RevAISTT
Rev AI streaming STT provider using a raw WebSocket connection.
Defined in: src/providers/stt/revai/RevAISTT.ts:308
Rev AI streaming STT provider using a raw WebSocket connection.
Remarks
RevAISTT extends LiveSTTProvider and connects to Rev AI’s streaming speech-to-text WebSocket API. All session options (audio content_type, language, profanity filtering, …) are passed as query parameters on the connection URL. After the upgrade, Rev AI sends a connected message — the provider waits for it before resolving connect(), as required by the API — and audio is then streamed as binary frames.
Rev AI responds with hypothesis messages: partial hypotheses contain the words recognized so far (emitted as interim results), and final hypotheses contain the punctuated, confidence-scored transcript of a completed segment. Each final hypothesis is emitted with isFinal: true and utteranceComplete: true, triggering the next pipeline stage.
Key features:
- Interim (partial) and final transcription results
- Nine languages (en, fr, de, it, ja, ko, cmn, pt, es)
- Profanity filtering, disfluency removal, and custom vocabularies
- Per-word timestamps and confidence scores on final hypotheses
- Automatic WebSocket reconnection with exponential backoff (via WebSocketManager)
- Proxy mode via RevAISTTConfig.proxyUrl (recommended for production so the access token stays server-side)
Transport: WebSocket (via WebSocketManager)
Browser support: All modern browsers (Chrome, Firefox, Safari, Edge). No peer dependencies required — the provider uses a raw WebSocket connection managed by the SDK’s built-in WebSocketManager.
Auth in browser: Rev AI authenticates the WebSocket via an access_token query parameter (headers are not supported on the upgrade). Direct mode places the resolved apiKey on the URL; proxy mode omits it and the proxy appends the token server-side.
Data flow:
Microphone -> AudioCapture -> sendAudio(chunk) -> binary frames -> Rev AI WS
|
CompositeVoice <- onTranscription(result) <---- hypothesis messages <--+
Example
import { RevAISTT } from 'composite-voice';
const stt = new RevAISTT({
proxyUrl: 'http://localhost:3001/api/proxy/revai',
language: 'en',
});
await stt.initialize();
stt.onTranscription((result) => {
if (result.isFinal) {
console.log('Final:', result.text);
} else {
console.log('Interim:', result.text);
}
});
await stt.connect();
// ... send audio chunks via stt.sendAudio(chunk) ...
await stt.disconnect(); // sends "EOS" and awaits the last hypothesis
See
- LiveSTTProvider for the base WebSocket STT class
- RevAISTTConfig for configuration options
- SonioxSTT for an alternative real-time STT provider
Extends
LiveSTTProvider
Constructors
Constructor
new RevAISTT(config, logger?): RevAISTT;
Defined in: src/providers/stt/revai/RevAISTT.ts:344
Create a new RevAISTT provider.
Parameters
| Parameter | Type | Description |
|---|---|---|
config | RevAISTTConfig | Rev AI STT configuration. Must include either apiKey or proxyUrl. |
logger? | Logger | Optional parent logger; a child will be derived. |
Returns
RevAISTT
Example
const stt = new RevAISTT({
apiKey: '02.abc123...',
sampleRate: 16000,
});
Overrides
LiveSTTProvider.constructor
Properties
| Property | Modifier | Type | Default value | Description | Overrides | Inherited from | Defined in |
|---|---|---|---|---|---|---|---|
config | public | RevAISTTConfig | undefined | STT-specific provider configuration. | LiveSTTProvider.config | - | src/providers/stt/revai/RevAISTT.ts:309 |
initialized | protected | boolean | false | Tracks whether initialize has completed successfully. | - | LiveSTTProvider.initialized | src/providers/base/BaseProvider.ts:97 |
logger | protected | Logger | undefined | Scoped logger instance for this provider. | - | LiveSTTProvider.logger | src/providers/base/BaseProvider.ts:94 |
roles | readonly | readonly ProviderRole[] | undefined | STT providers cover the 'stt' pipeline role by default. | - | LiveSTTProvider.roles | src/providers/base/BaseSTTProvider.ts:77 |
transcriptionCallback? | protected | (result) => void | undefined | Callback registered by the SDK or consumer to receive transcription results. Set via onTranscription. | - | LiveSTTProvider.transcriptionCallback | src/providers/base/BaseSTTProvider.ts:86 |
type | readonly | ProviderType | undefined | Communication transport this provider uses ('rest' or 'websocket'). | - | LiveSTTProvider.type | src/providers/base/BaseProvider.ts:74 |
Accessors
isProxyMode
Get Signature
get protected isProxyMode(): boolean;
Defined in: src/providers/base/BaseProvider.ts:286
Whether the provider is in proxy mode.
