Package videosdk.plugins.cartesia
Sub-modules
videosdk.plugins.cartesia.sttvideosdk.plugins.cartesia.tts
Classes
class CartesiaSTT (*,
api_key: str | None = None,
model: str = 'ink-whisper',
language: str = 'en',
sample_rate: int = 48000,
base_url: str = 'wss://api.cartesia.ai/stt/websocket')-
Expand source code
class CartesiaSTT(BaseSTT): def __init__( self, *, api_key: str | None = None, model: str = "ink-whisper", language: str = "en", sample_rate: int = 48000, base_url: str = "wss://api.cartesia.ai/stt/websocket", ) -> None: """Initialize the Cartesia STT plugin Args: api_key (str | None, optional): Cartesia API key. Uses CARTESIA_API_KEY environment variable if not provided. Defaults to None. model (str): The model to use for the STT plugin. Defaults to "ink-whisper". language (str): The language to use for the STT plugin, e.g. "en". Defaults to "en". sample_rate (int): The sample rate to use for the STT plugin. Defaults to 48000. base_url (str): The base URL to use for the STT plugin. Defaults to "wss://api.cartesia.ai/stt/websocket". """ super().__init__() self.api_key = api_key or os.getenv("CARTESIA_API_KEY") if not self.api_key: raise ValueError( "Cartesia API key must be provided either through api_key parameter or CARTESIA_API_KEY environment variable") self.model = model self.language = language self.sample_rate = sample_rate self.base_url = base_url self._session: Optional[aiohttp.ClientSession] = None self._ws: Optional[aiohttp.ClientWebSocketResponse] = None self._ws_task: Optional[asyncio.Task] = None self._last_interim_at = 0.0 self.input_sample_rate = sample_rate self.target_sample_rate = 16000 async def process_audio( self, audio_frames: bytes, language: Optional[str] = None, **kwargs: Any ) -> None: """Process audio frames and send to Cartesia's STT API""" if not self._ws: await self._connect_ws() self._ws_task = asyncio.create_task(self._listen_for_responses()) try: audio_data = np.frombuffer(audio_frames, dtype=np.int16) if audio_data.ndim == 1 and len(audio_data) % 2 == 0 and self.input_sample_rate == self.sample_rate and self.input_sample_rate != self.target_sample_rate: audio_data = audio_data.reshape(-1, 2).mean(axis=1).astype(np.int16) if self.input_sample_rate != self.target_sample_rate: audio_data = self._resample_audio(audio_data, self.input_sample_rate, self.target_sample_rate) audio_data = np.clip(audio_data, -32768, 32767) audio_data = audio_data.astype(np.int16) audio_bytes = audio_data.astype(np.int16).tobytes() await self._ws.send_bytes(audio_bytes) except Exception as e: self.emit("error", str(e)) if self._ws: await self._ws.close() self._ws = None if self._ws_task: self._ws_task.cancel() self._ws_task = None async def _listen_for_responses(self) -> None: """Background task to listen for WebSocket responses""" if not self._ws: return try: async for msg in self._ws: if msg.type == aiohttp.WSMsgType.TEXT: data = msg.json() responses = self._handle_ws_message(data) for response in responses: if self._transcript_callback: await self._transcript_callback(response) elif msg.type == aiohttp.WSMsgType.ERROR: error = f"WebSocket error: {self._ws.exception()}" self.emit("error", error) break elif msg.type == aiohttp.WSMsgType.CLOSED: logger.info("WebSocket connection closed") break except Exception as e: self.emit("error", f"Error listening for responses: {str(e)}") finally: if self._ws: await self._ws.close() self._ws = None async def _connect_ws(self) -> None: """Establish WebSocket connection with Cartesia's STT API""" if not self._session: self._session = aiohttp.ClientSession() query_params = { "model": self.model, "language": self.language, "encoding": "pcm_s16le", "sample_rate": str(self.target_sample_rate), "api_key": self.api_key, } headers = { "Cartesia-Version": "2024-11-13", "User-Agent": "VideoSDK-Cartesia-STT", } ws_url = f"{self.base_url}?