@edkimmel/expo-audio-stream 0.4.2 → 0.6.0

package/build/pipeline/types.d.ts

@@ -103,6 +103,19 @@ export interface PipelineUnderrunEvent {
 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>;
 /** Payload for `PipelineAudioFocusResumed` (empty — presence is the signal). */
@@ -121,6 +134,7 @@ export interface PipelineEventMap {
     PipelineZombieDetected: PipelineZombieDetectedEvent;
     PipelineUnderrun: PipelineUnderrunEvent;
     PipelineDrained: PipelineDrainedEvent;
+    PipelinePlaybackStopped: PipelinePlaybackStoppedEvent;
     PipelineAudioFocusLost: PipelineAudioFocusLostEvent;
     PipelineAudioFocusResumed: PipelineAudioFocusResumedEvent;
     PipelineFrequencyBands: PipelineFrequencyBandsEvent;

package/build/pipeline/types.d.ts.map

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package/build/pipeline/types.js.map

@@ -1 +1 @@
-{"version":3,"file":"types.js","sourceRoot":"","sources":["../../src/pipeline/types.ts"],"names":[],"mappings":"AAAA,+EAA+E;AAC/E,8CAA8C;AAC9C,+EAA+E","sourcesContent":["// ────────────────────────────────────────────────────────────────────────────\n// Native Audio Pipeline — V3 TypeScript Types\n// ────────────────────────────────────────────────────────────────────────────\n\nimport { PlaybackMode, FrequencyBandConfig, FrequencyBands } from \"../types\";\n\n// ── Connect ─────────────────────────────────────────────────────────────────\n\n/**\n * How the pipeline's playback should coexist with other audio on the device.\n *\n * - `'mixWithOthers'` (default): plays alongside other apps without\n *   interrupting them. On Android no audio focus is requested. Best for\n *   sound effects and short clips.\n * - `'duckOthers'`: requests audio focus with ducking. Other apps lower\n *   their volume but keep playing.\n * - `'doNotMix'`: requests exclusive audio focus. Other apps pause.\n */\nexport type PipelineAudioMode = 'mixWithOthers' | 'duckOthers' | 'doNotMix';\n\n/** Options passed to `connectPipeline()`. */\nexport interface ConnectPipelineOptions {\n  /** Sample rate in Hz (default 24000). */\n  sampleRate?: number;\n  /** Number of channels — 1 = mono, 2 = stereo (default 1). */\n  channelCount?: number;\n  /**\n   * How many ms of audio to accumulate in the jitter buffer before the\n   * priming gate opens and audio begins playing (default 80).\n   */\n  targetBufferMs?: number;\n  /**\n   * Playback mode hint for native optimizations. Affects thread priority and\n   */\n  playbackMode?: PlaybackMode;\n  /** Interval in ms for PipelineFrequencyBands events (default 100). */\n  frequencyBandIntervalMs?: number;\n  /** Optional frequency band crossover configuration. */\n  frequencyBandConfig?: FrequencyBandConfig;\n  /**\n   * How pipeline playback should coexist with other apps' audio.\n   * Default is `'mixWithOthers'` (matches expo-audio).\n   *\n   * Note: this is a **behavior change** vs. prior versions of this library,\n   * which effectively used `'doNotMix'`. Pass `'doNotMix'` explicitly to\n   * preserve that old behavior.\n   */\n  audioMode?: PipelineAudioMode;\n}\n\n/** Result returned from a successful `connectPipeline()` call. */\nexport interface ConnectPipelineResult {\n  sampleRate: number;\n  channelCount: number;\n  targetBufferMs: number;\n  /**\n   * Frame size in samples derived from the device HAL's\n   * `AudioTrack.getMinBufferSize()`. Useful for understanding the write\n   * granularity on the native side.