@edkimmel/expo-audio-stream 0.2.0

package/build/pipeline/types.d.ts

@@ -0,0 +1,132 @@
+import { PlaybackMode } from "../types";
+/** Options passed to `connectPipeline()`. */
+export interface ConnectPipelineOptions {
+    /** Sample rate in Hz (default 24000). */
+    sampleRate?: number;
+    /** Number of channels — 1 = mono, 2 = stereo (default 1). */
+    channelCount?: number;
+    /**
+     * How many ms of audio to accumulate in the jitter buffer before the
+     * priming gate opens and audio begins playing (default 80).
+     */
+    targetBufferMs?: number;
+    /**
+     * Playback mode hint for native optimizations. Affects thread priority and
+     */
+    playbackMode?: PlaybackMode;
+}
+/** Result returned from a successful `connectPipeline()` call. */
+export interface ConnectPipelineResult {
+    sampleRate: number;
+    channelCount: number;
+    targetBufferMs: number;
+    /**
+     * Frame size in samples derived from the device HAL's
+     * `AudioTrack.getMinBufferSize()`. Useful for understanding the write
+     * granularity on the native side.
+     */
+    frameSizeSamples: number;
+}
+/** Options passed to `pushPipelineAudio()` / `pushPipelineAudioSync()`. */
+export interface PushPipelineAudioOptions {
+    /** Base64-encoded PCM 16-bit signed LE audio data. */
+    audio: string;
+    /** Conversation turn identifier. */
+    turnId: string;
+    /** True if this is the first chunk of a new turn (resets jitter buffer). */
+    isFirstChunk?: boolean;
+    /** True if this is the final chunk of the current turn (marks end-of-stream). */
+    isLastChunk?: boolean;
+}
+/** Options passed to `invalidatePipelineTurn()`. */
+export interface InvalidatePipelineTurnOptions {
+    /** The new turn identifier — stale audio for the old turn is discarded. */
+    turnId: string;
+}
+/**
+ * Pipeline states reported via `PipelineStateChanged` events.
+ *
+ * - `idle`       — connected but no audio flowing
+ * - `connecting` — AudioTrack being created, focus being requested
+ * - `streaming`  — actively receiving and playing audio
+ * - `draining`   — end-of-stream marked, playing remaining buffer
+ * - `error`      — unrecoverable error (zombie, write failure, etc.)
+ */
+export type PipelineState = 'idle' | 'connecting' | 'streaming' | 'draining' | 'error';
+/** Payload for `PipelineStateChanged`. */
+export interface PipelineStateChangedEvent {
+    state: PipelineState;
+}
+/** Payload for `PipelinePlaybackStarted`. */
+export interface PipelinePlaybackStartedEvent {
+    turnId: string;
+}
+/** Payload for `PipelineError`. */
+export interface PipelineErrorEvent {
+    code: string;
+    message: string;
+}
+/** Payload for `PipelineZombieDetected`. */
+export interface PipelineZombieDetectedEvent {
+    playbackHead: number;
+    stalledMs: number;
+}
+/** Payload for `PipelineUnderrun`. */
+export interface PipelineUnderrunEvent {
+    count: number;
+}
+/** Payload for `PipelineDrained`. */
+export interface PipelineDrainedEvent {
+    turnId: string;
+}
+/** Payload for `PipelineAudioFocusLost` (empty — presence is the signal). */
+export type PipelineAudioFocusLostEvent = Record<string, never>;
+/** Payload for `PipelineAudioFocusResumed` (empty — presence is the signal). */
+export type PipelineAudioFocusResumedEvent = Record<string, never>;
+/**
+ * Map of all pipeline event names to their payload types.
+ * Used with `Pipeline.subscribe<K>()` for type-safe event subscriptions.
+ */
+export interface PipelineEventMap {
+    PipelineStateChanged: PipelineStateChangedEvent;
+    PipelinePlaybackStarted: PipelinePlaybackStartedEvent;
+    PipelineError: PipelineErrorEvent;
+    PipelineZombieDetected: PipelineZombieDetectedEvent;
+    PipelineUnderrun: PipelineUnderrunEvent;
+    PipelineDrained: PipelineDrainedEvent;
+    PipelineAudioFocusLost: PipelineAudioFocusLostEvent;
+    PipelineAudioFocusResumed: PipelineAudioFocusResumedEvent;
+}
+/** Union of all pipeline event name strings. */
+export type PipelineEventName = keyof PipelineEventMap;
+/** Jitter buffer telemetry counters. */
+export interface PipelineBufferTelemetry {
+    /** Current buffer level in milliseconds. */
+    bufferMs: number;
+    /** Current buffer level in samples. */
+    bufferSamples: number;
+    /** Whether the priming gate has opened. */
+    primed: boolean;
+    /** Total samples written by the producer since last reset. */
+    totalWritten: number;
+    /** Total samples read by the consumer since last reset. */
+    totalRead: number;
+    /** Number of underrun events. */
+    underrunCount: number;
+    /** Peak buffer level in samples. */
+    peakLevel: number;
+}
+/** Full pipeline telemetry snapshot. */
+export interface PipelineTelemetry extends PipelineBufferTelemetry {
+    /** Current pipeline state. */
+    state: PipelineState;
+    /** Total pushAudio/pushAudioSync calls since connect. */
+    totalPushCalls: number;
+    /** Total bytes pushed since connect. */
+    totalPushBytes: number;
+    /** Total write-loop iterations since connect. */
+    totalWriteLoops: number;
+    /** Current turn identifier. */
+    turnId: string;
+}
+//# sourceMappingURL=types.d.ts.map