Returns
boolean
true when proxyUrl is set.
Inherited from
LiveSTTProvider.isProxyMode
Methods
assertAuth()
protected assertAuth(): void;
Defined in: src/providers/base/BaseProvider.ts:272
Validate that auth is configured (either apiKey or proxyUrl).
Returns
void
Remarks
Call this in onInitialize() for any provider that requires external authentication. Native providers (NativeSTT, NativeTTS) and in-browser providers (WebLLM) should NOT call this method.
Throws
ProviderInitializationError Thrown when neither apiKey nor proxyUrl is set.
Inherited from
LiveSTTProvider.assertAuth
assertReady()
protected assertReady(): void;
Defined in: src/providers/base/BaseProvider.ts:255
Guard that throws if the provider has not been initialized.
Returns
void
Remarks
Call at the start of any method that requires the provider to be ready.
Throws
Error Thrown with a descriptive message when initialized is false.
Inherited from
LiveSTTProvider.assertReady
connect()
connect(): Promise<void>;
Defined in: src/providers/stt/revai/RevAISTT.ts:473
Open a WebSocket connection to Rev AI for streaming transcription.
Returns
Promise<void>
Remarks
Creates a WebSocketManager with reconnection disabled — a dead socket must surface immediately via onConnectionLost so the SDK (or a FallbackSTT chain) can recover. The SDK drives reconnection through connect. Waits for the socket to open, then waits for Rev AI’s connected message — the API requires it before any audio is sent. The connection timeout defaults to config.timeout (10 000 ms) and applies to both waits independently.
Throws
ProviderConnectionError Thrown when the provider is not initialized, the connection fails, or the connected message does not arrive in time (e.g. close code 4001 for an invalid access token).
Overrides
LiveSTTProvider.connect
disconnect()
disconnect(): Promise<void>;
Defined in: src/providers/stt/revai/RevAISTT.ts:772
Gracefully close the Rev AI WebSocket connection.
Returns
Promise<void>
Remarks
Sends the "EOS" text frame (exact capitalization required by Rev AI) so the service returns the last final hypothesis and closes the socket with code 1000, waits for that close (up to 3 s), then disconnects the underlying WebSocketManager. Closing the socket without sending EOS would trigger a 1007 (Invalid Payload) error upstream.
Throws
Re-throws any unexpected error during disconnection.
Overrides
LiveSTTProvider.disconnect
dispose()
dispose(): Promise<void>;
Defined in: src/providers/base/BaseProvider.ts:154
Clean up resources and dispose of the provider.
Returns
Promise<void>
Remarks
Delegates to the subclass hook onDispose and resets the initialized flag. If the provider is not initialized, the call is a no-op.
Throws
Re-throws any error raised by onDispose.
Inherited from
LiveSTTProvider.dispose
emitConnectionLost()
protected emitConnectionLost(message): void;
Defined in: src/providers/base/LiveSTTProvider.ts:175
Emit the standardized “connection lost” error result.
Parameters
| Parameter | Type | Description |
|---|---|---|
message | string | Human-readable description of why the connection died. |
Returns
void
Remarks
Every live STT provider MUST call this when its streaming connection dies unexpectedly mid-session (server-initiated close, reconnection give-up, fatal transport error) and will not recover on its own. This error-shaped result is the liveness signal the pipeline’s error handling (transcription.error) and FallbackSTT’s mid-session failover depend on — a provider that only logs and flips an internal flag leaves the session silently dead.
Do NOT call this for closes the application requested via disconnect().
Inherited from
LiveSTTProvider.emitConnectionLost
emitTranscription()
protected emitTranscription(result): void;
Defined in: src/providers/base/BaseSTTProvider.ts:206
Emit a transcription result to the registered callback.
Parameters
| Parameter | Type | Description |
|---|---|---|
result | TranscriptionResult | The transcription result to emit. |
Returns
void
Remarks
Subclasses call this method whenever transcribed text is available. If no callback has been registered via onTranscription, the result is logged as a warning and dropped.
Inherited from
LiveSTTProvider.emitTranscription
getConfig()
getConfig(): STTProviderConfig;
Defined in: src/providers/base/BaseSTTProvider.ts:225
Get a shallow copy of the current STT configuration.
Returns
A new STTProviderConfig object.
Inherited from
LiveSTTProvider.getConfig
initialize()
initialize(): Promise<void>;
Defined in: src/providers/base/BaseProvider.ts:127
Initialize the provider, making it ready for use.
Returns
Promise<void>
Remarks
Calls the subclass hook onInitialize. If the provider has already been initialized the call is a no-op.
Throws
ProviderInitializationError Thrown when onInitialize rejects. The original error is wrapped with the provider class name for diagnostics.
