stt

xAI speech-to-text service implementation.

This module provides integration with xAI’s real-time speech-to-text WebSocket API documented at https://docs.x.ai/developers/rest-api-reference/inference/voice.

pipecat.services.xai.stt.language_to_xai_stt_language(language: Language) → str | None[source]

Convert a Language enum to the xAI STT language code.

xAI STT accepts two-letter language codes (e.g. en, fr, de, ja). When set, the server applies Inverse Text Normalization.

Parameters:: language – The Language enum value to convert.
Returns:: The corresponding xAI STT language code, or None if not supported.

Bases: STTSettings

Settings for XAISTTService.

Parameters:

interim_results – When True, partial transcripts are emitted approximately every 500ms.
endpointing – Silence duration in milliseconds that triggers a speech-final event. Range 0-5000. Server default is 10ms.
multichannel – When True, transcribes each interleaved channel independently. Requires channels >= 2.
channels – Number of interleaved channels (2-8). Required when multichannel is True.
diarize – When True, the server attaches a speaker field to each word identifying the detected speaker.

interim_results: bool | _NotGiven

endpointing: int | None | _NotGiven

multichannel: bool | None | _NotGiven

channels: int | None | _NotGiven

diarize: bool | None | _NotGiven

class pipecat.services.xai.stt.XAISTTService(*, api_key: str, ws_url: str = 'wss://api.x.ai/v1/stt', sample_rate: int = 16000, encoding: str = 'pcm', settings: XAISTTSettings | None = None, ttfs_p99_latency: float | None = 1.0, **kwargs)[source]

Bases: WebsocketSTTService

xAI real-time speech-to-text service.

Streams audio to xAI’s WebSocket STT endpoint and emits interim and final transcription frames. The XAI_API_KEY is passed directly as a Bearer token on the WebSocket handshake.

The connection is persistent: audio is streamed continuously and the server emits transcript.partial events with is_final and speech_final flags to mark utterance boundaries. If the connection drops mid-session, the base class reconnects automatically.

Settings: alias of XAISTTSettings

__init__(*, api_key: str, ws_url: str = 'wss://api.x.ai/v1/stt', sample_rate: int = 16000, encoding: str = 'pcm', settings: XAISTTSettings | None = None, ttfs_p99_latency: float | None = 1.0, **kwargs)[source]

Initialize the xAI STT service.

Parameters:

api_key – xAI API key (used as Bearer for the WebSocket handshake).
ws_url – WebSocket endpoint URL. Defaults to wss://api.x.ai/v1/stt.
sample_rate – Audio sample rate in Hz. Supported values: 8000, 16000, 22050, 24000, 44100, 48000. Defaults to 16000.
encoding – Audio encoding. One of "pcm" (signed 16-bit LE), "mulaw", or "alaw". Defaults to "pcm".
settings – Runtime-updatable settings overriding defaults.
ttfs_p99_latency – P99 latency from speech end to final transcript in seconds. See https://github.com/pipecat-ai/stt-benchmark.
**kwargs – Additional arguments passed to WebsocketSTTService.

can_generate_metrics() → bool[source]

Check if the service can generate metrics.

Returns:: True if metrics generation is supported.

language_to_service_language(language: Language) → str | None[source]: Convert a Language enum to the xAI STT language code.

async start(frame: StartFrame)[source]: Start the speech-to-text service.

async stop(frame: EndFrame)[source]: Stop the speech-to-text service.

async cancel(frame: CancelFrame)[source]: Cancel the speech-to-text service.

async run_stt(audio: bytes) → AsyncGenerator[Frame | None, None][source]

Forward raw audio bytes to the xAI STT WebSocket.

Transcription frames are pushed from the receive task, not yielded from this coroutine.