@siteed/audio-studio 3.2.1-beta.0 → 3.2.1-beta.1

package/README.md

@@ -36,7 +36,7 @@ Cross-platform audio recording, analysis, and processing for React Native and Ex
 
 ## Features
 
-- **Recording** — real-time streaming, dual-stream (raw PCM + compressed), background recording, zero-latency start via `prepareRecording`
+- **Recording** — real-time streaming, dual-stream (raw PCM + compressed), max active duration limits, background recording, zero-latency start via `prepareRecording`
 - **Device management** — list/select input devices (Bluetooth, USB, wired), automatic fallback
 - **Interruption handling** — auto pause/resume during phone calls
 - **Audio analysis** — MFCC, spectral features, mel spectrogram, tempo, pitch, waveform preview
@@ -116,6 +116,35 @@ await startRecording({
 })
 ```
 
+### Max Active Recording Duration
+
+Use `maxDurationMs` to cap cumulative active recording time. Paused time does
+not count toward the limit. By default, the recorder emits
+`onMaxDurationReached` and continues recording so your app can decide what to
+do next. Set `autoStopOnMaxDuration: true` to stop automatically.
+
+```typescript
+const {
+    startRecording,
+    maxDurationMs,
+    maxDurationReached,
+} = useAudioRecorder()
+
+await startRecording({
+    sampleRate: 16000,
+    channels: 1,
+    maxDurationMs: 60_000,
+    autoStopOnMaxDuration: true,
+    onMaxDurationReached: (event) => {
+        console.log('Reached recording limit:', event.maxDurationMs)
+    },
+})
+```
+
+When auto-stop is enabled, the max-duration event is emitted before the stop
+finishes. Use the event and stream callbacks for immediate UI updates, then use
+normal recording state to observe that the recorder has stopped.
+
 ## Audio Analysis
 
 For live analysis during recording, `useAudioRecorder` keeps a recent analysis

package/android/src/main/java/net/siteed/audiostudio/AudioRecorderManager.kt

@@ -92,6 +92,12 @@ class AudioRecorderManager(
     private var lastEmitTime = SystemClock.elapsedRealtime()
     private var lastPauseTime = 0L
     private var pausedDuration = 0L
+    private val maxDurationLock = Any()
+    private var maxDurationRunnable: Runnable? = null
+    private var maxDurationTargetMs = 0L
+    private var maxDurationAccumulatedActiveMs = 0L
+    private var maxDurationSegmentStartElapsed = 0L
+    private var maxDurationReached = false
     private var lastEmittedSize = 0L
     private var lastEmittedCompressedSize = 0L
     private var streamPosition = 0L  // Track total bytes processed in the stream
@@ -650,6 +656,7 @@ class AudioRecorderManager(
                     "compressedFileUri" to compressedFile?.toURI().toString()
                 ) else null
             )
+            startMaxDurationTimer()
             promise.resolve(result)
 
         } catch (e: Exception) {
@@ -695,6 +702,123 @@ class AudioRecorderManager(
         return isSupported
     }
 
+    private fun getMaxDurationActiveMs(now: Long = SystemClock.elapsedRealtime()): Long {
+        return synchronized(maxDurationLock) {
+            if (maxDurationSegmentStartElapsed <= 0L) {
+                maxDurationAccumulatedActiveMs
+            } else {
+                maxDurationAccumulatedActiveMs + (now - maxDurationSegmentStartElapsed)
+            }
+        }
+    }
+
+    private fun startMaxDurationTimer() {
+        synchronized(maxDurationLock) {
+            maxDurationRunnable?.let { mainHandler.removeCallbacks(it) }
+            maxDurationRunnable = null
+            maxDurationTargetMs = recordingConfig.maxDurationMs
+            maxDurationAccumulatedActiveMs = 0L
+            maxDurationSegmentStartElapsed = 0L
+            maxDurationReached = false
+
+            if (maxDurationTargetMs <= 0L) {
+                return
+            }
+