{urlencode(query_params)}" try: self._ws = await self._session.ws_connect(ws_url, headers=headers) except Exception as e: logger.error(f"Error connecting to WebSocket: {str(e)}") if self._ws: await self._ws.close() self._ws = None raise def _handle_ws_message(self, msg: dict) -> list[STTResponse]: """Handle incoming WebSocket messages and generate STT responses""" responses = [] try: msg_type = msg.get("type") if msg_type == "transcript": transcript = msg.get("text", "") is_final = msg.get("is_final", False) language = msg.get("language", self.language) duration = msg.get("duration", 0.0) if transcript: current_time = time.time() if is_final: responses.append(STTResponse( event_type=SpeechEventType.FINAL, data=SpeechData( text=transcript, confidence=1.0, language=language, start_time=0.0, end_time=duration, ), metadata={ "model": self.model, "request_id": msg.get("request_id"), "duration": duration, } )) else: if current_time - self._last_interim_at > 0.1: responses.append(STTResponse( event_type=SpeechEventType.INTERIM, data=SpeechData( text=transcript, confidence=1.0, language=language, start_time=0.0, end_time=duration, ), metadata={ "model": self.model, "request_id": msg.get("request_id"), "duration": duration, } )) self._last_interim_at = current_time elif msg_type == "flush_done": logger.info("Cartesia STT: Flush completed") elif msg_type == "done": logger.info("Cartesia STT: Session ended") elif msg_type == "error": error_msg = msg.get("message", "Unknown error") error_code = msg.get("code", "unknown") self.emit("error", f"{error_code}: {error_msg}") except Exception as e: logger.error(f"Error handling WebSocket message: {str(e)}") return responses def _resample_audio(self, audio: np.ndarray, orig_sr: int, target_sr : int) -> np.ndarray : """ Use polyphase filtering for resampling, which is more accurate for integer-ratio conversions. Assumes input is np.int16. """ if orig_sr == target_sr: return audio gcd = math.gcd(orig_sr, target_sr) up = target_sr // gcd down = orig_sr // gcd return resample_poly(audio, up, down) async def aclose(self) -> None: """Cleanup resources""" if self._ws and not self._ws.closed: try: await self._ws.send_str("done") await asyncio.sleep(0.1) except Exception as e: logger.error(f"Error sending done command: {str(e)}") if self._ws_task: self._ws_task.cancel() try: await self._ws_task except asyncio.CancelledError: pass self._ws_task = None if self._ws: await self._ws.close() self._ws = None if self._session: await self._session.close() self._session = None await super().aclose()Base class for Speech-to-Text implementations
Initialize the Cartesia STT plugin
Args
api_key:str | None, optional- Cartesia API key. Uses CARTESIA_API_KEY environment variable if not provided. Defaults to None.
model:str- The model to use for the STT plugin. Defaults to "ink-whisper".
language:str- The language to use for the STT plugin, e.g. "en". Defaults to "en".
sample_rate:int- The sample rate to use for the STT plugin. Defaults to 48000.
base_url:str- The base URL to use for the STT plugin. Defaults to "wss://api.cartesia.ai/stt/websocket".
