handleMaxDurationReached])\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        maxDurationMs: state.maxDurationMs,\n        maxDurationReached: state.maxDurationReached,\n    }\n}\n"]}
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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    MaxDurationReachedEvent,\n    RecordingConfig,\n    RecordingStopReason,\n    StartRecordingResult,\n} from './AudioStudio.types'\nimport AudioStudioModule from './AudioStudioModule'\nimport { validateRecordingConfig } from './constants/platformLimitations'\nimport {\n    addAudioAnalysisListener,\n    addAudioEventListener,\n    AudioEventPayload,\n    addMaxDurationReachedListener,\n    addRecordingInterruptionListener,\n} from './events'\nimport { createNativeRecordingOptions } from './utils/nativeRecordingOptions'\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    maxDurationMs?: number\n    maxDurationReached?: boolean\n    lastRecordingReason?: RecordingStopReason\n}\n\ninterface RecorderReducerState {\n    isRecording: boolean\n    isPaused: boolean\n    durationMs: number\n    size: number\n    compression?: CompressionInfo\n    analysisData?: AudioAnalysis\n    maxDurationMs?: number\n    maxDurationReached?: boolean\n    lastRecordingReason?: RecordingStopReason\n}\n\ntype RecorderAction =\n    | { type: 'START' | 'PAUSE' | 'RESUME' }\n    | {\n          type: 'STOP'\n          payload: {\n              reason: RecordingStopReason\n          }\n      }\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              maxDurationMs?: number\n              maxDurationReached?: boolean\n          }\n      }\n    | {\n          type: 'MAX_DURATION_REACHED'\n          payload: MaxDurationReachedEvent\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 finiteOrZero(value: number): number {\n    return Number.isFinite(value) ? value : 0\n}\n\nfunction sanitizeSerializableValue<T>(value: T): T {\n    if (typeof value === 'number') {\n        return finiteOrZero(value) as T\n    }\n\n    if (Array.isArray(value)) {\n        return value.map((item) => sanitizeSerializableValue(item)) as T\n    }\n\n    if (value && typeof value === 'object') {\n        const sanitized: Record<string, unknown> = {}\n\n        for (const [key, nestedValue] of Object.entries(\n            value as Record<string, unknown>\n        )) {\n            sanitized[key] = sanitizeSerializableValue(nestedValue)\n        }\n\n        return sanitized as T\n    }\n\n    return value\n}\n\nfunction createSerializableAnalysis(analysis: AudioAnalysis): AudioAnalysis {\n    return sanitizeSerializableValue(analysis)\n}\n\nfunction createRecordingSnapshot(recording: AudioRecording): AudioRecording {\n    return sanitizeSerializableValue(recording)\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                maxDurationMs: undefined,\n                maxDurationReached: false,\n                lastRecordingReason: undefined,\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                lastRecordingReason: action.payload.reason,\n                // Preserve max-duration state after stop so UI and agentic\n                // validation can explain why recording ended. START resets it.\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                maxDurationMs: action.payload.maxDurationMs,\n                maxDurationReached: action.payload.maxDurationReached,\n            }\n            return newState\n        }\n        case 'MAX_DURATION_REACHED':\n            return {\n                ...state,\n                maxDurationMs: action.payload.maxDurationMs,\n                maxDurationReached: true,\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        maxDurationMs: undefined,\n        maxDurationReached: false,\n        lastRecordingReason: 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        maxDurationMs: undefined as number | undefined,\n        maxDurationReached: false,\n    })\n\n    const recordingConfigRef = useRef<RecordingConfig | null>(null)\n    const maxDurationHandledRef = useRef(false)\n    const stopFinalizationRef = useRef<Promise<AudioRecording> | 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                ? [\n                      ...(fullAnalysisRef.current?.dataPoints ?? []),\n                      ...analysis.dataPoints,\n                  ]\n                : undefined\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 when requested for stopRecording().analysisData.