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

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package/build/pipeline/types.js

@@ -0,0 +1,5 @@
+// ────────────────────────────────────────────────────────────────────────────
+// Native Audio Pipeline — V3 TypeScript Types
+// ────────────────────────────────────────────────────────────────────────────
+export {};
+//# sourceMappingURL=types.js.map

package/build/pipeline/types.js.map

@@ -0,0 +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 } from \"../types\";\n\n// ── Connect ─────────────────────────────────────────────────────────────────\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}\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/**\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}\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

@@ -0,0 +1,221 @@
+export type RecordingEncodingType = 'pcm_32bit' | 'pcm_16bit' | 'pcm_8bit';
+export type SampleRate = 16000 | 24000 | 44100 | 48000;
+export type BitDepth = 8 | 16 | 32;
+export declare const PlaybackModes: {
+    readonly REGULAR: "regular";
+    readonly VOICE_PROCESSING: "voiceProcessing";
+    readonly CONVERSATION: "conversation";
+};
+/**
+ * Defines different playback modes for audio processing
+ */
+export type PlaybackMode = (typeof PlaybackModes)[keyof typeof PlaybackModes];
+/**
+ * Configuration for audio playback settings
+ */
+export interface SoundConfig {
+    /**
+     * The sample rate for audio playback in Hz
+     */
+    sampleRate?: SampleRate;
+    /**
+     * The playback mode (regular, voiceProcessing, or conversation)
+     */
+    playbackMode?: PlaybackMode;
+    /**
+     * When true, resets to default configuration regardless of other parameters
+     */
+    useDefault?: boolean;
+    /**
+     * Enable jitter buffering for audio streams
+     */
+    enableBuffering?: boolean;
+    /**
+     * Automatically enable buffering based on network conditions
+     */
+    autoBuffer?: boolean;
+    /**
+     * Configuration for the jitter buffer when enableBuffering is true
+     */
+    bufferConfig?: Partial<IAudioBufferConfig>;
+}
+/**
+ * Configuration for buffered audio streaming
+ */
+export interface BufferedStreamConfig {
+    /**
+     * Turn ID for queue management
+     */
+    turnId: string;
+    /**
+     * Audio encoding format
+     */
+    encoding?: Encoding;
+    /**
+     * Buffer configuration options
+     */
+    bufferConfig?: Partial<IAudioBufferConfig>;
+    /**
+     * Callback for buffer health updates
+     */
+    onBufferHealth?: (metrics: IBufferHealthMetrics) => void;
+}
+export declare const EncodingTypes: {
+    readonly PCM_F32LE: "pcm_f32le";
+    readonly PCM_S16LE: "pcm_s16le";
+};
+/**
+ * Defines different encoding formats for audio data
+ */
+export type Encoding = (typeof EncodingTypes)[keyof typeof EncodingTypes];
+/**
+ * Smart buffering mode options
+ */
+export type SmartBufferMode = 'conservative' | 'balanced' | 'aggressive' | 'adaptive';
+/**
+ * Network condition indicators for smart buffering
+ */
+export interface NetworkConditions {
+    latency?: number;
+    jitter?: number;
+    packetLoss?: number;
+    bandwidth?: number;
+}
+/**
+ * Smart buffering configuration
+ */
+export interface SmartBufferConfig {
+    mode: SmartBufferMode;
+    networkConditions?: NetworkConditions;
+    adaptiveThresholds?: {
+        highLatencyMs?: number;
+        highJitterMs?: number;
+        packetLossPercent?: number;
+    };
+}
+export interface StartRecordingResult {
+    fileUri: string;
+    mimeType: string;
+    channels?: number;
+    bitDepth?: BitDepth;
+    sampleRate?: SampleRate;
+}
+export interface AudioDataEvent {
+    data: string | Float32Array;
+    data16kHz?: string | Float32Array;
+    position: number;
+    fileUri: string;
+    eventDataSize: number;
+    totalSize: number;
+    soundLevel?: number;
+}
+export interface RecordingConfig {
+    sampleRate?: SampleRate;
+    channels?: 1 | 2;
+    encoding?: RecordingEncodingType;
+    interval?: number;
+    enableProcessing?: boolean;
+    pointsPerSecond?: number;
+    onAudioStream?: (event: AudioDataEvent) => Promise<void>;
+}
+export interface Chunk {
+    text: string;
+    timestamp: [number, number | null];
+}
+export interface TranscriberData {
+    id: string;
+    isBusy: boolean;
+    text: string;
+    startTime: number;
+    endTime: number;
+    chunks: Chunk[];
+}
+export interface AudioRecording {
+    fileUri: string;
+    filename: string;
+    durationMs: number;
+    size: number;
+    channels: number;
+    bitDepth: BitDepth;
+    sampleRate: SampleRate;
+    mimeType: string;
+    transcripts?: TranscriberData[];
+    wavPCMData?: Float32Array;
+}
+/**
+ * Configuration for audio buffer management
+ */
+export interface IAudioBufferConfig {
+    targetBufferMs: number;
+    minBufferMs: number;
+    maxBufferMs: number;
+    frameIntervalMs: number;
+}
+/**
+ * Audio payload for playback containing base64 encoded audio data
+ */
+export interface IAudioPlayPayload {
+    audioData: string;
+    isFirst?: boolean;
+    isFinal?: boolean;
+}
+/**
+ * Processed audio frame with metadata
+ */
+export interface IAudioFrame {
+    sequenceNumber: number;
+    data: IAudioPlayPayload;
+    duration: number;
+    timestamp: number;
+}
+/**
+ * Buffer health states for quality monitoring
+ */
+export type BufferHealthState = 'idle' | 'healthy' | 'degraded' | 'critical';
+/**
+ * Comprehensive buffer health and quality metrics
+ */
+export interface IBufferHealthMetrics {
+    currentBufferMs: number;
+    targetBufferMs: number;
+    underrunCount: number;
+    overrunCount: number;
+    averageJitter: number;
+    bufferHealthState: BufferHealthState;
+    adaptiveAdjustmentsCount: number;
+}
+/**
+ * Interface for audio buffer management
+ */
+export interface IAudioBufferManager {
+    enqueueFrames(audioData: IAudioPlayPayload): void;
+    startPlayback(): void;
+    stopPlayback(): void;
+    isPlaying(): boolean;
+    getHealthMetrics(): IBufferHealthMetrics;
+    updateConfig(config: Partial<IAudioBufferConfig>): void;
+    applyAdaptiveAdjustments(): void;
+    destroy(): void;
+    getCurrentBufferMs(): number;
+}
+/**
+ * Interface for frame processing
+ */
+export interface IFrameProcessor {
+    parseChunk(payload: IAudioPlayPayload): IAudioFrame[];
+    reset(): void;
+}
+/**
+ * Interface for quality monitoring
+ */
+export interface IQualityMonitor {
+    recordFrameArrival(timestamp: number): void;
+    recordUnderrun(): void;
+    recordOverrun(): void;
+    updateBufferLevel(bufferMs: number): void;
+    getMetrics(): IBufferHealthMetrics;
+    getBufferHealthState(isPlaying: boolean, currentLatencyMs: number): BufferHealthState;
+    getRecommendedAdjustment(): number;
+    reset(): void;
+}
+//# sourceMappingURL=types.d.ts.map