Ancestors
- videosdk.agents.stt.stt.STT
- videosdk.agents.event_emitter.EventEmitter
- typing.Generic
Methods
async def aclose(self) ‑> None-
Expand source code
async def aclose(self) -> None: """Cleanup resources""" if self._ws and not self._ws.closed: try: await self._ws.send_str("done") await asyncio.sleep(0.1) except Exception as e: logger.error(f"Error sending done command: {str(e)}") if self._ws_task: self._ws_task.cancel() try: await self._ws_task except asyncio.CancelledError: pass self._ws_task = None if self._ws: await self._ws.close() self._ws = None if self._session: await self._session.close() self._session = None await super().aclose()Cleanup resources
async def process_audio(self, audio_frames: bytes, language: Optional[str] = None, **kwargs: Any) ‑> None-
Expand source code
async def process_audio( self, audio_frames: bytes, language: Optional[str] = None, **kwargs: Any ) -> None: """Process audio frames and send to Cartesia's STT API""" if not self._ws: await self._connect_ws() self._ws_task = asyncio.create_task(self._listen_for_responses()) try: audio_data = np.frombuffer(audio_frames, dtype=np.int16) if audio_data.ndim == 1 and len(audio_data) % 2 == 0 and self.input_sample_rate == self.sample_rate and self.input_sample_rate != self.target_sample_rate: audio_data = audio_data.reshape(-1, 2).mean(axis=1).astype(np.int16) if self.input_sample_rate != self.target_sample_rate: audio_data = self._resample_audio(audio_data, self.input_sample_rate, self.target_sample_rate) audio_data = np.clip(audio_data, -32768, 32767) audio_data = audio_data.astype(np.int16) audio_bytes = audio_data.astype(np.int16).tobytes() await self._ws.send_bytes(audio_bytes) except Exception as e: self.emit("error", str(e)) if self._ws: await self._ws.close() self._ws = None if self._ws_task: self._ws_task.cancel() self._ws_task = NoneProcess audio frames and send to Cartesia's STT API
class CartesiaTTS (*,
api_key: str | None = None,
model: str = 'sonic-2',
voice_id: Union[str, List[float]] = 'f786b574-daa5-4673-aa0c-cbe3e8534c02',
language: str = 'en',
base_url: str = 'https://api.cartesia.ai')-
Expand source code
class CartesiaTTS(TTS): def __init__( self, *, api_key: str | None = None, model: str = DEFAULT_MODEL, voice_id: Union[str, List[float]] = DEFAULT_VOICE_ID, language: str = "en", base_url: str = "https://api.cartesia.ai", ) -> None: """Initialize the Cartesia TTS plugin Args: api_key (str | None, optional): Cartesia API key. Uses CARTESIA_API_KEY environment variable if not provided. Defaults to None. model (str): The model to use for the TTS plugin. Defaults to "sonic-2". voice_id (Union[str, List[float]]): The voice ID to use for the TTS plugin. Defaults to "794f9389-aac1-45b6-b726-9d9369183238". api_key (str | None, optional): Cartesia API key. Uses CARTESIA_API_KEY environment variable if not provided. Defaults to None. language (str): The language to use for the TTS plugin. Defaults to "en". base_url (str): The base URL to use for the TTS plugin. Defaults to "https://api.cartesia.ai". """ super().__init__(sample_rate=CARTESIA_SAMPLE_RATE, num_channels=CARTESIA_CHANNELS) self.model = model self.language = language self.base_url = base_url self._voice = voice_id self._first_chunk_sent = False self._interrupted = False api_key = api_key or os.getenv("CARTESIA_API_KEY") if not api_key: raise ValueError( "Cartesia API key must be provided either through api_key parameter or CARTESIA_API_KEY environment variable") self._api_key = api_key self._ws_session: aiohttp.ClientSession | None = None self._ws_connection: aiohttp.ClientWebSocketResponse | None = None self._connection_lock = asyncio.Lock() self._receive_task: asyncio.Task | None = None self._context_futures: dict[str, asyncio.Future[None]] = {} def reset_first_audio_tracking(self) -> None: """Reset the first audio tracking state for the next TTS task""" self._first_chunk_sent = False async def synthesize( self, text: AsyncIterator[str] | str, voice_id: Optional[Union[str, List[float]]] = None, **kwargs: Any, ) -> None: """Synthesize text to speech using Cartesia's streaming WebSocket API.""" context_id = "" try: if not self.audio_track or not self.loop: self.emit("error", "Audio track or event loop not set") return if voice_id: self._voice = voice_id self._interrupted = False await self._ensure_ws_connection() if not self._ws_connection: raise RuntimeError("WebSocket connection is not available.") context_id = os.urandom(8).hex() done_future: asyncio.Future[None] = asyncio.get_event_loop().create_future() self._context_futures[context_id] = done_future async def _string_iterator(s: str) -> AsyncIterator[str]: yield s text_iterator = _string_iterator(text) if