\n            if (keepFullAnalysis && 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 finalizeRecordingStop = useCallback(\n        async (reason: RecordingStopReason) => {\n            if (stopFinalizationRef.current) {\n                return stopFinalizationRef.current\n            }\n\n            const finalizePromise = (async () => {\n                const nativeStopResult: AudioRecording | null =\n                    await audioStudio.stopRecording()\n\n                if (!nativeStopResult) {\n                    throw new Error('Failed to stop recording')\n                }\n\n                const stopResult = createRecordingSnapshot(nativeStopResult)\n\n                if (shouldKeepFullAnalysis(recordingConfigRef.current)) {\n                    stopResult.analysisData = createSerializableAnalysis(\n                        fullAnalysisRef.current\n                    )\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                stateRef.current.isRecording = false\n                stateRef.current.isPaused = false\n\n                // Note: We deliberately DON'T clear recordingConfigRef here to preserve callbacks.\n                logger?.debug(`recording stopped`, stopResult)\n                maxDurationHandledRef.current = false\n                dispatch({\n                    type: 'STOP',\n                    payload: { reason },\n                })\n\n                const stoppedCallback =\n                    recordingConfigRef.current?.onRecordingStopped\n                if (stoppedCallback) {\n                    try {\n                        void Promise.resolve(\n                            stoppedCallback(stopResult, reason)\n                        ).catch((error) => {\n                            logger?.error(\n                                `Error in recording stopped callback:`,\n                                error\n                            )\n                        })\n                    } catch (error) {\n                        logger?.error(\n                            `Error in recording stopped callback:`,\n                            error\n                        )\n                    }\n                }\n\n                return stopResult\n            })()\n\n            stopFinalizationRef.current = finalizePromise\n            try {\n                return await finalizePromise\n            } finally {\n                stopFinalizationRef.current = null\n            }\n        },\n        [audioStudio, dispatch, logger]\n    )\n\n    const handleMaxDurationReached = useCallback(\n        async (event: MaxDurationReachedEvent) => {\n            if (maxDurationHandledRef.current) {\n                return\n            }\n\n            maxDurationHandledRef.current = true\n            const config = recordingConfigRef.current\n            const callbackEvent: MaxDurationReachedEvent = {\n                ...event,\n                autoStopped:\n                    event.autoStopped || !!config?.autoStopOnMaxDuration,\n            }\n\n            stateRef.current.maxDurationMs = callbackEvent.maxDurationMs\n            stateRef.current.maxDurationReached = true\n            dispatch({\n                type: 'MAX_DURATION_REACHED',\n                payload: callbackEvent,\n            })\n\n            try {\n                config?.onMaxDurationReached?.(callbackEvent)\n            } catch (error) {\n                logger?.error(`Error in max duration callback:`, error)\n            }\n\n            if (config?.autoStopOnMaxDuration && stateRef.current.isRecording) {\n                try {\n                    await finalizeRecordingStop('maxDuration')\n                } catch (error) {\n                    logger?.error(`Error auto-stopping on max duration:`, error)\n                }\n            }\n        },\n        [dispatch, finalizeRecordingStop, logger]\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            if (\n                status.maxDurationReached === true &&\n                status.maxDurationMs != null &&\n                !stateRef.current.maxDurationReached\n            ) {\n                await handleMaxDurationReached({\n                    durationMs: status.durationMs,\n                    maxDurationMs: status.maxDurationMs,\n                    overrunMs: Math.max(\n                        0,\n                        status.durationMs - status.maxDurationMs\n                    ),\n                    autoStopped: false,\n                })\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            const statusMaxDurationReached = status.maxDurationReached ?? false\n            const preserveStoppedMaxDuration =\n                !status.isRecording &&\n                !status.isPaused &&\n                stateRef.current.maxDurationReached &&\n                !statusMaxDurationReached\n            const nextMaxDurationMs = preserveStoppedMaxDuration\n                ? stateRef.current.maxDurationMs\n                : status.maxDurationMs\n            const nextMaxDurationReached = preserveStoppedMaxDuration\n                ? true\n                : statusMaxDurationReached\n\n            if (\n                status.durationMs !== stateRef.current.durationMs ||\n                status.size !== stateRef.current.size ||\n                nextMaxDurationMs !== stateRef.current.maxDurationMs ||\n                nextMaxDurationReached !== stateRef.current.maxDurationReached\n            ) {\n                stateRef.current.durationMs = status.durationMs\n                stateRef.current.size = status.size\n                stateRef.current.compression = status.compression\n                stateRef.current.maxDurationMs = nextMaxDurationMs\n                stateRef.current.maxDurationReached = nextMaxDurationReached\n                dispatch({\n                    type: 'UPDATE_STATUS',\n                    payload: {\n                        durationMs: status.durationMs,\n                        size: status.size,\n                        compression: status.compression,\n                        maxDurationMs: nextMaxDurationMs,\n                        maxDurationReached: nextMaxDurationReached,\n                    },\n                })\n            }\n        } catch (error) {\n            logger?.error(`Error getting status:`, error)\n        }\n    }, [audioStudio, handleMaxDurationReached, 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            maxDurationMs: state.maxDurationMs,\n            maxDurationReached: state.maxDurationReached ?? false,\n        }\n    }, [\n        state.isRecording,\n        state.isPaused,\n        state.durationMs,\n        state.size,\n        state.compression,\n        state.maxDurationMs,\n        state.maxDurationReached,\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            maxDurationHandledRef.current = false\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 { onAudioStream, enableProcessing } = validatedOptions\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 hook-only values and undefineds that can't cross the native bridge.\n            // autoStopOnMaxDuration stays hook-owned so finalization can expose\n            // the same AudioRecording result as a manual stop.\n            const cleanOptions = createNativeRecordingOptions(validatedOptions)\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 { onAudioStream } = 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 hook-only values and undefineds that can't cross the native bridge.\n            const cleanOptions = createNativeRecordingOptions(recordingOptions)\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        return finalizeRecordingStop('manual')\n    }, [finalizeRecordingStop, logger])\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        const subscription = addMaxDurationReachedListener(async (event) => {\n            await handleMaxDurationReached(event)\n        })\n\n        return () => {\n            subscription.remove()\n        }\n    }, [handleMaxDurationReached])\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        maxDurationMs: state.maxDurationMs,\n        maxDurationReached: state.maxDurationReached,\n        lastRecordingReason: state.lastRecordingReason,\n    }\n}\n"]}

package/build/cjs/utils/nativeRecordingOptions.js

@@ -0,0 +1,13 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.createNativeRecordingOptions = createNativeRecordingOptions;
+const cleanNativeOptions_1 = require("./cleanNativeOptions");
+function createNativeRecordingOptions(recordingOptions) {
+    const { onAudioStream, onRecordingInterrupted, onMaxDurationReached, onRecordingStopped, onAudioAnalysis, keepFullAnalysis: _keepFullAnalysis, 
+    // Keep hook-owned auto-stop out of the native bridge so the hook can
+    // reuse the same finalization path as manual stop and expose the
+    // resulting AudioRecording consistently.
