@edkimmel/expo-audio-stream 0.4.2 → 0.5.0

package/build/types.d.ts

@@ -103,9 +103,17 @@ export interface RecordingConfig {
     enableProcessing?: boolean;
     pointsPerSecond?: number;
     onAudioStream?: (event: AudioDataEvent) => Promise<void>;
+    /** Fired when the native layer reports a mid-recording error (e.g. system
+     * interruption like Siri or a phone call). The consumer should treat the
+     * recording session as terminated and clean up. */
+    onError?: (event: MicrophoneErrorEvent) => void;
     /** Optional frequency band crossover configuration. */
     frequencyBandConfig?: FrequencyBandConfig;
 }
+export interface MicrophoneErrorEvent {
+    code: string;
+    message: string;
+}
 export interface Chunk {
     text: string;
     timestamp: [number, number | null];

package/build/types.d.ts.map

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package/build/types.js.map

@@ -1 +1 @@
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// Round-trip time in ms\n  jitter?: number; // Network jitter in ms\n  packetLoss?: number; // Packet loss percentage (0-100)\n  bandwidth?: number; // Available bandwidth estimate\n}\n\n/**\n * Smart buffering configuration\n */\nexport interface SmartBufferConfig {\n  mode: SmartBufferMode;\n  networkConditions?: NetworkConditions;\n  adaptiveThresholds?: {\n    highLatencyMs?: number; // Threshold to enable aggressive buffering\n    highJitterMs?: number; // Threshold to increase buffer size\n    packetLossPercent?: number; // Threshold to enable buffering\n  };\n}\n\nexport interface StartRecordingResult {\n  fileUri: string;\n  mimeType: string;\n  channels?: number;\n  bitDepth?: BitDepth;\n  sampleRate?: SampleRate;\n}\n\nexport interface AudioDataEvent {\n  data: string | Float32Array;\n  data16kHz?: string | Float32Array;\n  position: number;\n  fileUri: string;\n  eventDataSize: number;\n  totalSize: number;\n  soundLevel?: number;\n  /** Frequency band RMS energy, present when recording is active. */\n  frequencyBands?: FrequencyBands;\n}\n\nexport interface RecordingConfig {\n  sampleRate?: SampleRate; // Sample rate for recording\n  channels?: 1 | 2; // 1 or 2 (MONO or STEREO)\n  encoding?: RecordingEncodingType; // Encoding type for the recording\n  interval?: number; // Interval in milliseconds at which to emit recording data\n\n  // Optional parameters for audio processing\n  enableProcessing?: boolean; // Boolean to enable/disable audio processing (default is false)\n  pointsPerSecond?: number; // Number of data points to extract per second of audio (default is 1000)\n  onAudioStream?: (event: AudioDataEvent) => Promise<void>; // Callback function to handle audio stream\n  /** Optional frequency band crossover configuration. */\n  frequencyBandConfig?: FrequencyBandConfig;\n}\n\nexport interface Chunk {\n  text: string;\n  timestamp: [number, number | null];\n}\n\nexport interface TranscriberData {\n  id: string;\n  isBusy: boolean;\n  text: string;\n  startTime: number;\n  endTime: number;\n  chunks: Chunk[];\n}\n\nexport interface AudioRecording {\n  fileUri: string;\n  filename: string;\n  durationMs: number;\n  size: number;\n  channels: number;\n  bitDepth: BitDepth;\n  sampleRate: SampleRate;\n  mimeType: string;\n  transcripts?: TranscriberData[];\n  wavPCMData?: Float32Array; // Full PCM data for the recording in WAV format (only on web, for native use the fileUri)\n}\n\n// Audio Jitter Buffer Types\n\n/**\n * Configuration for audio buffer management\n */\nexport interface IAudioBufferConfig {\n  targetBufferMs: number; // Target buffer size in milliseconds\n  minBufferMs: number; // Minimum buffer size before underrun handling\n  maxBufferMs: number; // Maximum buffer size before overrun handling\n  frameIntervalMs: number; // Expected frame interval in milliseconds\n}\n\n/**\n * Audio payload for playback containing base64 encoded audio data\n */\nexport interface IAudioPlayPayload {\n  audioData: string; // Base64 encoded PCM audio data\n  isFirst?: boolean; // True if this is the first chunk in a stream\n  isFinal?: boolean; // True if this is the final chunk in a stream\n}\n\n/**\n * Processed audio frame with metadata\n */\nexport interface IAudioFrame {\n  sequenceNumber: number; // Sequential frame number\n  data: IAudioPlayPayload; // Original audio payload\n  duration: number; // Estimated frame duration in milliseconds\n  timestamp: number; // Frame timestamp when processed\n}\n\n/**\n * Buffer health states for quality monitoring\n */\nexport type BufferHealthState =\n  | 'idle'\n  | 'healthy'\n  | 'degraded'\n  | 'critical';\n\n/**\n * Comprehensive buffer health and