+            maxDurationSegmentStartElapsed = SystemClock.elapsedRealtime()
+        }
+        scheduleMaxDurationTimer()
+    }
+
+    private fun scheduleMaxDurationTimer() {
+        val remainingMs = synchronized(maxDurationLock) {
+            if (
+                maxDurationTargetMs <= 0L ||
+                maxDurationReached ||
+                !_isRecording.get() ||
+                isPaused.get()
+            ) {
+                return
+            }
+
+            maxDurationRunnable?.let { mainHandler.removeCallbacks(it) }
+            (maxDurationTargetMs - getMaxDurationActiveMs()).coerceAtLeast(0L)
+        }
+
+        val runnable = Runnable { emitMaxDurationReached() }
+        synchronized(maxDurationLock) {
+            maxDurationRunnable = runnable
+        }
+        mainHandler.postDelayed(runnable, remainingMs)
+    }
+
+    private fun pauseMaxDurationTimer() {
+        synchronized(maxDurationLock) {
+            maxDurationRunnable?.let { mainHandler.removeCallbacks(it) }
+            maxDurationRunnable = null
+            if (maxDurationSegmentStartElapsed > 0L) {
+                maxDurationAccumulatedActiveMs = getMaxDurationActiveMs()
+                maxDurationSegmentStartElapsed = 0L
+            }
+        }
+    }
+
+    private fun resumeMaxDurationTimer() {
+        synchronized(maxDurationLock) {
+            if (maxDurationTargetMs <= 0L || maxDurationReached) {
+                return
+            }
+            maxDurationSegmentStartElapsed = SystemClock.elapsedRealtime()
+        }
+        scheduleMaxDurationTimer()
+    }
+
+    private fun cancelMaxDurationTimer() {
+        synchronized(maxDurationLock) {
+            maxDurationRunnable?.let { mainHandler.removeCallbacks(it) }
+            maxDurationRunnable = null
+            maxDurationSegmentStartElapsed = 0L
+            if (!maxDurationReached) {
+                maxDurationTargetMs = 0L
+                maxDurationAccumulatedActiveMs = 0L
+            }
+        }
+    }
+
+    private fun emitMaxDurationReached() {
+        val event = synchronized(maxDurationLock) {
+            if (maxDurationTargetMs <= 0L || maxDurationReached) {
+                return
+            }
+            if (!_isRecording.get() || isPaused.get()) {
+                return
+            }
+
+            val durationMs = getMaxDurationActiveMs()
+            maxDurationReached = true
+            maxDurationRunnable = null
+            bundleOf(
+                "durationMs" to durationMs,
+                "maxDurationMs" to maxDurationTargetMs,
+                "overrunMs" to (durationMs - maxDurationTargetMs).coerceAtLeast(0L),
+                "streamUuid" to streamUuid,
+                "autoStopped" to recordingConfig.autoStopOnMaxDuration,
+            )
+        }
+
+        eventSender.sendExpoEvent(Constants.MAX_DURATION_REACHED_EVENT_NAME, event)
+        if (recordingConfig.autoStopOnMaxDuration) {
+            stopRecording(object : Promise {
+                override fun resolve(value: Any?) {
+                    LogUtils.d(CLASS_NAME, "Auto-stopped recording after maxDurationMs")
+                }
+                override fun reject(code: String?, message: String?, cause: Throwable?) {
+                    LogUtils.e(CLASS_NAME, "Failed to auto-stop recording after maxDurationMs: $message")
+                }
+            })
+        }
+    }
+
     private fun checkPermissions(options: Map<String, Any?>, promise: Promise): Boolean {
         if (!permissionUtils.checkRecordingPermission(enableBackgroundAudio)) {
             promise.reject(
@@ -996,6 +1120,7 @@ class AudioRecorderManager(
 
     fun stopRecording(promise: Promise) {
         val stopStartTime = System.currentTimeMillis()
+        cancelMaxDurationTimer()
         