+    autoStopOnMaxDuration: _autoStopOnMaxDuration, ...options } = recordingOptions;
+    return (0, cleanNativeOptions_1.cleanNativeOptions)(options);
+}
+//# sourceMappingURL=nativeRecordingOptions.js.map

package/build/cjs/utils/nativeRecordingOptions.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"nativeRecordingOptions.js","sourceRoot":"","sources":["../../../src/utils/nativeRecordingOptions.ts"],"names":[],"mappings":";;AAGA,oEAgBC;AAlBD,6DAAyD;AAEzD,SAAgB,4BAA4B,CAAC,gBAAiC;IAC1E,MAAM,EACF,aAAa,EACb,sBAAsB,EACtB,oBAAoB,EACpB,kBAAkB,EAClB,eAAe,EACf,gBAAgB,EAAE,iBAAiB;IACnC,qEAAqE;IACrE,iEAAiE;IACjE,yCAAyC;IACzC,qBAAqB,EAAE,sBAAsB,EAC7C,GAAG,OAAO,EACb,GAAG,gBAAgB,CAAA;IAEpB,OAAO,IAAA,uCAAkB,EAAC,OAAO,CAAC,CAAA;AACtC,CAAC","sourcesContent":["import { RecordingConfig } from '../AudioStudio.types'\nimport { cleanNativeOptions } from './cleanNativeOptions'\n\nexport function createNativeRecordingOptions(recordingOptions: RecordingConfig) {\n    const {\n        onAudioStream,\n        onRecordingInterrupted,\n        onMaxDurationReached,\n        onRecordingStopped,\n        onAudioAnalysis,\n        keepFullAnalysis: _keepFullAnalysis,\n        // Keep hook-owned auto-stop out of the native bridge so the hook can\n        // reuse the same finalization path as manual stop and expose the\n        // resulting AudioRecording consistently.\n        autoStopOnMaxDuration: _autoStopOnMaxDuration,\n        ...options\n    } = recordingOptions\n\n    return cleanNativeOptions(options)\n}\n"]}

package/build/cjs/utils/nativeRecordingOptions.test.js

@@ -0,0 +1,30 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+const nativeRecordingOptions_1 = require("./nativeRecordingOptions");
+describe('createNativeRecordingOptions', () => {
+    it('keeps maxDurationMs native but leaves auto-stop finalization hook-owned', () => {
+        const nativeOptions = (0, nativeRecordingOptions_1.createNativeRecordingOptions)({
+            maxDurationMs: 1500,
+            autoStopOnMaxDuration: true,
+            onMaxDurationReached: jest.fn(),
+            onRecordingStopped: jest.fn(),
+            onRecordingInterrupted: jest.fn(),
+            onAudioAnalysis: jest.fn(),
+            onAudioStream: jest.fn(),
+            keepFullAnalysis: true,
+            sampleRate: 16000,
+        });
+        expect(nativeOptions).toMatchObject({
+            maxDurationMs: 1500,
+            sampleRate: 16000,
+        });
+        expect(nativeOptions).not.toHaveProperty('autoStopOnMaxDuration');
+        expect(nativeOptions).not.toHaveProperty('onMaxDurationReached');
+        expect(nativeOptions).not.toHaveProperty('onRecordingStopped');
+        expect(nativeOptions).not.toHaveProperty('onRecordingInterrupted');
+        expect(nativeOptions).not.toHaveProperty('onAudioAnalysis');
+        expect(nativeOptions).not.toHaveProperty('onAudioStream');
+        expect(nativeOptions).not.toHaveProperty('keepFullAnalysis');
+    });
+});
+//# sourceMappingURL=nativeRecordingOptions.test.js.map

package/build/cjs/utils/nativeRecordingOptions.test.js.map

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

@@ -1 +1 @@
-{"version":3,"file":"AudioStudio.types.js","sourceRoot":"","sources":["../../src/AudioStudio.types.ts"],"names":[],"mappings":"AA+VA,4EAA4E;AAC5E,MAAM,CAAC,MAAM,2BAA2B,GAAG;IACvC,8CAA8C;IAC9C,KAAK,EAAE,OAAO;IACd,sDAAsD;IACtD,QAAQ,EAAE,UAAU;CACd,CAAA","sourcesContent":["// packages/audio-studio/src/AudioStudio.types.ts\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DecodingConfig,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport type { 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    /** Configured maximum active recording duration in milliseconds, if enabled */\n    maxDurationMs?: number\n    /** Whether the current recording session has reached the configured maximum duration */\n    maxDurationReached?: boolean\n}\n\ninterface AudioDataEventBase {\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 interface AudioDataEventRaw extends AudioDataEventBase {\n    /** Audio data as base64 string (native), Float32Array (web), or Int16Array (web) */\n    data: string | Float32Array | Int16Array\n    streamFormat?: undefined | 'raw'\n}\n\nexport interface AudioDataEventFloat32 extends AudioDataEventBase {\n    /** Audio data as Float32Array with samples in [-1, 1] range */\n    data: Float32Array\n    streamFormat: 'float32'\n}\n\nexport type AudioDataEvent = AudioDataEventRaw | AudioDataEventFloat32\n\n/**\n * Audio encoding types supported by the library.