Inherited from
LiveSTTProvider.initialize
isFinal()
isFinal(result): boolean;
Defined in: src/providers/base/BaseSTTProvider.ts:174
Is this a final segment (but not necessarily utterance-complete)?
Parameters
| Parameter | Type | Description |
|---|---|---|
result | TranscriptionResult | The transcription result to check. |
Returns
boolean
true when this is a final segment.
Remarks
A final segment represents committed text, but multi-segment providers (e.g., Deepgram) may emit several final segments for a single utterance. Only the last one will have isUtteranceComplete return true.
Inherited from
LiveSTTProvider.isFinal
isInterim()
isInterim(result): boolean;
Defined in: src/providers/base/BaseSTTProvider.ts:159
Is this an interim (partial, non-final) result?
Parameters
| Parameter | Type | Description |
|---|---|---|
result | TranscriptionResult | The transcription result to check. |
Returns
boolean
true when this is an interim result.
Remarks
Interim results update as the user speaks and are replaced by subsequent results. Useful for display but not for triggering downstream processing.
Inherited from
LiveSTTProvider.isInterim
isPreflight()
isPreflight(result): boolean;
Defined in: src/providers/base/BaseSTTProvider.ts:144
Is this a preflight/eager end-of-turn signal?
Parameters
| Parameter | Type | Description |
|---|---|---|
result | TranscriptionResult | The transcription result to check. |
Returns
boolean
true when this is a preflight signal.
Remarks
Used by the eager LLM pipeline for speculative generation. Only providers with preflight support (e.g., Deepgram Flux) need to override this.
Inherited from
LiveSTTProvider.isPreflight
isReady()
isReady(): boolean;
Defined in: src/providers/base/BaseProvider.ts:178
Check whether the provider has been initialized and is ready.
Returns
boolean
true when initialize has completed successfully and dispose has not yet been called.
Inherited from
LiveSTTProvider.isReady
isUtteranceComplete()
isUtteranceComplete(result): boolean;
Defined in: src/providers/base/BaseSTTProvider.ts:129
Is this result a complete utterance ready for LLM processing?
Parameters
| Parameter | Type | Description |
|---|---|---|
result | TranscriptionResult | The transcription result to check. |
Returns
boolean
true when the utterance is complete.
Remarks
The orchestrator calls this to decide when to send transcribed text to the LLM. Concrete providers override this when they have domain- specific endpointing logic (e.g., DeepgramSTT checks speech_final).
Inherited from
LiveSTTProvider.isUtteranceComplete
isWebSocketConnected()
isWebSocketConnected(): boolean;
Defined in: src/providers/stt/revai/RevAISTT.ts:837
Check whether the Rev AI WebSocket connection is currently open.
Returns
boolean
true when connected and ready to receive audio.
onConfigUpdate()
protected onConfigUpdate(_config): void;
Defined in: src/providers/base/BaseProvider.ts:242
Hook called after updateConfig merges new values.
Parameters
| Parameter | Type | Description |
|---|---|---|
_config | Partial<BaseProviderConfig> | The partial configuration that was merged. |
Returns
void
Remarks
The default implementation is a no-op. Override in subclasses to react to runtime configuration changes (e.g. reconnect with a new API key).
Inherited from
LiveSTTProvider.onConfigUpdate
onDispose()
protected onDispose(): Promise<void>;
Defined in: src/providers/stt/revai/RevAISTT.ts:378
Disconnect the WebSocket (if connected) and release the manager.
Returns
Promise<void>
Overrides
LiveSTTProvider.onDispose
onInitialize()
protected onInitialize(): Promise<void>;
Defined in: src/providers/stt/revai/RevAISTT.ts:362
Validate that either apiKey or proxyUrl is configured.
Returns
Promise<void>
Throws
ProviderInitializationError Thrown when neither apiKey nor proxyUrl is set.
Overrides
LiveSTTProvider.onInitialize
onTranscription()
onTranscription(callback): void;
Defined in: src/providers/base/BaseSTTProvider.ts:191
Register a callback to receive transcription results.
Parameters
| Parameter | Type | Description |
|---|---|---|
callback | (result) => void | Function invoked with each TranscriptionResult. |
Returns
void
Remarks
All STT providers — regardless of transport — deliver text through this callback. CompositeVoice registers it during pipeline setup so that transcription results flow into the conversation manager and, ultimately, the LLM provider.
Inherited from
LiveSTTProvider.onTranscription
processAudio()
processAudio(chunk): void;
Defined in: src/providers/base/LiveSTTProvider.ts:140
Process a raw audio chunk by sending it over the WebSocket.
Parameters
| Parameter | Type | Description |
|---|---|---|
chunk | ArrayBuffer | Raw audio data as an ArrayBuffer. |
Returns
void
Remarks
Legacy alias for sendAudio. Delegates to sendAudioToSocket.
