@edkimmel/expo-audio-stream

Native audio recording and low-latency playback for Expo/React Native. Designed for real-time voice AI applications: microphone capture, chunked PCM playback, and a jitter-buffered native pipeline for streaming audio from AI backends.

Install

npx expo install @edkimmel/expo-audio-stream

Quick Start

Microphone Recording

import { ExpoPlayAudioStream } from "@edkimmel/expo-audio-stream";

const { recordingResult, subscription } =
  await ExpoPlayAudioStream.startMicrophone({
    sampleRate: 16000,
    channels: 1,
    encoding: "pcm_16bit",
    interval: 100,
    onAudioStream: async (event) => {
      // event.data: base64-encoded PCM chunk
      // event.soundLevel: current mic level (dB)
      // event.frequencyBands: { low, mid, high } (0–1) if configured
      sendToBackend(event.data);
    },
    frequencyBandConfig: {
      lowCrossoverHz: 300,
      highCrossoverHz: 2000,
    },
  });

// Later:
await ExpoPlayAudioStream.stopMicrophone();
subscription?.remove();

Chunked Playback (playSound)

For playing base64-encoded PCM audio in a queue with turn management:

import {
  ExpoPlayAudioStream,
  EncodingTypes,
} from "@edkimmel/expo-audio-stream";

await ExpoPlayAudioStream.setSoundConfig({
  sampleRate: 24000,
  playbackMode: "conversation",
});

// Enqueue chunks as they arrive
await ExpoPlayAudioStream.playSound(
  base64Chunk,
  "turn-1",
  EncodingTypes.PCM_S16LE
);

// Listen for playback completion
const sub = ExpoPlayAudioStream.subscribeToSoundChunkPlayed(async (e) => {
  if (e.isFinal) console.log("Turn finished playing");
});

Native Pipeline (recommended for AI voice streaming)

The Pipeline class provides jitter-buffered, low-latency playback with a native write thread. Use this for streaming audio from AI backends over WebSockets.

import { Pipeline } from "@edkimmel/expo-audio-stream";

// Connect with desired config
const result = await Pipeline.connect({
  sampleRate: 24000,
  channelCount: 1,
  targetBufferMs: 80,
  frequencyBandIntervalMs: 100, // optional: emit frequency bands every 100ms
});

// Subscribe to events
const errorSub = Pipeline.onError((err) => {
  console.error(`Pipeline error: ${err.code} - ${err.message}`);
});

const focusSub = Pipeline.onAudioFocus(({ focused }) => {
  if (!focused) {
    // Another app took audio focus; re-request audio on regain
  }
});

// Hot path: push audio synchronously from WebSocket handler
ws.onmessage = (msg) => {
  Pipeline.pushAudioSync({
    audio: msg.data, // base64 PCM16 LE
    turnId: currentTurnId,
    isFirstChunk: isFirst,
    isLastChunk: isLast,
  });
};

// On new turn, invalidate stale audio
Pipeline.invalidateTurn({ turnId: newTurnId });

// Tear down
await Pipeline.disconnect();
errorSub.remove();
focusSub.remove();

API Reference

ExpoPlayAudioStream

All methods are static.

Lifecycle

| Method | Returns | Description | |--------|---------|-------------| | destroy() | void | Release all resources. Resets internal state on both platforms. |

Permissions

| Method | Returns | Description | |--------|---------|-------------| | requestPermissionsAsync() | Promise<PermissionResult> | Prompt the user for microphone permission. | | getPermissionsAsync() | Promise<PermissionResult> | Check the current microphone permission status. |

Microphone

| Method | Returns | Description | |--------|---------|-------------| | startMicrophone(config) | Promise<{ recordingResult, subscription? }> | Start mic capture. Audio is delivered as base64 PCM via onAudioStream or subscribeToAudioEvents. | | stopMicrophone() | Promise<AudioRecording \| null> | Stop mic capture and return recording metadata. | | toggleSilence(isSilent) | void | Mute/unmute the mic stream without stopping the session. Silenced frames are zero-filled. | | promptMicrophoneModes() | void | (iOS only) Show the system voice isolation picker (iOS 15+). |

Sound Playback

| Method | Returns | Description | |--------|---------|-------------| | playSound(audio, turnId, encoding?) | Promise<void> | Enqueue a base64 PCM chunk for playback. | | stopSound() | Promise<void> | Stop playback and clear the queue. | | setSoundConfig(config) | Promise<void> | Update playback sample rate and mode. |

Event Subscriptions

| Method | Returns | Description | |--------|---------|-------------| | subscribeToAudioEvents(callback) | Subscription | Receive AudioDataEvent during mic capture. | | subscribeToSoundChunkPlayed(callback) | Subscription | Notified when a chunk finishes playing. isFinal is true when the queue drains. | | subscribe(eventName, callback) | Subscription | Generic event listener for any module event. |

Pipeline

All methods are static. The pipeline manages its own native write thread, jitter buffer, and audio focus.