quality metrics\n */\nexport interface IBufferHealthMetrics {\n  currentBufferMs: number; // Current buffer level in milliseconds\n  targetBufferMs: number; // Target buffer level in milliseconds\n  underrunCount: number; // Total number of buffer underruns\n  overrunCount: number; // Total number of buffer overruns\n  averageJitter: number; // Average network jitter in milliseconds\n  bufferHealthState: BufferHealthState; // Current buffer health assessment\n  adaptiveAdjustmentsCount: number; // Number of adaptive adjustments made\n}\n\n/**\n * Interface for audio buffer management\n */\nexport interface IAudioBufferManager {\n  enqueueFrames(audioData: IAudioPlayPayload): void;\n  startPlayback(): void;\n  stopPlayback(): void;\n  isPlaying(): boolean;\n  getHealthMetrics(): IBufferHealthMetrics;\n  updateConfig(config: Partial<IAudioBufferConfig>): void;\n  applyAdaptiveAdjustments(): void;\n  destroy(): void;\n  getCurrentBufferMs(): number;\n}\n\n/**\n * Interface for frame processing\n */\nexport interface IFrameProcessor {\n  parseChunk(payload: IAudioPlayPayload): IAudioFrame[];\n  reset(): void;\n}\n\n/**\n * Interface for quality monitoring\n */\nexport interface IQualityMonitor {\n  recordFrameArrival(timestamp: number): void;\n  recordUnderrun(): void;\n  recordOverrun(): void;\n  updateBufferLevel(bufferMs: number): void;\n  getMetrics(): IBufferHealthMetrics;\n  getBufferHealthState(\n    isPlaying: boolean,\n    currentLatencyMs: number\n  ): BufferHealthState;\n  getRecommendedAdjustment(): number;\n  reset(): void;\n}\n"]}
+{"version":3,"file":"types.js","sourceRoot":"","sources":["../src/types.ts"],"names":[],"mappings":"AAOA,MAAM,CAAC,MAAM,aAAa,GAAG;IAC3B,OAAO,EAAE,SAAS;IAClB,gBAAgB,EAAE,iBAAiB;IACnC,YAAY,EAAE,cAAc;CACpB,CAAC;AAgCX,MAAM,CAAC,MAAM,aAAa,GAAG;IAC3B,SAAS,EAAE,WAAW;IACtB,SAAS,EAAE,WAAW;CACd,CAAC","sourcesContent":["export type RecordingEncodingType =\n  | 'pcm_32bit'\n  | 'pcm_16bit'\n  | 'pcm_8bit';\nexport type SampleRate = 16000 | 24000 | 44100 | 48000;\nexport type BitDepth = 8 | 16 | 32;\n\nexport const PlaybackModes = {\n  REGULAR: 'regular',\n  VOICE_PROCESSING: 'voiceProcessing',\n  CONVERSATION: 'conversation',\n} as const;\n/**\n * Defines different playback modes for audio processing\n */\nexport type PlaybackMode =\n  (typeof PlaybackModes)[keyof typeof PlaybackModes];\n\n/**\n * Configuration for buffered audio streaming\n */\nexport interface BufferedStreamConfig {\n  /**\n   * Turn ID for queue management\n   */\n  turnId: string;\n\n  /**\n   * Audio encoding format\n   */\n  encoding?: Encoding;\n\n  /**\n   * Buffer configuration options\n   */\n  bufferConfig?: Partial<IAudioBufferConfig>;\n\n  /**\n   * Callback for buffer health updates\n   */\n  onBufferHealth?: (metrics: IBufferHealthMetrics) => void;\n}\n\nexport const EncodingTypes = {\n  PCM_F32LE: 'pcm_f32le',\n  PCM_S16LE: 'pcm_s16le',\n} as const;\n\n/**\n * Defines different encoding formats for audio data\n */\nexport type Encoding =\n  (typeof EncodingTypes)[keyof typeof EncodingTypes];\n\n/** RMS energy per frequency band, range [0, 1]. */\nexport interface FrequencyBands {\n  low: number;\n  mid: number;\n  high: number;\n}\n\n/** Crossover frequency configuration for band analysis. */\nexport interface FrequencyBandConfig {\n  /** Low/mid crossover in Hz (default 300). */\n  lowCrossoverHz?: number;\n  /** Mid/high crossover in Hz (default 2000). */\n  highCrossoverHz?: number;\n}\n\n/**\n * Smart buffering mode options\n */\nexport type SmartBufferMode =\n  | 'conservative'\n  | 'balanced'\n  | 'aggressive'\n  | 'adaptive';\n\n/**\n * Network condition indicators for smart buffering\n */\nexport interface NetworkConditions {\n  latency?: number; // Round-trip time in ms\n  jitter?: number; // Network jitter in ms\n  packetLoss?: number; // Packet loss percentage (0-100)\n  bandwidth?: number; // Available bandwidth estimate\n}\n\n/**\n * Smart buffering configuration\n */\nexport interface SmartBufferConfig {\n  mode: SmartBufferMode;\n  networkConditions?: NetworkConditions;\n  adaptiveThresholds?: {\n    highLatencyMs?: number; // Threshold to enable aggressive buffering\n    highJitterMs?: number; // Threshold to increase buffer size\n    packetLossPercent?: number; // Threshold to enable buffering\n  };\n}\n\nexport interface StartRecordingResult {\n  fileUri: string;\n  mimeType: string;\n  channels?: number;\n  bitDepth?: BitDepth;\n  sampleRate?: SampleRate;\n}\n\nexport interface AudioDataEvent {\n  data: string | Float32Array;\n  data16kHz?: string | Float32Array;\n  position: number;\n  fileUri: string;\n  eventDataSize: number;\n  totalSize: number;\n  soundLevel?: number;\n  /** Frequency band RMS energy, present when recording is active. */\n  frequencyBands?: FrequencyBands;\n}\n\nexport interface RecordingConfig {\n  sampleRate?: SampleRate; // Sample rate for recording\n  channels?: 1 | 2; // 1 or 2 (MONO or STEREO)\n  encoding?: RecordingEncodingType; // Encoding type for the recording\n  interval?: number; // Interval in milliseconds at which to emit recording data\n\n  // Optional parameters for audio processing\n  enableProcessing?: boolean; // Boolean to enable/disable audio processing (default is false)\n  pointsPerSecond?: number; // Number of data points to extract per second of audio (default is 1000)\n  onAudioStream?: (event: AudioDataEvent) => Promise<void>; // Callback function to handle audio stream\n  /** Fired when the native layer reports a mid-recording error (e.g. system\n   * interruption like Siri or a phone call). The consumer should treat the\n   * recording session as terminated and clean up. */\n  onError?: (event: MicrophoneErrorEvent) => void;\n  /** Optional frequency band crossover configuration. */\n  frequencyBandConfig?: FrequencyBandConfig;\n}\n\nexport interface MicrophoneErrorEvent {\n  code: string;\n  message: string;\n}\n\nexport interface Chunk {\n  text: string;\n  timestamp: [number, number | null];\n}\n\nexport interface TranscriberData {\n  id: string;\n  isBusy: boolean;\n  text: string;\n  startTime: number;\n  endTime: number;\n  chunks: Chunk[];\n}\n\nexport interface AudioRecording {\n  fileUri: string;\n  filename: string;\n  durationMs: number;\n  size: number;\n  channels: number;\n  bitDepth: BitDepth;\n  sampleRate: SampleRate;\n  mimeType: string;\n  transcripts?: TranscriberData[];\n  wavPCMData?: Float32Array; // Full PCM data for the recording in WAV format (only on web, for native use the fileUri)\n}\n\n// Audio Jitter Buffer Types\n\n/**\n * Configuration for audio buffer management\n */\nexport interface IAudioBufferConfig {\n  targetBufferMs: number; // Target buffer size in milliseconds\n  minBufferMs: number; // Minimum buffer size before underrun handling\n  maxBufferMs: number; // Maximum buffer size before overrun handling\n  frameIntervalMs: number; // Expected frame interval in milliseconds\n}\n\n/**\n * Audio payload for playback containing base64 encoded audio data\n */\nexport interface IAudioPlayPayload {\n  audioData: string; // Base64 encoded PCM audio data\n  isFirst?: boolean; // True if this is the first chunk in a stream\n  isFinal?: boolean; // True if this is the final chunk in a stream\n}\n\n/**\n * Processed audio frame with metadata\n */\nexport interface IAudioFrame {\n  sequenceNumber: number; // Sequential frame number\n  data: IAudioPlayPayload; // Original audio payload\n  duration: number; // Estimated frame duration in milliseconds\n  timestamp: number; // Frame timestamp when processed\n}\n\n/**\n * Buffer health states for quality monitoring\n */\nexport type BufferHealthState =\n  | 'idle'\n  | 'healthy'\n  | 'degraded'\n  | 'critical';\n\n/**\n * Comprehensive buffer health and quality metrics\n */\nexport interface IBufferHealthMetrics {\n  currentBufferMs: number; // Current buffer level in milliseconds\n  targetBufferMs: number; // Target buffer level in milliseconds\n  underrunCount: number; // Total number of buffer underruns\n  overrunCount: number; // Total number of buffer overruns\n  averageJitter: number; // Average network jitter in milliseconds\n  bufferHealthState: BufferHealthState; // Current buffer health assessment\n  adaptiveAdjustmentsCount: number; // Number of adaptive adjustments made\n}\n\n/**\n * Interface for audio buffer management\n */\nexport interface IAudioBufferManager {\n  enqueueFrames(audioData: IAudioPlayPayload): void;\n  startPlayback(): void;\n  stopPlayback(): void;\n  isPlaying(): boolean;\n  getHealthMetrics(): IBufferHealthMetrics;\n  updateConfig(config: Partial<IAudioBufferConfig>): void;\n  applyAdaptiveAdjustments(): void;\n  destroy(): void;\n  getCurrentBufferMs(): number;\n}\n\n/**\n * Interface for frame processing\n */\nexport interface IFrameProcessor {\n  parseChunk(payload: IAudioPlayPayload): IAudioFrame[];\n  reset(): void;\n}\n\n/**\n * Interface for quality monitoring\n */\nexport interface IQualityMonitor {\n  recordFrameArrival(timestamp: number): void;\n  recordUnderrun(): void;\n  recordOverrun(): void;\n  updateBufferLevel(bufferMs: number): void;\n  getMetrics(): IBufferHealthMetrics;\n  getBufferHealthState(\n    isPlaying: boolean,\n    currentLatencyMs: number\n  ): BufferHealthState;\n  getRecommendedAdjustment(): number;\n  reset(): void;\n}\n"]}