\n *\n * Platform support:\n * - `pcm_8bit`: Android only (iOS/Web will fallback to 16-bit)\n * - `pcm_16bit`: All platforms\n * - `pcm_32bit`: All platforms\n *\n * @see {@link https://github.com/deeeed/audiolab/blob/main/packages/audio-studio/docs/PLATFORM_LIMITATIONS.md | Platform Limitations}\n */\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\n\n/**\n * Supported audio sample rates in Hz.\n * All platforms support these standard rates.\n */\nexport type SampleRate = 16000 | 44100 | 48000\n\n/**\n * Audio bit depth (bits per sample).\n *\n * Platform support:\n * - `8`: Android only (iOS/Web will fallback to 16)\n * - `16`: All platforms (recommended for compatibility)\n * - `32`: All platforms\n *\n * @see {@link https://github.com/deeeed/audiolab/blob/main/packages/audio-studio/docs/PLATFORM_LIMITATIONS.md | Platform Limitations}\n */\nexport type BitDepth = 8 | 16 | 32\n\n/**\n * PCM format string representation.\n * @deprecated Use `EncodingType` directly\n */\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    /**\n     * Full analysis data for the recording if processing was enabled and\n     * `keepFullAnalysis` was not set to `false`.\n     */\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 MaxDurationReachedEvent {\n    /** Active recording duration that triggered the event, in milliseconds */\n    durationMs: number\n    /** Configured active recording duration limit, in milliseconds */\n    maxDurationMs: number\n    /** Amount by which timer delivery exceeded the limit, in milliseconds */\n    overrunMs: number\n    /** Active stream identifier when available */\n    streamUuid?: string\n    /** Whether the recorder was configured to stop automatically after this event */\n    autoStopped: boolean\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/** Android platform specific configuration options */\nexport interface AndroidConfig {\n    /**\n     * Audio focus strategy for handling interruptions and background behavior\n     *\n     * - `'background'`: Continue recording when app loses focus (voice recorders, transcription apps)\n     * - `'interactive'`: Pause when losing focus, resume when gaining (music apps, games)\n     * - `'communication'`: Maintain priority for real-time communication (video calls, voice chat)\n     * - `'none'`: No automatic audio focus management (custom handling)\n     *\n     * @default 'background' when keepAwake=true, 'interactive' otherwise\n     */\n    audioFocusStrategy?: 'background' | 'interactive' | 'communication' | 'none'\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         * Prefer raw stream over container format (Android only)\n         * - true: Use raw AAC stream (.aac files) like in v2.10.6\n         * - false/undefined: Use M4A container (.m4a files) for better seeking support\n         * Note: iOS always produces M4A containers and ignores this flag\n         */\n        preferRawStream?: boolean\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    /**\n     * Encoding type for the recording.\n     *\n     * Platform limitations:\n     * - `pcm_8bit`: Android only (iOS/Web will fallback to `pcm_16bit` with warning)\n     * - `pcm_16bit`: All platforms (recommended for cross-platform compatibility)\n     * - `pcm_32bit`: All platforms\n     *\n     * The library will automatically validate and adjust the encoding based on\n     * platform capabilities. A warning will be logged if fallback is required.