\n   */\n  frameSizeSamples: number;\n}\n\n// ── Push Audio ──────────────────────────────────────────────────────────────\n\n/** Options passed to `pushPipelineAudio()` / `pushPipelineAudioSync()`. */\nexport interface PushPipelineAudioOptions {\n  /** Base64-encoded PCM 16-bit signed LE audio data. */\n  audio: string;\n  /** Conversation turn identifier. */\n  turnId: string;\n  /** True if this is the first chunk of a new turn (resets jitter buffer). */\n  isFirstChunk?: boolean;\n  /** True if this is the final chunk of the current turn (marks end-of-stream). */\n  isLastChunk?: boolean;\n}\n\n// ── Invalidate Turn ─────────────────────────────────────────────────────────\n\n/** Options passed to `invalidatePipelineTurn()`. */\nexport interface InvalidatePipelineTurnOptions {\n  /** The new turn identifier — stale audio for the old turn is discarded. */\n  turnId: string;\n}\n\n// ── State ───────────────────────────────────────────────────────────────────\n\n/**\n * Pipeline states reported via `PipelineStateChanged` events.\n *\n * - `idle`       — connected but no audio flowing\n * - `connecting` — AudioTrack being created, focus being requested\n * - `streaming`  — actively receiving and playing audio\n * - `draining`   — end-of-stream marked, playing remaining buffer\n * - `error`      — unrecoverable error (zombie, write failure, etc.)\n */\nexport type PipelineState =\n  | 'idle'\n  | 'connecting'\n  | 'streaming'\n  | 'draining'\n  | 'error';\n\n// ── Events ──────────────────────────────────────────────────────────────────\n\n/** Payload for `PipelineStateChanged`. */\nexport interface PipelineStateChangedEvent {\n  state: PipelineState;\n}\n\n/** Payload for `PipelinePlaybackStarted`. */\nexport interface PipelinePlaybackStartedEvent {\n  turnId: string;\n}\n\n/** Payload for `PipelineError`. */\nexport interface PipelineErrorEvent {\n  code: string;\n  message: string;\n}\n\n/** Payload for `PipelineZombieDetected`. */\nexport interface PipelineZombieDetectedEvent {\n  playbackHead: number;\n  stalledMs: number;\n}\n\n/** Payload for `PipelineUnderrun`. */\nexport interface PipelineUnderrunEvent {\n  count: number;\n}\n\n/** Payload for `PipelineDrained`. */\nexport interface PipelineDrainedEvent {\n  turnId: string;\n}\n\n/** Payload for `PipelineAudioFocusLost` (empty — presence is the signal). */\nexport type PipelineAudioFocusLostEvent = Record<string, never>;\n\n/** Payload for `PipelineAudioFocusResumed` (empty — presence is the signal). */\nexport type PipelineAudioFocusResumedEvent = Record<string, never>;\n\n/** Payload for `PipelineFrequencyBands`. */\nexport interface PipelineFrequencyBandsEvent extends FrequencyBands {}\n\n/**\n * Map of all pipeline event names to their payload types.\n * Used with `Pipeline.subscribe<K>()` for type-safe event subscriptions.\n */\nexport interface PipelineEventMap {\n  PipelineStateChanged: PipelineStateChangedEvent;\n  PipelinePlaybackStarted: PipelinePlaybackStartedEvent;\n  PipelineError: PipelineErrorEvent;\n  PipelineZombieDetected: PipelineZombieDetectedEvent;\n  PipelineUnderrun: PipelineUnderrunEvent;\n  PipelineDrained: PipelineDrainedEvent;\n  PipelineAudioFocusLost: PipelineAudioFocusLostEvent;\n  PipelineAudioFocusResumed: PipelineAudioFocusResumedEvent;\n  PipelineFrequencyBands: PipelineFrequencyBandsEvent;\n}\n\n/** Union of all pipeline event name strings. */\nexport type PipelineEventName = keyof PipelineEventMap;\n\n// ── Telemetry ───────────────────────────────────────────────────────────────\n\n/** Jitter buffer telemetry counters. */\nexport interface PipelineBufferTelemetry {\n  /** Current buffer level in milliseconds. */\n  bufferMs: number;\n  /** Current buffer level in samples. */\n  bufferSamples: number;\n  /** Whether the priming gate has opened. */\n  primed: boolean;\n  /** Total samples written by the producer since last reset. */\n  totalWritten: number;\n  /** Total samples read by the consumer since last reset. */\n  totalRead: number;\n  /** Number of underrun events. */\n  underrunCount: number;\n  /** Peak buffer level in samples. */\n  peakLevel: number;\n}\n\n/** Full pipeline telemetry snapshot. */\nexport interface PipelineTelemetry extends PipelineBufferTelemetry {\n  /** Current pipeline state. */\n  state: PipelineState;\n  /** Total pushAudio/pushAudioSync calls since connect. */\n  totalPushCalls: number;\n  /** Total bytes pushed since connect. */\n  totalPushBytes: number;\n  /** Total write-loop iterations since connect. */\n  totalWriteLoops: number;\n  /** Current turn identifier. */\n  turnId: string;\n}\n"]}