package/ios/AudioPipeline.swift

@@ -17,6 +17,7 @@ protocol PipelineListener: AnyObject {
     func onZombieDetected(stalledMs: Int64)
     func onUnderrun(count: Int)
     func onDrained(turnId: String)
+    func onPlaybackStopped(turnId: String)
     func onAudioFocusLost()
     func onAudioFocusResumed()
     func onFrequencyBands(low: Float, mid: Float, high: Float)
@@ -84,6 +85,10 @@ class AudioPipeline: SharedAudioEngineDelegate {
     /// This prevents duplicate chains and stale callbacks from re-entering after a rebuild.
     private var scheduleGeneration: Int = 0
 
+    /// Pending PlaybackStopped dispatch — cancelled on new turn / disconnect.
+    /// Always mutate from the main queue to avoid races with the drain timer.
+    private var pendingPlaybackStoppedWork: DispatchWorkItem?
+
     // ── Timers ──────────────────────────────────────────────────────────
     private var stateTimer: DispatchSourceTimer?
     private var zombieTimer: DispatchSourceTimer?
@@ -122,7 +127,7 @@ class AudioPipeline: SharedAudioEngineDelegate {
     // Connect / Disconnect
     // ════════════════════════════════════════════════════════════════════
 
-    func connect() {
+    func connect() throws {
         guard !running else {
             Logger.debug("[\(AudioPipeline.TAG)] connect() called while already running — ignoring")
             return
@@ -197,12 +202,17 @@ class AudioPipeline: SharedAudioEngineDelegate {
         } catch {
             Logger.debug("[\(AudioPipeline.TAG)] connect() failed: \(error)")
             setState(.error)
-            listener?.onError(code: "CONNECT_FAILED", message: error.localizedDescription)
             disconnect()
+            throw error
         }
     }
 
     func disconnect() {
+        // Cancel any pending PlaybackStopped before tearing down.
+        // DispatchWorkItem.cancel is thread-safe; we may be on any queue here.
+        pendingPlaybackStoppedWork?.cancel()
+        pendingPlaybackStoppedWork = nil
+
         running = false
         // Invalidate all in-flight completion handlers before detaching.
         scheduleGeneration += 1
@@ -365,6 +375,9 @@ class AudioPipeline: SharedAudioEngineDelegate {
             lastReportedUnderrunCount = 0
             setState(.streaming)
             frequencyBandAnalyzer?.reset()
+            DispatchQueue.main.async { [weak self] in
+                self?.cancelPendingPlaybackStopped()
+            }
         }
 
         // ── Decode base64 → PCM shorts ──────────────────────────────────
@@ -407,6 +420,9 @@ class AudioPipeline: SharedAudioEngineDelegate {
         lastReportedUnderrunCount = 0
         setState(.idle)
         frequencyBandAnalyzer?.reset()
+        DispatchQueue.main.async { [weak self] in
+            self?.cancelPendingPlaybackStopped()
+        }
     }
 
     // ════════════════════════════════════════════════════════════════════
@@ -433,6 +449,18 @@ class AudioPipeline: SharedAudioEngineDelegate {
         ]
     }
 
+    /// Current platform output latency in milliseconds.
+    ///
+    /// Reads `AVAudioSession.sharedInstance().outputLatency`. The value
+    /// reflects total HW output latency to the speaker and changes on
+    /// audio route changes (built-in vs. Bluetooth vs. wired).
+    ///
+    /// Returns 0 if not running.
+    func outputLatencyMs() -> Double {
+        guard running else { return 0 }
+        return AVAudioSession.sharedInstance().outputLatency * 1000.0
+    }
+
     // ════════════════════════════════════════════════════════════════════
     // Scheduling loop
     // ════════════════════════════════════════════════════════════════════
@@ -548,11 +576,48 @@ class AudioPipeline: SharedAudioEngineDelegate {
         if buf.isDrained() && currentState == .draining {
             if let tid = turnId {
                 listener?.onDrained(turnId: tid)
+                schedulePlaybackStopped(turnId: tid)
             }
             setState(.idle)
         }
     }
 
+    /// Schedule a `PlaybackStopped` event approximately `outputLatencyMs`
+    /// after `Drained`. Cancels any previously pending dispatch.
+    ///
+    /// Approximation note: at drain detection, up to `PRE_SCHEDULE_COUNT`
+    /// pre-scheduled buffers may still be in the player node's chain — but
+    /// they read silence from the empty jitter buffer, so the last audible
+    /// sample stops emitting roughly `outputLatency` after this point.
+    private func schedulePlaybackStopped(turnId: String) {
+        pendingPlaybackStoppedWork?.cancel()
+        pendingPlaybackStoppedWork = nil
+
+        let latencyMs = outputLatencyMs()
+        let work = DispatchWorkItem { [weak self] in
+            guard let self = self else { return }
+            self.pendingPlaybackStoppedWork = nil
+            self.listener?.onPlaybackStopped(turnId: turnId)
+        }
+        pendingPlaybackStoppedWork = work
+
+        if latencyMs > 0 {
+            DispatchQueue.main.asyncAfter(
+                deadline: .now() + .milliseconds(Int(latencyMs.rounded())),
+                execute: work
+            )
+        } else {
+            DispatchQueue.main.async(execute: work)
+        }
+    }
+
+    /// Cancel any pending PlaybackStopped dispatch. Call from turn-boundary
+    /// transitions (new pushAudio, invalidateTurn) and disconnect.
+    private func cancelPendingPlaybackStopped() {
+        pendingPlaybackStoppedWork?.cancel()
+        pendingPlaybackStoppedWork = nil
+    }
+
     // ════════════════════════════════════════════════════════════════════
     // Zombie detection
     // ════════════════════════════════════════════════════════════════════