\n     *\n     * @default 'pcm_16bit'\n     * @see {@link EncodingType}\n     * @see {@link https://github.com/deeeed/audiolab/blob/main/packages/audio-studio/docs/PLATFORM_LIMITATIONS.md | Platform Limitations}\n     */\n    encoding?: EncodingType\n\n    /** Interval in milliseconds at which to emit recording data (minimum: 10ms) */\n    interval?: number\n\n    /** Interval in milliseconds at which to emit analysis data (minimum: 10ms) */\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    /**\n     * Whether `useAudioRecorder` should retain every audio-analysis data point\n     * and attach the full history to `stopRecording().analysisData`.\n     *\n     * Defaults to `true` for backwards compatibility. Set to `false` for\n     * long-running recordings when you only need live `analysisData` state or\n     * per-callback `onAudioAnalysis` chunks; this avoids unbounded JS memory\n     * growth in the hook without disabling native analysis processing.\n     */\n    keepFullAnalysis?: boolean\n\n    /** iOS-specific configuration */\n    ios?: IOSConfig\n\n    /** Android-specific configuration */\n    android?: AndroidConfig\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    /**\n     * Maximum cumulative active recording duration, in milliseconds.\n     *\n     * Paused time does not count. Set to undefined, 0, or a negative value to disable.\n     */\n    maxDurationMs?: number\n\n    /**\n     * Stop recording automatically when maxDurationMs is reached.\n     *\n     * Defaults to false. The MaxDurationReached event is emitted before the stop request.\n     * The automatic stop result is not returned to onMaxDurationReached; use the\n     * event and stream callbacks for immediate UI updates.\n     */\n    autoStopOnMaxDuration?: boolean\n\n    /**\n     * Optional callback invoked when maxDurationMs is reached.\n     *\n     * If autoStopOnMaxDuration is true, this callback is invoked before the\n     * recorder finishes stopping. The final stop result is not passed here.\n     */\n    onMaxDurationReached?: (_: MaxDurationReachedEvent) => 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    /**\n     * Format for the audio stream data delivered to `onAudioStream`.\n     *\n     * - `'raw'` (default): base64-encoded PCM bytes on native, Float32Array on web\n     * - `'float32'`: Float32Array with samples in [-1, 1] on all platforms.\n     *   Eliminates base64 encode/decode overhead on the native bridge.\n     *   Android (new arch): delivered as Float32Array via JSI.\n     *   iOS: delivered as regular Array<number>, normalized to Float32Array in JS.\n     *\n     * @default 'raw'\n     */\n    streamFormat?: 'float32' | 'raw'\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        /** Whether to show pause/resume actions in the notification (default: true) */\n        showPauseResumeActions?: boolean\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    /** Configured maximum active recording duration in milliseconds, if enabled */\n    maxDurationMs?: number\n    /** Whether the current recording session has reached the configured maximum duration */\n    maxDurationReached?: boolean\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    /** Optional callback invoked when maxDurationMs is reached */\n    onMaxDurationReached?: (_: MaxDurationReachedEvent) => 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"]}
+{"version":3,"file":"AudioStudio.types.js","sourceRoot":"","sources":["../../src/AudioStudio.types.ts"],"names":[],"mappings":"AAiWA,4EAA4E;AAC5E,MAAM,CAAC,MAAM,2BAA2B,GAAG;IACvC,8CAA8C;IAC9C,KAAK,EAAE,OAAO;IACd,sDAAsD;IACtD,QAAQ,EAAE,UAAU;CACd,CAAA","sourcesContent":["// packages/audio-studio/src/AudioStudio.types.ts\nimport {\n    AudioAnalysis,\n    AudioFeaturesOptions,\n    DecodingConfig,\n} from './AudioAnalysis/AudioAnalysis.types'\nimport type { 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    /** Configured maximum active recording duration in milliseconds, if enabled */\n    maxDurationMs?: number\n    /** Whether the current recording session has reached the configured maximum duration */\n    maxDurationReached?: boolean\n}\n\ninterface AudioDataEventBase {\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 interface AudioDataEventRaw extends AudioDataEventBase {\n    /** Audio data as base64 string (native), Float32Array (web), or Int16Array (web) */\n    data: string | Float32Array | Int16Array\n    streamFormat?: undefined | 'raw'\n}\n\nexport interface AudioDataEventFloat32 extends AudioDataEventBase {\n    /** Audio data as Float32Array with samples in [-1, 1] range */\n    data: Float32Array\n    streamFormat: 'float32'\n}\n\nexport type AudioDataEvent = AudioDataEventRaw | AudioDataEventFloat32\n\n/**\n * Audio encoding types supported by the library.