package/build/types.d.ts.map

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package/build/types.js

@@ -0,0 +1,10 @@
+export const PlaybackModes = {
+    REGULAR: 'regular',
+    VOICE_PROCESSING: 'voiceProcessing',
+    CONVERSATION: 'conversation',
+};
+export const EncodingTypes = {
+    PCM_F32LE: 'pcm_f32le',
+    PCM_S16LE: 'pcm_s16le',
+};
+//# sourceMappingURL=types.js.map

package/build/types.js.map

@@ -0,0 +1 @@
+{"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;AAmEX,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 audio playback settings\n */\nexport interface SoundConfig {\n  /**\n   * The sample rate for audio playback in Hz\n   */\n  sampleRate?: SampleRate;\n\n  /**\n   * The playback mode (regular, voiceProcessing, or conversation)\n   */\n  playbackMode?: PlaybackMode;\n\n  /**\n   * When true, resets to default configuration regardless of other parameters\n   */\n  useDefault?: boolean;\n\n  /**\n   * Enable jitter buffering for audio streams\n   */\n  enableBuffering?: boolean;\n\n  /**\n   * Automatically enable buffering based on network conditions\n   */\n  autoBuffer?: boolean;\n\n  /**\n   * Configuration for the jitter buffer when enableBuffering is true\n   */\n  bufferConfig?: Partial<IAudioBufferConfig>;\n}\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/**\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}\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}\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/expo-module.config.json

@@ -0,0 +1,9 @@
+{
+  "platforms": ["ios", "android"],
+  "ios": {
+    "modules": ["ExpoPlayAudioStreamModule"]
+  },
+  "android": {
+    "modules": ["expo.modules.audiostream.ExpoPlayAudioStreamModule"]
+  }
+}