+{"version":3,"file":"types.js","sourceRoot":"","sources":["../../src/pipeline/types.ts"],"names":[],"mappings":"AAAA,+EAA+E;AAC/E,8CAA8C;AAC9C,+EAA+E","sourcesContent":["// ────────────────────────────────────────────────────────────────────────────\n// Native Audio Pipeline — V3 TypeScript Types\n// ────────────────────────────────────────────────────────────────────────────\n\nimport { PlaybackMode, FrequencyBandConfig, FrequencyBands } from \"../types\";\n\n// ── Connect ─────────────────────────────────────────────────────────────────\n\n/**\n * How the pipeline's playback should coexist with other audio on the device.\n *\n * - `'mixWithOthers'` (default): plays alongside other apps without\n *   interrupting them. On Android no audio focus is requested. Best for\n *   sound effects and short clips.\n * - `'duckOthers'`: requests audio focus with ducking. Other apps lower\n *   their volume but keep playing.\n * - `'doNotMix'`: requests exclusive audio focus. Other apps pause.\n */\nexport type PipelineAudioMode = 'mixWithOthers' | 'duckOthers' | 'doNotMix';\n\n/** Options passed to `connectPipeline()`. */\nexport interface ConnectPipelineOptions {\n  /** Sample rate in Hz (default 24000). */\n  sampleRate?: number;\n  /** Number of channels — 1 = mono, 2 = stereo (default 1). */\n  channelCount?: number;\n  /**\n   * How many ms of audio to accumulate in the jitter buffer before the\n   * priming gate opens and audio begins playing (default 80).\n   */\n  targetBufferMs?: number;\n  /**\n   * Playback mode hint for native optimizations. Affects thread priority and\n   */\n  playbackMode?: PlaybackMode;\n  /** Interval in ms for PipelineFrequencyBands events (default 100). */\n  frequencyBandIntervalMs?: number;\n  /** Optional frequency band crossover configuration. */\n  frequencyBandConfig?: FrequencyBandConfig;\n  /**\n   * How pipeline playback should coexist with other apps' audio.\n   * Default is `'mixWithOthers'` (matches expo-audio).\n   *\n   * Note: this is a **behavior change** vs. prior versions of this library,\n   * which effectively used `'doNotMix'`. Pass `'doNotMix'` explicitly to\n   * preserve that old behavior.\n   */\n  audioMode?: PipelineAudioMode;\n}\n\n/** Result returned from a successful `connectPipeline()` call. */\nexport interface ConnectPipelineResult {\n  sampleRate: number;\n  channelCount: number;\n  targetBufferMs: number;\n  /**\n   * Frame size in samples derived from the device HAL's\n   * `AudioTrack.getMinBufferSize()`. Useful for understanding the write\n   * granularity on the native side.\n   */\n  frameSizeSamples: number;\n}\n\n// ── Push Audio ──────────────────────────────────────────────────────────────\n\n/** Options passed to `pushPipelineAudio()` / `pushPipelineAudioSync()`. */\nexport interface PushPipelineAudioOptions {\n  /** Base64-encoded PCM 16-bit signed LE audio data. */\n  audio: string;\n  /** Conversation turn identifier. */\n  turnId: string;\n  /** True if this is the first chunk of a new turn (resets jitter buffer). */\n  isFirstChunk?: boolean;\n  /** True if this is the final chunk of the current turn (marks end-of-stream). */\n  isLastChunk?: boolean;\n}\n\n// ── Invalidate Turn ─────────────────────────────────────────────────────────\n\n/** Options passed to `invalidatePipelineTurn()`. */\nexport interface InvalidatePipelineTurnOptions {\n  /** The new turn identifier — stale audio for the old turn is discarded. */\n  turnId: string;\n}\n\n// ── State ───────────────────────────────────────────────────────────────────\n\n/**\n * Pipeline states reported via `PipelineStateChanged` events.\n *\n * - `idle`       — connected but no audio flowing\n * - `connecting` — AudioTrack being created, focus being requested\n * - `streaming`  — actively receiving and playing audio\n * - `draining`   — end-of-stream marked, playing remaining buffer\n * - `error`      — unrecoverable error (zombie, write failure, etc.)