\n *\n * Platform support:\n * - `pcm_8bit`: Android only (iOS/Web will fallback to 16-bit)\n * - `pcm_16bit`: All platforms\n * - `pcm_32bit`: All platforms\n *\n * @see {@link https://github.com/deeeed/audiolab/blob/main/packages/audio-studio/docs/PLATFORM_LIMITATIONS.md | Platform Limitations}\n */\nexport type EncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit'\n\n/**\n * Supported audio sample rates in Hz.\n * All platforms support these standard rates.\n */\nexport type SampleRate = 16000 | 44100 | 48000\n\n/**\n * Audio bit depth (bits per sample).\n *\n * Platform support:\n * - `8`: Android only (iOS/Web will fallback to 16)\n * - `16`: All platforms (recommended for compatibility)\n * - `32`: All platforms\n *\n * @see {@link https://github.com/deeeed/audiolab/blob/main/packages/audio-studio/docs/PLATFORM_LIMITATIONS.md | Platform Limitations}\n */\nexport type BitDepth = 8 | 16 | 32\n\n/**\n * PCM format string representation.\n * @deprecated Use `EncodingType` directly\n */\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    /**\n     * Full analysis data for the recording if processing was enabled and\n     * `keepFullAnalysis` was not set to `false`.\n     */\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 MaxDurationReachedEvent {\n    /** Active recording duration that triggered the event, in milliseconds */\n    durationMs: number\n    /** Configured active recording duration limit, in milliseconds */\n    maxDurationMs: number\n    /** Amount by which timer delivery exceeded the limit, in milliseconds */\n    overrunMs: number\n    /** Active stream identifier when available */\n    streamUuid?: string\n    /** Whether the recorder was configured to stop automatically after this event */\n    autoStopped: boolean\n}\n\nexport type RecordingStopReason = 'manual' | 'maxDuration'\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/** Android platform specific configuration options */\nexport interface AndroidConfig {\n    /**\n     * Audio focus strategy for handling interruptions and background behavior\n     *\n     * - `'background'`: Continue recording when app loses focus (voice recorders, transcription apps)\n     * - `'interactive'`: Pause when losing focus, resume when gaining (music apps, games)\n     * - `'communication'`: Maintain priority for real-time communication (video calls, voice chat)\n     * - `'none'`: No automatic audio focus management (custom handling)\n     *\n     * @default 'background' when keepAwake=true, 'interactive' otherwise\n     */\n    audioFocusStrategy?: 'background' | 'interactive' | 'communication' | 'none'\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         * Prefer raw stream over container format (Android only)\n         * - true: Use raw AAC stream (.aac files) like in v2.10.6\n         * - false/undefined: Use M4A container (.m4a files) for better seeking support\n         * Note: iOS always produces M4A containers and ignores this flag\n         */\n        preferRawStream?: boolean\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    /**\n     * Encoding type for the recording.\n     *\n     * Platform limitations:\n     * - `pcm_8bit`: Android only (iOS/Web will fallback to `pcm_16bit` with warning)\n     * - `pcm_16bit`: All platforms (recommended for cross-platform compatibility)\n     * - `pcm_32bit`: All platforms\n     *\n     * The library will automatically validate and adjust the encoding based on\n     * platform capabilities. A warning will be logged if fallback is required.