package/ios/ExpoPlayAudioStreamModule.swift

@@ -48,6 +48,7 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
             PipelineIntegration.EVENT_ZOMBIE_DETECTED,
             PipelineIntegration.EVENT_UNDERRUN,
             PipelineIntegration.EVENT_DRAINED,
+            PipelineIntegration.EVENT_PLAYBACK_STOPPED,
             PipelineIntegration.EVENT_AUDIO_FOCUS_LOST,
             PipelineIntegration.EVENT_AUDIO_FOCUS_RESUMED,
             PipelineIntegration.EVENT_FREQUENCY_BANDS,
@@ -209,6 +210,15 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
 
                 promise.resolve(result)
             } catch {
+                // Reset session + engine state so a subsequent connect can recover.
+                // Without this, a partial failure (e.g. setActive denial after the
+                // iOS local-network permission prompt) leaves the session stuck and
+                // every retry fails the same way.
+                self._pipelineIntegration?.removeAsDelegate(from: self.sharedAudioEngine)
+                self._pipelineIntegration?.disconnect()
+                self.sharedAudioEngine.teardown()
+                try? AVAudioSession.sharedInstance().setActive(false, options: .notifyOthersOnDeactivation)
+                self.isAudioSessionInitialized = false
                 promise.reject("PIPELINE_CONNECT_ERROR", error.localizedDescription)
             }
         }
@@ -248,6 +258,10 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
         Function("getPipelineState") { () -> String in
             return self.pipelineIntegration.getState()
         }
+
+        Function("getPipelineOutputLatencyMs") { () -> Double in
+            return self.pipelineIntegration.outputLatencyMs()
+        }
     }
 
     private func ensureAudioSessionInitialized(settings recordingSettings: RecordingSettings? = nil) throws {

package/ios/Microphone.swift

@@ -28,6 +28,14 @@ class Microphone {
     private var isSilent: Bool = false
     private var frequencyBandAnalyzer: FrequencyBandAnalyzer?
     private var frequencyBandConfig: (lowCrossoverHz: Float, highCrossoverHz: Float)?
+
+    /// Interval (in ms) the consumer last requested. Used to rebuild the tap
+    /// with the same cadence when resuming after an interruption.
+    private var lastIntervalMs: Int = 100
+    /// Set when an audio session interruption begins while recording is active.
+    /// Cleared either by `stopRecording` (consumer chose disconnect) or by the
+    /// `.ended` handler after a successful auto-resume.
+    private var pendingInterruptionResume: Bool = false
     
     init() {
         NotificationCenter.default.addObserver(
@@ -36,8 +44,18 @@ class Microphone {
             name: AVAudioSession.routeChangeNotification,
             object: nil
         )
+        NotificationCenter.default.addObserver(
+            self,
+            selector: #selector(handleInterruption),
+            name: AVAudioSession.interruptionNotification,
+            object: nil
+        )
     }
-    
+
+    deinit {
+        NotificationCenter.default.removeObserver(self)
+    }
+
     /// Handles audio route changes (e.g. headphones connected/disconnected)
     /// - Parameter notification: The notification object containing route change information
     @objc private func handleRouteChange(notification: Notification) {
@@ -46,7 +64,7 @@ class Microphone {
               let reason = AVAudioSession.RouteChangeReason(rawValue: reasonValue) else {
             return
         }
-        
+
         Logger.debug("[Microphone] Route is changed \(reason)")
 