Inherited from
LiveSTTProvider.processAudio
resolveApiKey()
protected resolveApiKey(): Promise<string>;
Defined in: src/providers/base/BaseProvider.ts:321
Resolve the API key, calling the factory if apiKey is a function.
Returns
Promise<string>
The resolved API key string, or 'proxy' in proxy mode.
Inherited from
LiveSTTProvider.resolveApiKey
resolveAuthHeader()
protected resolveAuthHeader(defaultAuthType?): Promise<string | undefined>;
Defined in: src/providers/base/BaseProvider.ts:366
Resolve Authorization header value for the configured auth type.
Parameters
| Parameter | Type | Default value | Description |
|---|---|---|---|
defaultAuthType | "token" | "bearer" | 'token' | The default auth type for this provider. |
Returns
Promise<string | undefined>
The Authorization header value, or undefined in proxy mode.
Remarks
If apiKey is a factory function it is called to get a fresh token. Returns the header value for REST or server-side WebSocket connections:
'token'→'Token <apiKey>''bearer'→'Bearer <apiKey>'
Returns undefined in proxy mode.
Inherited from
LiveSTTProvider.resolveAuthHeader
resolveBaseUrl()
protected resolveBaseUrl(defaultUrl?): string | undefined;
Defined in: src/providers/base/BaseProvider.ts:307
Resolve the base URL for this provider.
Parameters
| Parameter | Type | Description |
|---|---|---|
defaultUrl? | string | The provider’s default API URL. Pass undefined to let the underlying SDK use its own default. |
Returns
string | undefined
The resolved URL, or undefined when all sources are unset.
Remarks
Priority: proxyUrl > endpoint > defaultUrl.
For WebSocket providers (this.type === 'websocket'), the proxy URL’s http(s) scheme is automatically converted to ws(s).
When no URL is configured and defaultUrl is undefined, the return value is undefined — this lets SDK-based providers (Anthropic, OpenAI) fall back to their own built-in defaults.
Inherited from
LiveSTTProvider.resolveBaseUrl
resolveWsProtocols()
protected resolveWsProtocols(defaultAuthType?): Promise<string[] | undefined>;
Defined in: src/providers/base/BaseProvider.ts:342
Resolve WebSocket subprotocol for authentication.
Parameters
| Parameter | Type | Default value | Description |
|---|---|---|---|
defaultAuthType | "token" | "bearer" | 'token' | The default auth type for this provider. |
Returns
Promise<string[] | undefined>
Subprotocol array for new WebSocket(url, protocols), or undefined.
Remarks
If apiKey is a factory function it is called to get a fresh token. Returns the subprotocol array for direct mode based on authType:
'token'→['token', apiKey](Deepgram default)'bearer'→['bearer', apiKey](OAuth/Bearer tokens)
Returns undefined in proxy mode (no client-side auth needed).
Inherited from
LiveSTTProvider.resolveWsProtocols
sendAudio()
sendAudio(chunk): void;
Defined in: src/providers/base/LiveSTTProvider.ts:128
Send an audio chunk for real-time transcription.
Parameters
| Parameter | Type | Description |
|---|---|---|
chunk | ArrayBuffer | Raw audio data as an ArrayBuffer. |
Returns
void
Remarks
This is the public method required by the ILiveSTTProvider interface. It delegates to sendAudioToSocket, which subclasses implement to forward audio data over the WebSocket connection. For providers that manage their own audio (e.g. NativeSTT), sendAudioToSocket is a no-op.
Inherited from
LiveSTTProvider.sendAudio
sendAudioToSocket()
protected sendAudioToSocket(chunk): void;
Defined in: src/providers/stt/revai/RevAISTT.ts:746
Send a raw audio chunk to Rev AI for streaming transcription.
Parameters
| Parameter | Type | Description |
|---|---|---|
chunk | ArrayBuffer | Raw audio data captured from the microphone. |
Returns
void
Remarks
The ArrayBuffer is forwarded as a binary WebSocket frame, which is the format required by Rev AI’s streaming API. The audio must match the declared content_type. If the connection is not open, the chunk is silently dropped and a warning is logged.
Overrides
LiveSTTProvider.sendAudioToSocket
updateConfig()
updateConfig(config): void;
Defined in: src/providers/base/BaseProvider.ts:201
Merge partial configuration updates into the current config.
Parameters
| Parameter | Type | Description |
|---|---|---|
config | Partial<BaseProviderConfig> | A partial configuration object whose keys will overwrite existing values. |
Returns
void
Remarks
After merging, the subclass hook onConfigUpdate is called so providers can react to changed values at runtime.
Inherited from
LiveSTTProvider.updateConfig