Lifecycle

| Method | Returns | Description | |--------|---------|-------------| | connect(options?) | Promise<ConnectPipelineResult> | Create the native audio track, jitter buffer, and write thread. Config is immutable per session. | | disconnect() | Promise<void> | Tear down the pipeline and release all native resources. |

Audio Push

| Method | Returns | Description | |--------|---------|-------------| | pushAudio(options) | Promise<void> | Push base64 PCM16 LE audio (async, with error propagation). | | pushAudioSync(options) | boolean | Push audio synchronously. No Promise overhead -- use in WebSocket onmessage for minimum latency. Returns false on failure. |

Turn Management

| Method | Returns | Description | |--------|---------|-------------| | invalidateTurn(options) | Promise<void> | Discard buffered audio for the old turn. The jitter buffer is reset. |

State & Telemetry

| Method | Returns | Description | |--------|---------|-------------| | getState() | PipelineState | Current state: idle, connecting, streaming, draining, or error. | | getTelemetry() | PipelineTelemetry | Snapshot of buffer levels, push counts, write loops, underruns, etc. |

Event Subscriptions

| Method | Returns | Description | |--------|---------|-------------| | subscribe(eventName, listener) | EventSubscription | Type-safe subscription to any pipeline event. | | onError(listener) | { remove } | Convenience: handles both PipelineError and PipelineZombieDetected. | | onAudioFocus(listener) | { remove } | Convenience: { focused: true/false } on audio focus changes. |

Configuration Types

RecordingConfig

interface RecordingConfig {
  sampleRate?: 16000 | 24000 | 44100 | 48000;
  channels?: 1 | 2;
  encoding?: "pcm_32bit" | "pcm_16bit" | "pcm_8bit";
  interval?: number; // ms between audio data emissions (default 1000)
  onAudioStream?: (event: AudioDataEvent) => Promise<void>;
  frequencyBandConfig?: FrequencyBandConfig; // enable frequency band analysis on mic audio
}

SoundConfig

interface SoundConfig {
  sampleRate?: 16000 | 24000 | 44100 | 48000;
  playbackMode?: "regular" | "voiceProcessing" | "conversation";
  useDefault?: boolean; // reset to defaults
}

ConnectPipelineOptions

interface ConnectPipelineOptions {
  sampleRate?: number;              // default 24000
  channelCount?: number;            // default 1 (mono)
  targetBufferMs?: number;          // ms to buffer before priming gate opens (default 80)
  frequencyBandIntervalMs?: number; // emit PipelineFrequencyBands every N ms (omit to disable)
  frequencyBandConfig?: FrequencyBandConfig; // crossover frequencies (optional)
}

PushPipelineAudioOptions

interface PushPipelineAudioOptions {
  audio: string;           // base64-encoded PCM 16-bit signed LE
  turnId: string;
  isFirstChunk?: boolean;  // resets jitter buffer
  isLastChunk?: boolean;   // marks end-of-stream, begins drain
}

FrequencyBandConfig

interface FrequencyBandConfig {
  lowCrossoverHz?: number;  // boundary between low and mid bands (default 300)
  highCrossoverHz?: number; // boundary between mid and high bands (default 2000)
}

FrequencyBands

interface FrequencyBands {
  low: number;  // 0–1, dB-scaled RMS energy below lowCrossoverHz
  mid: number;  // 0–1, dB-scaled RMS energy between crossovers
  high: number; // 0–1, dB-scaled RMS energy above highCrossoverHz
}

Events

Core Events

| Event | Payload | Description | |-------|---------|-------------| | AudioData | { encoded, position, deltaSize, totalSize, soundLevel, frequencyBands?, ... } | Emitted during mic capture at the configured interval. Includes frequencyBands when frequencyBandConfig is set. | | SoundChunkPlayed | { isFinal: boolean } | A queued chunk finished playing. isFinal when the queue is empty. | | SoundStarted | (none) | Playback began for a new turn. | | DeviceReconnected | { reason } | Audio route changed (headphones, Bluetooth, etc). |

Pipeline Events

| Event | Payload | Description | |-------|---------|-------------| | PipelineStateChanged | { state } | Pipeline state transition. | | PipelinePlaybackStarted | { turnId } | Priming gate opened, audio is now audible. | | PipelineError | { code, message } | Non-recoverable error. | | PipelineZombieDetected | { playbackHead, stalledMs } | Audio track stalled. | | PipelineUnderrun | { count } | Jitter buffer underrun (silence inserted). | | PipelineDrained | { turnId } | All buffered audio for the turn has been played. | | PipelineFrequencyBands | { low, mid, high } | Frequency band energy (0–1) emitted at frequencyBandIntervalMs. | | PipelineAudioFocusLost | (empty) | Another app took audio focus. | | PipelineAudioFocusResumed | (empty) | Audio focus regained. |

Constants

import {
  EncodingTypes,           // { PCM_F32LE: "pcm_f32le", PCM_S16LE: "pcm_s16le" }
  PlaybackModes,           // { REGULAR, VOICE_PROCESSING, CONVERSATION }
  AudioEvents,             // { AudioData, SoundChunkPlayed, SoundStarted, DeviceReconnected }
  SuspendSoundEventTurnId, // "suspend-sound-events" -- suppresses playback events
} from "@edkimmel/expo-audio-stream";

Platform Notes

iOS

• Uses AVAudioEngine with AVAudioPlayerNode for sound playback and pipeline audio.
• Microphone capture via AVAudioEngine.inputNode tap.
• Audio session configured as .playAndRecord with .voiceChat mode.
• Voice processing (AEC/noise reduction) available via voiceProcessing and conversation playback modes.
• promptMicrophoneModes() exposes the iOS 15+ system voice isolation picker.

Android

• Uses AudioTrack (float PCM, MODE_STREAM) for sound playback.
• Microphone capture via AudioRecord with VOICE_RECOGNITION source for far-field mic gain.
• AEC, noise suppression, and AGC applied via AudioEffectsManager.

License

MIT