createStreamingSTT()

Create a streaming STT engine instance for real-time recognition with partial results and endpoint detection. This is ideal for live microphone input and conversational interfaces. For batch transcription of complete audio files, use createSTT() instead.

function createStreamingSTT(
  options: StreamingSttInitOptions
): Promise<StreamingSttEngine>

Parameters

options

StreamingSttInitOptions

required

Streaming STT initialization options.

Show StreamingSttInitOptions properties

modelPath

ModelPathConfig

required

Model directory path configuration.

modelType

OnlineSTTModelType | 'auto'

required

Online model type. Options:

'transducer': Transducer models (e.g., Zipformer)
'paraformer': Paraformer streaming models
'zipformer2_ctc': Zipformer2 CTC models
'nemo_ctc': NeMo CTC models
'tone_ctc': Tone CTC models
'auto': Detect from model directory (recommended)

enableEndpoint

boolean

default:true

Enable endpoint (end of utterance) detection.

endpointConfig

EndpointConfig

Endpoint detection rules. Defaults match Kotlin implementation.

Show EndpointConfig structure

rule1

EndpointRule

Rule 1: e.g., 2.4s trailing silence, no speech required.

rule2

EndpointRule

Rule 2: e.g., 1.4s trailing silence, speech required.

rule3

EndpointRule

Rule 3: e.g., max utterance length 20s.

decodingMethod

'greedy_search' | 'modified_beam_search'

default:"greedy_search"

Decoding strategy.

maxActivePaths

number

default:4

Max active paths for beam search.

hotwordsFile

string

Path to hotwords file (transducer/nemo_transducer only).

hotwordsScore

number

Hotwords score/weight.

numThreads

number

default:1

Number of threads for inference.

provider

string

Execution provider (e.g., 'cpu', 'coreml', 'xnnpack').

ruleFsts

string

Path(s) to rule FSTs for ITN.

ruleFars

string

Path(s) to rule FARs for ITN.

blankPenalty

number

Blank penalty for CTC models.

debug

boolean

default:false

Enable debug logging.

enableInputNormalization

boolean

default:true

Enable adaptive input normalization in processAudioChunk(). When true, audio is scaled so peak is ~0.8 to handle varying device levels. Set false if your audio is already normalized.

Returns

Promise<StreamingSttEngine>

StreamingSttEngine

A streaming STT engine instance.

Show StreamingSttEngine interface

instanceId

string

Unique instance identifier.

createStream

(hotwords?: string) => Promise<SttStream>

Create a new recognition stream. Optional hotwords string.

destroy

() => Promise<void>

Release native resources. Must be called when done.

SttStream Interface

The stream object returned by engine.createStream() provides methods for feeding audio and getting recognition results.

SttStream

object

Show methods

streamId

string

Unique stream identifier.

acceptWaveform

(samples: number[], sampleRate: number) => Promise<void>

Feed PCM audio samples (float in [-1, 1]) to the stream.

inputFinished

() => Promise<void>

Signal that no more audio will be fed.

decode

() => Promise<void>

Run decoding on accumulated audio (call when isReady() returns true).

isReady

() => Promise<boolean>

Check if there’s enough audio to decode.

getResult

() => Promise<StreamingSttResult>

Get current partial or final recognition result.

isEndpoint

() => Promise<boolean>

Check if endpoint (end of utterance) was detected.

reset

() => Promise<void>

Reset stream state for reuse.

release

() => Promise<void>

Release native stream resources. Do not use after calling.

processAudioChunk

(samples: number[] | Float32Array, sampleRate: number) => Promise<{ result: StreamingSttResult; isEndpoint: boolean }>

Convenience method: feed audio, auto-decode while ready, return result and endpoint status. Reduces bridge round-trips from 5 to 1 per chunk.

Examples

Basic Real-Time Recognition

import { createStreamingSTT, assetModelPath } from 'react-native-sherpa-onnx/stt';

// Create streaming engine with auto-detection
const engine = await createStreamingSTT({
  modelPath: assetModelPath('models/streaming-zipformer-en'),
  modelType: 'auto',
});

// Create a stream
const stream = await engine.createStream();

// Feed audio chunks (e.g., from microphone)
await stream.acceptWaveform(audioSamples, 16000);

// Check if ready to decode
if (await stream.isReady()) {
  await stream.decode();
  const result = await stream.getResult();
  console.log('Partial result:', result.text);
}

// Check for end of utterance
if (await stream.isEndpoint()) {
  const finalResult = await stream.getResult();
  console.log('Final result:', finalResult.text);
  await stream.reset(); // Ready for next utterance
}

// Clean up
await stream.release();
await engine.destroy();

Simplified with processAudioChunk()

The processAudioChunk() method combines multiple operations into a single call, reducing latency:

import { createStreamingSTT, assetModelPath } from 'react-native-sherpa-onnx/stt';

const engine = await createStreamingSTT({
  modelPath: assetModelPath('models/streaming-zipformer-en'),
  modelType: 'transducer',
});

const stream = await engine.createStream();

// Process audio chunk (feeds, decodes while ready, returns result)
const { result, isEndpoint } = await stream.processAudioChunk(
  audioSamples,
  16000
);

console.log('Result:', result.text);

if (isEndpoint) {
  console.log('End of utterance detected');
  await stream.reset();
}

await stream.release();
await engine.destroy();

Live Microphone Recognition

import { createStreamingSTT, assetModelPath } from 'react-native-sherpa-onnx/stt';
import { createPcmLiveStream } from 'react-native-sherpa-onnx/audio';

const engine = await createStreamingSTT({
  modelPath: assetModelPath('models/streaming-zipformer-en'),
  modelType: 'auto',
  enableEndpoint: true,
});

const stream = await engine.createStream();