         switch reason {
@@ -55,7 +73,7 @@ class Microphone {
                 stopRecording(resolver: nil)
                 DispatchQueue.main.asyncAfter(deadline: .now() + 0.5) { [weak self] in
                     guard let self = self, let settings = self.recordingSettings else { return }
-                    
+
                     _ = startRecording(settings: self.recordingSettings!, intervalMilliseconds: 100, frequencyBandConfig: self.frequencyBandConfig)
                 }
             }
@@ -65,6 +83,96 @@ class Microphone {
             break
         }
     }
+
+    /// Handles audio session interruptions (e.g. Siri, phone call, alarm).
+    ///
+    /// On `.began` we stop the engine (iOS deactivates the session regardless)
+    /// but keep `isRecording = true` and emit an `INTERRUPTED` error so the
+    /// consumer can decide:
+    ///   - Call `stopRecording` to disconnect explicitly (clears the resume flag).
+    ///   - Do nothing to let the library auto-resume when the system gives the
+    ///     mic back on `.ended`.
+    @objc private func handleInterruption(notification: Notification) {
+        guard let info = notification.userInfo,
+              let typeValue = info[AVAudioSessionInterruptionTypeKey] as? UInt,
+              let type = AVAudioSession.InterruptionType(rawValue: typeValue) else {
+            return
+        }
+
+        switch type {
+        case .began:
+            Logger.debug("[Microphone] Audio session interruption began")
+            guard isRecording else { return }
+            // Tear down the engine but leave isRecording true so the consumer's
+            // intent is preserved. The .ended branch will decide whether to
+            // resume based on whether stopRecording was called in between.
+            pendingInterruptionResume = true
+            if let engine = audioEngine {
+                engine.inputNode.removeTap(onBus: 0)
+                engine.stop()
+            }
+            delegate?.onMicrophoneError("INTERRUPTED", "Audio session interrupted by system")
+        case .ended:
+            Logger.debug("[Microphone] Audio session interruption ended")
+            guard pendingInterruptionResume else { return }
+            pendingInterruptionResume = false
+            // iOS uses AVAudioSessionInterruptionOptionKey.shouldResume to hint
+            // whether the app can safely reactivate. Absent means another app
+            // took over (e.g. an ongoing call) — in that case we surface a
+            // terminal error instead of attempting a doomed restart.
+            let optionsValue = info[AVAudioSessionInterruptionOptionKey] as? UInt ?? 0
+            let options = AVAudioSession.InterruptionOptions(rawValue: optionsValue)
+            guard options.contains(.shouldResume) else {
+                Logger.debug("[Microphone] System did not signal shouldResume — treating as terminal stop")
+                isRecording = false
+                delegate?.onMicrophoneError(
+                    "RESUME_DENIED",
+                    "System did not permit microphone resume after interruption"
+                )
+                return
+            }
+            do {
+                try AVAudioSession.sharedInstance().setActive(true)
+                try installTapAndStartEngine(intervalMilliseconds: lastIntervalMs)
+                Logger.debug("[Microphone] Auto-resumed after interruption")
+            } catch {
+                Logger.debug("[Microphone] Auto-resume failed: \(error.localizedDescription)")
+                // Engine couldn't restart — surface as terminal stop.
+                isRecording = false
+                delegate?.onMicrophoneError(
+                    "RESUME_FAILED",
+                    "Could not restart microphone after interruption: \(error.localizedDescription)"
+                )
+            }
+        @unknown default:
+            break
+        }
+    }
+
+    /// Installs the audio tap on the input node and starts the engine.
+    /// Shared between fresh `startRecording` and post-interruption resume.
+    private func installTapAndStartEngine(intervalMilliseconds: Int) throws {
+        let hardwareFormat = audioEngine.inputNode.inputFormat(forBus: 0)
+        let intervalSamples = AVAudioFrameCount(
+            Double(intervalMilliseconds) / 1000.0 * hardwareFormat.sampleRate
+        )
+        let tapBufferSize = max(intervalSamples, 256)
+
+        audioEngine.inputNode.installTap(onBus: 0, bufferSize: tapBufferSize, format: nil) { [weak self] (buffer, time) in
+            guard let self = self else { return }
+
+            guard buffer.frameLength > 0 else {
+                Logger.debug("Error: received empty buffer in tap callback")
+                self.delegate?.onMicrophoneError("READ_ERROR", "Received empty audio buffer")
+                return
+            }
+
+            self.processAudioBuffer(buffer)
+            self.lastBufferTime = time
+        }
+
+        try audioEngine.start()
+    }
     
     func toggleSilence(isSilent: Bool) {
         Logger.debug("[Microphone] toggleSilence")
@@ -101,6 +209,7 @@ class Microphone {
         recordingSettings = newSettings  // Update the class property with the new settings
 
         self.frequencyBandConfig = frequencyBandConfig
+        self.lastIntervalMs = intervalMilliseconds
         // Analyzer uses the desired (target) sample rate, not hardware rate
         let targetRate = Int(settings.desiredSampleRate ?? settings.sampleRate)
         let fbConfig = frequencyBandConfig ?? (lowCrossoverHz: Float(300), highCrossoverHz: Float(2000))
@@ -110,30 +219,9 @@ class Microphone {
             highCrossoverHz: fbConfig.highCrossoverHz
         )
 
-        // Compute tap buffer size from interval so Core Audio delivers at the right cadence
-        let intervalSamples = AVAudioFrameCount(
-            Double(intervalMilliseconds) / 1000.0 * hardwareFormat.sampleRate
-        )
-        let tapBufferSize = max(intervalSamples, 256) // floor at 256 frames (~5ms at 48kHz)
-
-        // Pass nil for format to use the hardware's native format, avoiding format mismatch crashes.
-        // Core Audio does not support format conversion (e.g. Float32 -> Int16) on the tap itself.
-        audioEngine.inputNode.installTap(onBus: 0, bufferSize: tapBufferSize, format: nil) { [weak self] (buffer, time) in
-            guard let self = self else { return }
-
-            guard buffer.frameLength > 0 else {
-                Logger.debug("Error: received empty buffer in tap callback")
-                self.delegate?.onMicrophoneError("READ_ERROR", "Received empty audio buffer")
-                return
-            }
-
-            self.processAudioBuffer(buffer)
-            self.lastBufferTime = time
-        }
-        
         do {
             startTime = Date()
-            try audioEngine.start()
+            try installTapAndStartEngine(intervalMilliseconds: intervalMilliseconds)
             isRecording = true
             Logger.debug("Debug: Recording started successfully.")
             return StartRecordingResult(
@@ -151,6 +239,9 @@ class Microphone {
     }
     