\n */\nexport type PipelineState =\n  | 'idle'\n  | 'connecting'\n  | 'streaming'\n  | 'draining'\n  | 'error';\n\n// ── Events ──────────────────────────────────────────────────────────────────\n\n/** Payload for `PipelineStateChanged`. */\nexport interface PipelineStateChangedEvent {\n  state: PipelineState;\n}\n\n/** Payload for `PipelinePlaybackStarted`. */\nexport interface PipelinePlaybackStartedEvent {\n  turnId: string;\n}\n\n/** Payload for `PipelineError`. */\nexport interface PipelineErrorEvent {\n  code: string;\n  message: string;\n}\n\n/** Payload for `PipelineZombieDetected`. */\nexport interface PipelineZombieDetectedEvent {\n  playbackHead: number;\n  stalledMs: number;\n}\n\n/** Payload for `PipelineUnderrun`. */\nexport interface PipelineUnderrunEvent {\n  count: number;\n}\n\n/** Payload for `PipelineDrained`. */\nexport interface PipelineDrainedEvent {\n  turnId: string;\n}\n\n/**\n * Payload for `PipelinePlaybackStopped`.\n *\n * Fired when the last sample physically leaves the speaker, approximately\n * `outputLatencyMs` after `PipelineDrained` for the same turn. Pairs with\n * `PipelinePlaybackStarted` (start-of-emission ↔ end-of-emission).\n *\n * Note: this is a physical-world milestone, distinct from `state: 'idle'`\n * (the pipeline-state-machine value reported via `PipelineStateChanged`).\n */\nexport interface PipelinePlaybackStoppedEvent {\n  turnId: string;\n}\n\n/** Payload for `PipelineAudioFocusLost` (empty — presence is the signal). */\nexport type PipelineAudioFocusLostEvent = Record<string, never>;\n\n/** Payload for `PipelineAudioFocusResumed` (empty — presence is the signal). */\nexport type PipelineAudioFocusResumedEvent = Record<string, never>;\n\n/** Payload for `PipelineFrequencyBands`. */\nexport interface PipelineFrequencyBandsEvent extends FrequencyBands {}\n\n/**\n * Map of all pipeline event names to their payload types.\n * Used with `Pipeline.subscribe<K>()` for type-safe event subscriptions.\n */\nexport interface PipelineEventMap {\n  PipelineStateChanged: PipelineStateChangedEvent;\n  PipelinePlaybackStarted: PipelinePlaybackStartedEvent;\n  PipelineError: PipelineErrorEvent;\n  PipelineZombieDetected: PipelineZombieDetectedEvent;\n  PipelineUnderrun: PipelineUnderrunEvent;\n  PipelineDrained: PipelineDrainedEvent;\n  PipelinePlaybackStopped: PipelinePlaybackStoppedEvent;\n  PipelineAudioFocusLost: PipelineAudioFocusLostEvent;\n  PipelineAudioFocusResumed: PipelineAudioFocusResumedEvent;\n  PipelineFrequencyBands: PipelineFrequencyBandsEvent;\n}\n\n/** Union of all pipeline event name strings. */\nexport type PipelineEventName = keyof PipelineEventMap;\n\n// ── Telemetry ───────────────────────────────────────────────────────────────\n\n/** Jitter buffer telemetry counters. */\nexport interface PipelineBufferTelemetry {\n  /** Current buffer level in milliseconds. */\n  bufferMs: number;\n  /** Current buffer level in samples. */\n  bufferSamples: number;\n  /** Whether the priming gate has opened. */\n  primed: boolean;\n  /** Total samples written by the producer since last reset. */\n  totalWritten: number;\n  /** Total samples read by the consumer since last reset. */\n  totalRead: number;\n  /** Number of underrun events. */\n  underrunCount: number;\n  /** Peak buffer level in samples. */\n  peakLevel: number;\n}\n\n/** Full pipeline telemetry snapshot. */\nexport interface PipelineTelemetry extends PipelineBufferTelemetry {\n  /** Current pipeline state. */\n  state: PipelineState;\n  /** Total pushAudio/pushAudioSync calls since connect. */\n  totalPushCalls: number;\n  /** Total bytes pushed since connect. */\n  totalPushBytes: number;\n  /** Total write-loop iterations since connect. */\n  totalWriteLoops: number;\n  /** Current turn identifier. */\n  turnId: string;\n}\n"]}