// Create live microphone stream
const mic = createPcmLiveStream({ sampleRate: 16000 });

// Handle audio data
const unsubscribeData = mic.onData(async (samples, sampleRate) => {
  const { result, isEndpoint } = await stream.processAudioChunk(
    samples,
    sampleRate
  );
  
  console.log('Live transcription:', result.text);
  
  if (isEndpoint) {
    console.log('Utterance complete:', result.text);
    await stream.reset();
  }
});

// Start recording
await mic.start();

// Later: stop recording
await mic.stop();
unsubscribeData();
await stream.release();
await engine.destroy();

Custom Endpoint Detection

import { createStreamingSTT, assetModelPath } from 'react-native-sherpa-onnx/stt';

const engine = await createStreamingSTT({
  modelPath: assetModelPath('models/streaming-zipformer-en'),
  modelType: 'transducer',
  enableEndpoint: true,
  endpointConfig: {
    rule1: {
      mustContainNonSilence: false,
      minTrailingSilence: 3.0, // 3 seconds of silence
      minUtteranceLength: 0,
    },
    rule2: {
      mustContainNonSilence: true,
      minTrailingSilence: 1.2, // 1.2 seconds of silence after speech
      minUtteranceLength: 0,
    },
    rule3: {
      mustContainNonSilence: false,
      minTrailingSilence: 0,
      minUtteranceLength: 30, // Max 30 seconds
    },
  },
});

const stream = await engine.createStream();

// ... use stream for recognition

await stream.release();
await engine.destroy();

With Hotwords for Contextual Biasing

import { createStreamingSTT, assetModelPath } from 'react-native-sherpa-onnx/stt';

const engine = await createStreamingSTT({
  modelPath: assetModelPath('models/streaming-zipformer-en'),
  modelType: 'transducer',
  hotwordsFile: '/path/to/hotwords.txt',
  hotwordsScore: 2.0,
});

// Create stream with additional runtime hotwords
const stream = await engine.createStream('COVID-19\nSHERPA-ONNX\nREACT-NATIVE');

const { result, isEndpoint } = await stream.processAudioChunk(
  audioSamples,
  16000
);

console.log(result.text); // Hotwords will have higher confidence

await stream.release();
await engine.destroy();

Multiple Concurrent Streams

import { createStreamingSTT, assetModelPath } from 'react-native-sherpa-onnx/stt';

const engine = await createStreamingSTT({
  modelPath: assetModelPath('models/streaming-zipformer-en'),
  modelType: 'auto',
});

// Create multiple streams from the same engine
const stream1 = await engine.createStream();
const stream2 = await engine.createStream();

// Use streams independently
const result1 = await stream1.processAudioChunk(audio1, 16000);
const result2 = await stream2.processAudioChunk(audio2, 16000);

console.log('Stream 1:', result1.result.text);
console.log('Stream 2:', result2.result.text);

// Clean up
await stream1.release();
await stream2.release();
await engine.destroy();

Helper Functions

mapDetectedToOnlineType()

Map detected STT model type (from detectSttModel) to a streaming model type.

function mapDetectedToOnlineType(
  detectedType: string | undefined
): OnlineSTTModelType

Throws if the detected type doesn’t support streaming.

Example

import { detectSttModel, mapDetectedToOnlineType, createStreamingSTT } from 'react-native-sherpa-onnx/stt';

const detection = await detectSttModel(modelPath);
const onlineType = mapDetectedToOnlineType(detection.modelType);

const engine = await createStreamingSTT({
  modelPath,
  modelType: onlineType,
});

getOnlineTypeOrNull()

Check if a detected model type supports streaming.

function getOnlineTypeOrNull(
  detectedType: string | undefined
): OnlineSTTModelType | null

Returns the online model type if supported, or null if streaming is not available.

Example

import { detectSttModel, getOnlineTypeOrNull } from 'react-native-sherpa-onnx/stt';

const detection = await detectSttModel(modelPath);
const onlineType = getOnlineTypeOrNull(detection.modelType);

if (onlineType) {
  console.log('Supports streaming:', onlineType);
  // Use createStreamingSTT
} else {
  console.log('Offline only, use createSTT');
}

Core API

Speech-to-Text

Text-to-Speech

Audio & Models

Parameters

Returns

SttStream Interface

Examples

Basic Real-Time Recognition

Simplified with processAudioChunk()

Live Microphone Recognition

Custom Endpoint Detection

With Hotwords for Contextual Biasing

Multiple Concurrent Streams

Helper Functions

mapDetectedToOnlineType()

Example

getOnlineTypeOrNull()

Example

See Also

Build docs developers (and LLMs) love

Core API

Speech-to-Text

Text-to-Speech

Audio & Models

Documentation Index

​Parameters

​Returns

​SttStream Interface

​Examples

​Basic Real-Time Recognition

​Simplified with processAudioChunk()

​Live Microphone Recognition

​Custom Endpoint Detection

​With Hotwords for Contextual Biasing

​Multiple Concurrent Streams

​Helper Functions

​mapDetectedToOnlineType()

​Example

​getOnlineTypeOrNull()

​Example

​See Also

Build docs developers (and LLMs) love

Parameters

Returns

SttStream Interface

Examples

Basic Real-Time Recognition

Simplified with processAudioChunk()

Live Microphone Recognition

Custom Endpoint Detection

With Hotwords for Contextual Biasing

Multiple Concurrent Streams

Helper Functions

mapDetectedToOnlineType()

Example

getOnlineTypeOrNull()

Example

See Also