         synchronized(audioRecordLock) {
             if (!_isRecording.get()) {
@@ -1246,6 +1371,7 @@ class AudioRecorderManager(
             acquireWakeLock()
             pausedDuration += System.currentTimeMillis() - lastPauseTime
             isPaused.set(false)
+            resumeMaxDurationTimer()
             
             synchronized(audioRecordLock) {
                 // Double-check audioRecord is valid after potential reinitialization
@@ -1292,6 +1418,7 @@ class AudioRecorderManager(
             lastPauseTime = System.currentTimeMillis()
             isPaused.set(true)
             pausedBySystemInterruption.set(isSystemInterruption)
+            pauseMaxDurationTimer()
 
             if (recordingConfig.showNotification) {
                 notificationManager.pauseUpdates()
@@ -1381,6 +1508,8 @@ class AudioRecorderManager(
                 "mimeType" to mimeType,
                 "size" to totalDataSize,
                 "interval" to recordingConfig.interval,
+                "maxDurationMs" to if (recordingConfig.maxDurationMs > 0) recordingConfig.maxDurationMs else null,
+                "maxDurationReached" to maxDurationReached,
                 "compression" to compressionBundle
             )
         }
@@ -1756,6 +1885,7 @@ class AudioRecorderManager(
     }
 
     fun cleanup() {
+        cancelMaxDurationTimer()
         synchronized(audioRecordLock) {
             try {
                 if (_isRecording.get()) {

package/android/src/main/java/net/siteed/audiostudio/AudioStudioModule.kt

@@ -90,6 +90,7 @@ class AudioStudioModule : Module(), EventSender, AudioStreamDecoderDelegate {
             Constants.AUDIO_EVENT_NAME,
             Constants.AUDIO_ANALYSIS_EVENT_NAME,
             Constants.RECORDING_INTERRUPTED_EVENT_NAME,
+            Constants.MAX_DURATION_REACHED_EVENT_NAME,
             Constants.TRIM_PROGRESS_EVENT,
             Constants.DEVICE_CHANGED_EVENT, // Add device changed event name
             Constants.AUDIO_STREAM_CHUNK_EVENT,

package/android/src/main/java/net/siteed/audiostudio/Constants.kt

@@ -9,6 +9,7 @@ object Constants {
     const val AUDIO_EVENT_NAME = "AudioData"
     const val AUDIO_ANALYSIS_EVENT_NAME = "AudioAnalysis"
     const val RECORDING_INTERRUPTED_EVENT_NAME = "onRecordingInterrupted"
+    const val MAX_DURATION_REACHED_EVENT_NAME = "MaxDurationReached"
     const val TRIM_PROGRESS_EVENT = "TrimProgress"
     const val DEVICE_CHANGED_EVENT = "deviceChangedEvent"
     const val AUDIO_STREAM_CHUNK_EVENT = "AudioDataStreamChunk"
@@ -38,4 +39,4 @@ object Constants {
     const val DEVICE_TYPE_WIRED_HEADPHONES = "wired_headphones"
     const val DEVICE_TYPE_SPEAKER = "speaker"
     const val DEVICE_TYPE_UNKNOWN = "unknown"
-}
+}

package/android/src/main/java/net/siteed/audiostudio/RecordingConfig.kt

@@ -67,6 +67,8 @@ data class RecordingConfig(
     val audioFocusStrategy: String? = null,
     val bufferDurationSeconds: Double? = null,
     val streamFormat: String = "raw",
+    val maxDurationMs: Long = 0L,
+    val autoStopOnMaxDuration: Boolean = false,
 ) {
     companion object {
         fun fromMap(options: Map<String, Any?>?): Result<Pair<RecordingConfig, AudioFormatInfo>> {
@@ -160,6 +162,8 @@ data class RecordingConfig(
                 audioFocusStrategy = audioFocusStrategy,
                 bufferDurationSeconds = (options["bufferDurationSeconds"] as? Number)?.toDouble(),
                 streamFormat = options.getStringOrDefault("streamFormat", "raw"),
+                maxDurationMs = options.getNumberOrDefault("maxDurationMs", 0L),
+                autoStopOnMaxDuration = options.getBooleanOrDefault("autoStopOnMaxDuration", false),
             )
 
             // Validate sample rate and channels
@@ -256,4 +260,4 @@ data class AudioFormatInfo(
     val format: Int,