isinstance(text, str) else text send_task = asyncio.create_task(self._send_task(text_iterator, context_id)) await done_future await send_task except Exception as e: self.emit("error", f"TTS synthesis failed: {str(e)}") finally: if context_id and context_id in self._context_futures: del self._context_futures[context_id] async def _send_task(self, text_iterator: AsyncIterator[str], context_id: str): """The dedicated task for sending text chunks over the WebSocket.""" has_sent_transcript = False try: voice_payload: dict[str, Any] = {} if isinstance(self._voice, str): voice_payload["mode"] = "id" voice_payload["id"] = self._voice else: voice_payload["mode"] = "embedding" voice_payload["embedding"] = self._voice base_payload = { "model_id": self.model, "language": self.language, "voice": voice_payload, "output_format": {"container": "raw", "encoding": "pcm_s16le", "sample_rate": self.sample_rate}, "add_timestamps": True, "context_id": context_id, } async for text_chunk in text_iterator: if self._interrupted: break if text_chunk and text_chunk.strip(): if not has_sent_transcript: pass payload = {**base_payload, "transcript": text_chunk + " ", "continue": True} if self._ws_connection and not self._ws_connection.closed: await self._ws_connection.send_str(json.dumps(payload)) has_sent_transcript = True except Exception as e: future = self._context_futures.get(context_id) if future and not future.done(): future.set_exception(e) finally: if has_sent_transcript and not self._interrupted: final_payload = {**base_payload, "transcript": " ", "continue": False} if self._ws_connection and not self._ws_connection.closed: await self._ws_connection.send_str(json.dumps(final_payload)) async def _receive_loop(self): """A single, long-running task that handles all incoming messages from the WebSocket.""" try: while self._ws_connection and not self._ws_connection.closed: msg = await self._ws_connection.receive() if msg.type in (aiohttp.WSMsgType.CLOSED, aiohttp.WSMsgType.ERROR): break if msg.type != aiohttp.WSMsgType.TEXT: continue data = json.loads(msg.data) context_id = data.get("context_id") future = self._context_futures.get(context_id) if not future or future.done(): continue if data.get("type") == "error": future.set_exception(RuntimeError(f"Cartesia API error: {json.dumps(data)}")) elif "data" in data and data["data"]: await self._stream_audio(base64.b64decode(data["data"])) elif data.get("done"): future.set_result(None) except Exception as e: for future in self._context_futures.values(): if not future.done(): future.set_exception(e) finally: self._context_futures.clear() async def _ensure_ws_connection(self) -> None: """Establishes or re-establishes the WebSocket connection if needed.""" async with self._connection_lock: if self._ws_connection and not self._ws_connection.closed: return if self._receive_task and not self._receive_task.done(): self._receive_task.cancel() if self._ws_connection: await self._ws_connection.close() if self._ws_session: await self._ws_session.close() try: self._ws_session = aiohttp.ClientSession() ws_url = self.base_url.replace('http', 'ws', 1) full_ws_url = f"{ws_url}/tts/websocket?api_key={self._api_key}&cartesia_version={API_VERSION}" self._ws_connection = await asyncio.wait_for( self._ws_session.ws_connect(full_ws_url, heartbeat=30.0), timeout=5.0 ) self._receive_task = asyncio.create_task(self._receive_loop()) except Exception as e: self.emit("error", f"Failed to establish WebSocket connection: {e}") raise async def _stream_audio(self, audio_chunk: bytes): """Streams a chunk of audio to the audio track.""" if self._interrupted or not audio_chunk: return if not self._first_chunk_sent and self._first_audio_callback: self._first_chunk_sent = True await self._first_audio_callback() if self.audio_track: await self.audio_track.add_new_bytes(audio_chunk) async def interrupt(self) -> None: """Interrupts any ongoing TTS, stopping audio playback and network activity.""" self._interrupted = True if self.audio_track: self.audio_track.interrupt() if self._ws_connection and not self._ws_connection.closed: await self._ws_connection.close() async def aclose(self) -> None: """Gracefully cleans up all resources.""" await super().aclose() self._interrupted = True if self._receive_task and not self._receive_task.done(): self._receive_task.cancel() if self._ws_connection and not self._ws_connection.closed: await self._ws_connection.close() if self._ws_session and not self._ws_session.closed: await self._ws_session.close()Base class for Text-to-Speech implementations
Initialize the Cartesia TTS plugin
Args
api_key:str | None, optional- Cartesia API key. Uses CARTESIA_API_KEY environment variable if not provided. Defaults to None.