     public func stopRecording(resolver promise: Promise?) {
+        // Clear resume intent up-front so a stopRecording call during an
+        // active interruption window cancels the auto-resume on .ended.
+        pendingInterruptionResume = false
         guard self.isRecording else {
             if let promiseResolver = promise {
                 promiseResolver.resolve(nil)

package/ios/PipelineIntegration.swift

@@ -20,6 +20,7 @@ class PipelineIntegration: PipelineListener {
     static let EVENT_ZOMBIE_DETECTED     = "PipelineZombieDetected"
     static let EVENT_UNDERRUN            = "PipelineUnderrun"
     static let EVENT_DRAINED             = "PipelineDrained"
+    static let EVENT_PLAYBACK_STOPPED    = "PipelinePlaybackStopped"
     static let EVENT_AUDIO_FOCUS_LOST    = "PipelineAudioFocusLost"
     static let EVENT_AUDIO_FOCUS_RESUMED = "PipelineAudioFocusResumed"
     static let EVENT_FREQUENCY_BANDS    = "PipelineFrequencyBands"
@@ -72,7 +73,7 @@ class PipelineIntegration: PipelineListener {
             sharedEngine: sharedEngine,
             listener: self
         )
-        p.connect()
+        try p.connect()
         pipeline = p
 
         return [
@@ -152,6 +153,11 @@ class PipelineIntegration: PipelineListener {
         return pipeline?.getState().rawValue ?? PipelineState.idle.rawValue
     }
 
+    /// Current platform output latency in milliseconds. Returns 0 if not connected.
+    func outputLatencyMs() -> Double {
+        return pipeline?.outputLatencyMs() ?? 0
+    }
+
     /// Register the pipeline as a delegate on the shared engine.
     /// Called by the module after connect() so route changes and interruptions
     /// are forwarded to the AudioPipeline instance.
@@ -206,6 +212,10 @@ class PipelineIntegration: PipelineListener {
         sendEvent(PipelineIntegration.EVENT_DRAINED, ["turnId": turnId])
     }
 
+    func onPlaybackStopped(turnId: String) {
+        sendEvent(PipelineIntegration.EVENT_PLAYBACK_STOPPED, ["turnId": turnId])
+    }
+
     func onAudioFocusLost() {
         sendEvent(PipelineIntegration.EVENT_AUDIO_FOCUS_LOST, [:])
     }

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@edkimmel/expo-audio-stream",
-  "version": "0.4.2",
+  "version": "0.5.0",
   "description": "Expo Play Audio Stream module",
   "main": "build/index.js",
   "types": "build/index.d.ts",
@@ -9,7 +9,6 @@
     "build": "expo-module build",
     "clean": "expo-module clean",
     "lint": "expo-module lint",
-    "test": "expo-module test",
     "prepare": "expo-module prepare && husky || true",
     "prepublishOnly": "expo-module prepublishOnly",
     "expo-module": "expo-module",

package/plugin/build/index.js

@@ -2,12 +2,17 @@
 Object.defineProperty(exports, "__esModule", { value: true });
 const config_plugins_1 = require("@expo/config-plugins");
 const MICROPHONE_USAGE = 'Allow $(PRODUCT_NAME) to access your microphone';
+// AVFoundation enumerates AirPlay/Continuity audio devices on the local network
+// even though we don't use them — without this description, iOS shows a generic
+// prompt and a denial leaves the audio session unable to activate.
+const LOCAL_NETWORK_USAGE = 'Allow $(PRODUCT_NAME) to discover audio devices on your local network';
 const withRecordingPermission = (config, existingPerms) => {
     if (!existingPerms) {
         console.warn('No previous permissions provided');
     }
     config = (0, config_plugins_1.withInfoPlist)(config, (config) => {
         config.modResults['NSMicrophoneUsageDescription'] = config.modResults['NSMicrophoneUsageDescription'] || MICROPHONE_USAGE;
+        config.modResults['NSLocalNetworkUsageDescription'] = config.modResults['NSLocalNetworkUsageDescription'] || LOCAL_NETWORK_USAGE;
         // Add audio to UIBackgroundModes to allow background audio recording
         const existingBackgroundModes = config.modResults.UIBackgroundModes || [];
         if (!existingBackgroundModes.includes('audio')) {

package/plugin/src/index.ts

@@ -6,6 +6,10 @@ import {
 } from '@expo/config-plugins'
 
 const MICROPHONE_USAGE = 'Allow $(PRODUCT_NAME) to access your microphone'
+// AVFoundation enumerates AirPlay/Continuity audio devices on the local network
+// even though we don't use them — without this description, iOS shows a generic
+// prompt and a denial leaves the audio session unable to activate.
+const LOCAL_NETWORK_USAGE = 'Allow $(PRODUCT_NAME) to discover audio devices on your local network'
 
 const withRecordingPermission: ConfigPlugin<{
     microphonePermission: string
@@ -15,6 +19,7 @@ const withRecordingPermission: ConfigPlugin<{
     }
     config = withInfoPlist(config, (config) => {
         config.modResults['NSMicrophoneUsageDescription'] = config.modResults['NSMicrophoneUsageDescription'] || MICROPHONE_USAGE
+        config.modResults['NSLocalNetworkUsageDescription'] = config.modResults['NSLocalNetworkUsageDescription'] || LOCAL_NETWORK_USAGE
 