     val mimeType: String,
     val fileExtension: String
-)
+)

package/build/cjs/AudioStudio.types.js.map

@@ -1 +1 @@
-{"version":3,"file":"AudioStudio.types.js","sourceRoot":"","sources":["../../src/AudioStudio.types.ts"],"names":[],"mappings":";;;AA8UA,4EAA4E;AAC/D,QAAA,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 { AudioAnalysisEvent } from './events'\n\nexport interface CompressionInfo {\n    /** Size of the compressed audio data in bytes */\n    size: number\n    /** MIME type of the compressed audio (e.g., 'audio/aac', 'audio/opus') */\n    mimeType: string\n    /** Bitrate of the compressed audio in bits per second */\n    bitrate: number\n    /** Format of the compression (e.g., 'aac', 'opus') */\n    format: string\n    /** URI to the compressed audio file if available */\n    compressedFileUri?: string\n}\n\nexport interface AudioStreamStatus {\n    /** Indicates whether audio recording is currently active */\n    isRecording: boolean\n    /** Indicates whether recording is in a paused state */\n    isPaused: boolean\n    /** Duration of the current recording in milliseconds */\n    durationMs: number\n    /** Size of the recorded audio data in bytes */\n    size: number\n    /** Interval in milliseconds at which recording data is emitted */\n    interval: number\n    /** Interval in milliseconds at which analysis data is emitted */\n    intervalAnalysis: number\n    /** MIME type of the recorded audio (e.g., 'audio/wav') */\n    mimeType: string\n    /** Information about audio compression if enabled */\n    compression?: CompressionInfo\n}\n\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 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    /** 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    /** Analysis data for the recording if processing was enabled */\n    analysisData?: AudioAnalysis // Analysis data for the recording depending on enableProcessing flag\n    /** Optional callback to handle recording interruptions */\n    onRecordingInterrupted?: (_: RecordingInterruptionEvent) => void\n}\n\n/**\n * Represents an event emitted during the trimming process to report progress.\n */\nexport interface TrimProgressEvent {\n    /**\n     * The percentage of the trimming process that has been completed, ranging from 0 to 100.\n     */\n    progress: number\n\n    /**\n     * The number of bytes that have been processed so far. This is optional and may not be provided in all implementations.\n     */\n    bytesProcessed?: number\n\n    /**\n     * The total number of bytes to process. This is optional and may not be provided in all implementations.\n     */\n    totalBytes?: number\n}\n\n/**\n * Defines a time range in milliseconds for trimming operations.\n */\nexport interface TimeRange {\n    /**\n     * The start time of the range in milliseconds.\n     */\n    startTimeMs: number\n\n    /**\n     * The end time of the range in milliseconds.\n     */\n    endTimeMs: number\n}\n\n/**\n * Options for configuring the audio trimming operation.\n */\nexport interface TrimAudioOptions {\n    /**\n     * The URI of the audio file to trim.\n     */\n    fileUri: string\n\n    /**\n     * The mode of trimming to apply.\n     * - `'single'`: Trims the audio to a single range defined by `startTimeMs` and `endTimeMs`.\n     * - `'keep'`: Keeps the specified `ranges` and removes all other portions of the audio.\n     * - `'remove'`: Removes the specified `ranges` and keeps the remaining portions of the audio.\n     * @default 'single'\n     */\n    mode?: 'single' | 'keep' | 'remove'\n\n    /**\n     * An array of time ranges to keep or remove, depending on the `mode`.\n     * - Required for `'keep'` and `'remove'` modes.\n     * - Ignored when `mode` is `'single'`.