package/build/types.d.ts

@@ -103,9 +103,30 @@ 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;
+    /**
+     * True when the recording session has stopped and the caller must call
+     * stopMicrophone() and reconnect to resume. False for transient conditions
+     * where recording continues (READ_ERROR on iOS) or the library will
+     * auto-recover (INTERRUPTED + autoResuming: true).
+     */
+    isFatal: boolean;
+    /**
+     * True only when code is INTERRUPTED — the library is waiting for the
+     * system to return the audio session and will reinstall the tap
+     * automatically. Always false when isFatal is true.
+     */
+    autoResuming: boolean;
+}
 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   * True when the recording session has stopped and the caller must call\n   * stopMicrophone() and reconnect to resume. False for transient conditions\n   * where recording continues (READ_ERROR on iOS) or the library will\n   * auto-recover (INTERRUPTED + autoResuming: true).\n   */\n  isFatal: boolean;\n  /**\n   * True only when code is INTERRUPTED — the library is waiting for the\n   * system to return the audio session and will reinstall the tap\n   * automatically. Always false when isFatal is true.\n   */\n  autoResuming: boolean;\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

@@ -3,6 +3,7 @@ import AVFoundation
 import ExpoModulesCore
 
 let audioDataEvent: String = "AudioData"
+let microphoneErrorEvent: String = "MicrophoneError"
 let deviceReconnectedEvent: String = "DeviceReconnected"
 
 
@@ -17,6 +18,7 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
         if _microphone == nil {
             _microphone = Microphone()
             _microphone?.delegate = self
+            _microphone?.sharedAudioEngine = sharedAudioEngine
         }
         return _microphone!
     }
@@ -41,6 +43,7 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
         // Defines event names that the module can send to JavaScript.
         Events([
             audioDataEvent,
+            microphoneErrorEvent,
             deviceReconnectedEvent,
             PipelineIntegration.EVENT_STATE_CHANGED,
             PipelineIntegration.EVENT_PLAYBACK_STARTED,
@@ -48,12 +51,16 @@ 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,
         ])
 
         AsyncFunction("destroy") { (promise: Promise) in
+            // Stop the microphone before tearing down the engine so the tap is
+            // cleanly removed and the capture callback cannot fire on a dead node.
+            self._microphone?.stopRecording(resolver: nil)
             self._pipelineIntegration?.destroy()
             self._pipelineIntegration = nil
             self.sharedAudioEngine.teardown()
@@ -119,13 +126,17 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
                 pointsPerSecond: nil
             )
 
-            if !isAudioSessionInitialized {
-                do {
+            do {
+                if !isAudioSessionInitialized {
                     try ensureAudioSessionInitialized(settings: settings)
-                } catch {
-                    promise.reject("ERROR", "Failed to init audio session \(error.localizedDescription)")
-                    return
                 }
+                // Ensure the shared engine is configured so AEC is applied to mic audio.
+                if !self.sharedAudioEngine.isConfigured {
+                    try self.sharedAudioEngine.configure(playbackMode: .conversation)
+                }
+            } catch {
+                promise.reject("ERROR", "Failed to init audio session: \(error.localizedDescription)")
+                return
             }
 
             if let result = self.microphone.startRecording(settings: settings, intervalMilliseconds: interval, frequencyBandConfig: fbConfig) {
@@ -209,6 +220,15 @@ 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 +268,10 @@ 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 {
@@ -320,13 +344,20 @@ public class ExpoPlayAudioStreamModule: Module, MicrophoneDataDelegate, Pipeline
         sendEvent(audioDataEvent, eventBody)
     }
 
-    func onMicrophoneError(_ error: String, _ errorMessage: String) {
-        let eventBody: [String: Any] = [
-            "error": error,
-            "errorMessage": errorMessage,
-            "streamUuid": ""
-        ]
-        sendEvent(audioDataEvent, eventBody)
+    func onMicrophoneError(_ error: MicrophoneErrorInfo) {
+        // Rich structured channel for new consumers
+        sendEvent(microphoneErrorEvent, [
+            "code": error.code,
+            "message": error.message,
+            "isFatal": error.isFatal,
+            "autoResuming": error.autoResuming,
+        ])
+        // Backward-compat: keep the error variant on AudioData for existing consumers
+        sendEvent(audioDataEvent, [
+            "error": error.code,
+            "errorMessage": error.message,
+            "streamUuid": "",
+        ])
     }
 
     func onDeviceReconnected(_ reason: AVAudioSession.RouteChangeReason) {