         // Add audio to UIBackgroundModes to allow background audio recording
         const existingBackgroundModes =

package/src/events.ts

@@ -21,6 +21,10 @@ export interface AudioEventPayload {
   streamUuid: string;
   soundLevel?: number;
   frequencyBands?: { low: number; mid: number; high: number };
+  /** Set by native when a mid-recording error occurs (interruption, read failure).
+   * When present, `encoded` is absent and the recording is no longer active. */
+  error?: string;
+  errorMessage?: string;
 }
 
 export const DeviceReconnectedReasons = {

package/src/index.ts

@@ -59,9 +59,12 @@ export class ExpoPlayAudioStream {
   }> {
     let subscription: Subscription | undefined;
     try {
-      const { onAudioStream, ...options } = recordingConfig;
+      const { onAudioStream, onError, ...options } = recordingConfig;
 
-      if (onAudioStream && typeof onAudioStream == "function") {
+      if (
+        (onAudioStream && typeof onAudioStream == "function") ||
+        (onError && typeof onError == "function")
+      ) {
         subscription = addAudioEventListener(
           async (event: AudioEventPayload) => {
             const {
@@ -72,7 +75,13 @@ export class ExpoPlayAudioStream {
               encoded,
               soundLevel,
               frequencyBands,
+              error,
+              errorMessage,
             } = event;
+            if (error) {
+              onError?.({ code: error, message: errorMessage ?? "" });
+              return;
+            }
             if (!encoded) {
               console.error(
                 `[ExpoPlayAudioStream] Encoded audio data is missing`
@@ -259,6 +268,7 @@ export type {
   PipelineZombieDetectedEvent,
   PipelineUnderrunEvent,
   PipelineDrainedEvent,
+  PipelinePlaybackStoppedEvent,
   PipelineAudioFocusLostEvent,
   PipelineAudioFocusResumedEvent,
 } from "./pipeline";

package/src/pipeline/index.ts

@@ -103,6 +103,22 @@ export class Pipeline {
     return ExpoPlayAudioStreamModule.getPipelineTelemetry() as PipelineTelemetry;
   }
 
+  /**
+   * Query the platform's current output latency — i.e., how long after a
+   * sample is written to the native buffer before it actually leaves the
+   * speaker.
+   *
+   * Value can change mid-session, notably on audio route changes such as
+   * switching from built-in speaker to Bluetooth (Bluetooth typically adds
+   * 100+ ms). **Always query at the moment you care; do not cache.**
+   *
+   * Returns 0 if the pipeline is not connected or the platform cannot
+   * report a value.
+   */
+  static getOutputLatencyMs(): number {
+    return ExpoPlayAudioStreamModule.getPipelineOutputLatencyMs() as number;
+  }
+
   // ════════════════════════════════════════════════════════════════════════
   // Event subscriptions
   // ════════════════════════════════════════════════════════════════════════
@@ -211,6 +227,7 @@ export type {
   PipelineZombieDetectedEvent,
   PipelineUnderrunEvent,
   PipelineDrainedEvent,
+  PipelinePlaybackStoppedEvent,
   PipelineAudioFocusLostEvent,
   PipelineAudioFocusResumedEvent,
 } from './types';

package/src/pipeline/types.ts

@@ -135,6 +135,20 @@ export interface PipelineDrainedEvent {
   turnId: string;
 }
 
+/**
+ * Payload for `PipelinePlaybackStopped`.
+ *
+ * Fired when the last sample physically leaves the speaker, approximately
+ * `outputLatencyMs` after `PipelineDrained` for the same turn. Pairs with
+ * `PipelinePlaybackStarted` (start-of-emission ↔ end-of-emission).
+ *
+ * Note: this is a physical-world milestone, distinct from `state: 'idle'`
+ * (the pipeline-state-machine value reported via `PipelineStateChanged`).
+ */
+export interface PipelinePlaybackStoppedEvent {
+  turnId: string;
+}
+
 /** Payload for `PipelineAudioFocusLost` (empty — presence is the signal). */
 export type PipelineAudioFocusLostEvent = Record<string, never>;
 
@@ -155,6 +169,7 @@ export interface PipelineEventMap {
   PipelineZombieDetected: PipelineZombieDetectedEvent;
   PipelineUnderrun: PipelineUnderrunEvent;
   PipelineDrained: PipelineDrainedEvent;
+  PipelinePlaybackStopped: PipelinePlaybackStoppedEvent;
   PipelineAudioFocusLost: PipelineAudioFocusLostEvent;
   PipelineAudioFocusResumed: PipelineAudioFocusResumedEvent;
   PipelineFrequencyBands: PipelineFrequencyBandsEvent;