\n     *\n     * @default 'pcm_16bit'\n     * @see {@link EncodingType}\n     * @see {@link https://github.com/deeeed/audiolab/blob/main/packages/audio-studio/docs/PLATFORM_LIMITATIONS.md | Platform Limitations}\n     */\n    encoding?: EncodingType\n\n    /** Interval in milliseconds at which to emit recording data (minimum: 10ms) */\n    interval?: number\n\n    /** Interval in milliseconds at which to emit analysis data (minimum: 10ms) */\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    /**\n     * Whether `useAudioRecorder` should retain every audio-analysis data point\n     * and attach the full history to `stopRecording().analysisData`.\n     *\n     * Defaults to `true` for backwards compatibility. Set to `false` for\n     * long-running recordings when you only need live `analysisData` state or\n     * per-callback `onAudioAnalysis` chunks; this avoids unbounded JS memory\n     * growth in the hook without disabling native analysis processing.\n     */\n    keepFullAnalysis?: boolean\n\n    /** iOS-specific configuration */\n    ios?: IOSConfig\n\n    /** Android-specific configuration */\n    android?: AndroidConfig\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    /**\n     * Maximum cumulative active recording duration, in milliseconds.\n     *\n     * Paused time does not count. Set to undefined, 0, or a negative value to disable.\n     */\n    maxDurationMs?: number\n\n    /**\n     * Stop recording automatically when maxDurationMs is reached.\n     *\n     * Defaults to false. When used with `useAudioRecorder`, the\n     * MaxDurationReached event is emitted immediately, then the hook stops the\n     * recorder and exposes the final result through `onRecordingStopped`.\n     */\n    autoStopOnMaxDuration?: boolean\n\n    /**\n     * Optional callback invoked when maxDurationMs is reached.\n     *\n     * This remains an immediate threshold callback. If\n     * autoStopOnMaxDuration is true, use `onRecordingStopped` for the full\n     * recording result after stop completes.\n     */\n    onMaxDurationReached?: (_: MaxDurationReachedEvent) => void\n\n    /**\n     * Optional callback invoked after a recording has fully stopped and the\n     * final `AudioRecording` result is available.\n     *\n     * The reason is `manual` when stopped through `stopRecording()` and\n     * `maxDuration` when stopped by `autoStopOnMaxDuration`.\n     */\n    onRecordingStopped?: (\n        recording: AudioRecording,\n        reason: RecordingStopReason\n    ) => void | Promise<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    /**\n     * Format for the audio stream data delivered to `onAudioStream`.\n     *\n     * - `'raw'` (default): base64-encoded PCM bytes on native, Float32Array on web\n     * - `'float32'`: Float32Array with samples in [-1, 1] on all platforms.\n     *   Eliminates base64 encode/decode overhead on the native bridge.\n     *   Android (new arch): delivered as Float32Array via JSI.\n     *   iOS: delivered as regular Array<number>, normalized to Float32Array in JS.\n     *\n     * @default 'raw'\n     */\n    streamFormat?: 'float32' | 'raw'\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        /** Whether to show pause/resume actions in the notification (default: true) */\n        showPauseResumeActions?: boolean\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    /** Configured maximum active recording duration in milliseconds, if enabled */\n    maxDurationMs?: number\n    /** Whether the current recording session has reached the configured maximum duration */\n    maxDurationReached?: boolean\n    /** Analysis data for the recording if processing was enabled */\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    /** Reason associated with the last completed recording */\n    lastRecordingReason?: RecordingStopReason\n    /** Optional callback to handle recording interruptions */\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n    /** Optional callback invoked when maxDurationMs is reached */\n    onMaxDurationReached?: (_: MaxDurationReachedEvent) => void\n    /** Optional callback invoked when a recording fully stops */\n    onRecordingStopped?: (\n        recording: AudioRecording,\n        reason: RecordingStopReason\n    ) => void | Promise<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"]}