model:str- The model to use for the TTS plugin. Defaults to "sonic-2".
voice_id:Union[str, List[float]]- The voice ID to use for the TTS plugin. Defaults to "794f9389-aac1-45b6-b726-9d9369183238".
api_key:str | None, optional- Cartesia API key. Uses CARTESIA_API_KEY environment variable if not provided. Defaults to None.
language:str- The language to use for the TTS plugin. Defaults to "en".
base_url:str- The base URL to use for the TTS plugin. Defaults to "https://api.cartesia.ai".
Ancestors
- videosdk.agents.tts.tts.TTS
- videosdk.agents.event_emitter.EventEmitter
- typing.Generic
Methods
async def aclose(self) ‑> None-
Expand source code
async def aclose(self) -> None: """Gracefully cleans up all resources.""" await super().aclose() self._interrupted = True if self._receive_task and not self._receive_task.done(): self._receive_task.cancel() if self._ws_connection and not self._ws_connection.closed: await self._ws_connection.close() if self._ws_session and not self._ws_session.closed: await self._ws_session.close()Gracefully cleans up all resources.
async def interrupt(self) ‑> None-
Expand source code
async def interrupt(self) -> None: """Interrupts any ongoing TTS, stopping audio playback and network activity.""" self._interrupted = True if self.audio_track: self.audio_track.interrupt() if self._ws_connection and not self._ws_connection.closed: await self._ws_connection.close()Interrupts any ongoing TTS, stopping audio playback and network activity.
def reset_first_audio_tracking(self) ‑> None-
Expand source code
def reset_first_audio_tracking(self) -> None: """Reset the first audio tracking state for the next TTS task""" self._first_chunk_sent = FalseReset the first audio tracking state for the next TTS task
async def synthesize(self,
text: AsyncIterator[str] | str,
voice_id: Optional[Union[str, List[float]]] = None,
**kwargs: Any) ‑> None-
Expand source code
async def synthesize( self, text: AsyncIterator[str] | str, voice_id: Optional[Union[str, List[float]]] = None, **kwargs: Any, ) -> None: """Synthesize text to speech using Cartesia's streaming WebSocket API.""" context_id = "" try: if not self.audio_track or not self.loop: self.emit("error", "Audio track or event loop not set") return if voice_id: self._voice = voice_id self._interrupted = False await self._ensure_ws_connection() if not self._ws_connection: raise RuntimeError("WebSocket connection is not available.") context_id = os.urandom(8).hex() done_future: asyncio.Future[None] = asyncio.get_event_loop().create_future() self._context_futures[context_id] = done_future async def _string_iterator(s: str) -> AsyncIterator[str]: yield s text_iterator = _string_iterator(text) if isinstance(text, str) else text send_task = asyncio.create_task(self._send_task(text_iterator, context_id)) await done_future await send_task except Exception as e: self.emit("error", f"TTS synthesis failed: {str(e)}") finally: if context_id and context_id in self._context_futures: del self._context_futures[context_id]Synthesize text to speech using Cartesia's streaming WebSocket API.