\n     */\n    ranges?: TimeRange[]\n\n    /**\n     * The start time in milliseconds for the `'single'` mode.\n     * - If not provided, trimming starts from the beginning of the audio (0 ms).\n     */\n    startTimeMs?: number\n\n    /**\n     * The end time in milliseconds for the `'single'` mode.\n     * - If not provided, trimming extends to the end of the audio.\n     */\n    endTimeMs?: number\n\n    /**\n     * The name of the output file. If not provided, a default name will be generated.\n     */\n    outputFileName?: string\n\n    /**\n     * Configuration for the output audio format.\n     */\n    outputFormat?: {\n        /**\n         * The format of the output audio file.\n         * - `'wav'`: Waveform Audio File Format (uncompressed).\n         * - `'aac'`: Advanced Audio Coding (compressed). Not supported on web platforms.\n         * - `'opus'`: Opus Interactive Audio Codec (compressed).\n         */\n        format: 'wav' | 'aac' | 'opus'\n\n        /**\n         * The sample rate of the output audio in Hertz (Hz).\n         * - If not provided, the input audio's sample rate is used.\n         */\n        sampleRate?: number\n\n        /**\n         * The number of channels in the output audio (e.g., 1 for mono, 2 for stereo).\n         * - If not provided, the input audio's channel count is used.\n         */\n        channels?: number\n\n        /**\n         * The bit depth of the output audio, applicable to PCM formats like `'wav'`.\n         * - If not provided, the input audio's bit depth is used.\n         */\n        bitDepth?: number\n\n        /**\n         * The bitrate of the output audio in bits per second, applicable to compressed formats like `'aac'`.\n         * - If not provided, a default bitrate is used based on the format.\n         */\n        bitrate?: number\n    }\n\n    /**\n     * Options for decoding the input audio file.\n     * - See `DecodingConfig` for details.\n     */\n    decodingOptions?: DecodingConfig\n}\n\n/**\n * Result of the audio trimming operation.\n */\nexport interface TrimAudioResult {\n    /**\n     * The URI of the trimmed audio file.\n     */\n    uri: string\n\n    /**\n     * The filename of the trimmed audio file.\n     */\n    filename: string\n\n    /**\n     * The duration of the trimmed audio in milliseconds.\n     */\n    durationMs: number\n\n    /**\n     * The size of the trimmed audio file in bytes.\n     */\n    size: number\n\n    /**\n     * The sample rate of the trimmed audio in Hertz (Hz).\n     */\n    sampleRate: number\n\n    /**\n     * The number of channels in the trimmed audio (e.g., 1 for mono, 2 for stereo).\n     */\n    channels: number\n\n    /**\n     * The bit depth of the trimmed audio, applicable to PCM formats like `'wav'`.\n     */\n    bitDepth: number\n\n    /**\n     * The MIME type of the trimmed audio file (e.g., `'audio/wav'`, `'audio/mpeg'`).\n     */\n    mimeType: string\n\n    /**\n     * Information about compression if the output format is compressed.\n     */\n    compression?: {\n        /**\n         * The format of the compression (e.g., `'aac'`, `'mp3'`, `'opus'`).\n         */\n        format: string\n\n        /**\n         * The bitrate of the compressed audio in bits per second.\n         */\n        bitrate: number\n\n        /**\n         * The size of the compressed audio file in bytes.\n         */\n        size: number\n    }\n\n    /**\n     * Information about the processing time.\n     */\n    processingInfo?: {\n        /**\n         * The time it took to process the audio in milliseconds.\n         */\n        durationMs: number\n    }\n}\n"]}
+{"version":3,"file":"AudioStudio.types.js","sourceRoot":"","sources":["../../src/AudioStudio.types.ts"],"names":[],"mappings":";;;AA+VA,4EAA4E;AAC/D,QAAA,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"]}