cartesia 0.0.5rc1__py2.py3-none-any.whl → 0.0.6__py2.py3-none-any.whl

cartesia/__init__.py

@@ -1,3 +1,3 @@
-from cartesia.tts import CartesiaTTS
+from cartesia.tts import AsyncCartesiaTTS, CartesiaTTS
 
-__all__ = ["CartesiaTTS"]
+__all__ = ["CartesiaTTS", "AsyncCartesiaTTS"]

cartesia/tts.py

@@ -3,19 +3,27 @@ import base64
 import json
 import os
 import uuid
+from types import TracebackType
 from typing import Any, AsyncGenerator, Dict, Generator, List, Optional, Tuple, TypedDict, Union
 
 import aiohttp
 import httpx
+import logging
 import requests
 from websockets.sync.client import connect
 
-DEFAULT_MODEL_ID = "genial-planet-1346"
+from cartesia.utils import retry_on_connection_error, retry_on_connection_error_async
+
+DEFAULT_MODEL_ID = ""
 DEFAULT_BASE_URL = "api.cartesia.ai"
 DEFAULT_API_VERSION = "v0"
-DEFAULT_TIMEOUT = 60  # seconds
+DEFAULT_TIMEOUT = 30  # seconds
 DEFAULT_NUM_CONNECTIONS = 10  # connections per client
 
+BACKOFF_FACTOR = 1
+MAX_RETRIES = 3
+
+logger = logging.getLogger(__name__)
 
 class AudioOutput(TypedDict):
     audio: bytes
@@ -74,7 +82,6 @@ class CartesiaTTS:
     To enable interrupt handling along the websocket, set `experimental_ws_handle_interrupts=True`.
 
     Examples:
-
         >>> client = CartesiaTTS()
 
         # Load available voices and their metadata (excluding the embeddings).
@@ -96,14 +103,13 @@ class CartesiaTTS:
     """
 
     def __init__(self, *, api_key: str = None, experimental_ws_handle_interrupts: bool = False):
-        """
-        Args:
-            api_key: The API key to use for authorization.
-                If not specified, the API key will be read from the environment variable
-                `CARTESIA_API_KEY`.
-            experimental_ws_handle_interrupts: Whether to handle interrupts when generating
-                audio using the websocket. This is an experimental feature and may have bugs
-                or be deprecated in the future.
+        """Args:
+        api_key: The API key to use for authorization.
+            If not specified, the API key will be read from the environment variable
+            `CARTESIA_API_KEY`.
+        experimental_ws_handle_interrupts: Whether to handle interrupts when generating
+            audio using the websocket. This is an experimental feature and may have bugs
+            or be deprecated in the future.
         """
         self.base_url = os.environ.get("CARTESIA_BASE_URL", DEFAULT_BASE_URL)
         self.api_key = api_key or os.environ.get("CARTESIA_API_KEY")
@@ -111,7 +117,6 @@ class CartesiaTTS:
         self.headers = {"X-API-Key": self.api_key, "Content-Type": "application/json"}
         self.websocket = None
         self.experimental_ws_handle_interrupts = experimental_ws_handle_interrupts
-        self.refresh_websocket()
 
     def get_voices(self, skip_embeddings: bool = True) -> Dict[str, VoiceMetadata]:
         """Returns a mapping from voice name -> voice metadata.
@@ -144,18 +149,23 @@ class CartesiaTTS:
             >>> audio = client.generate(transcript="Hello world!", voice=embedding)
         """
         params = {"select": "id, name, description"} if skip_embeddings else None
-        response = httpx.get(f"{self._http_url()}/voices", headers=self.headers, params=params)
+        response = httpx.get(
+            f"{self._http_url()}/voices",
+            headers=self.headers,
+            params=params,
+            timeout=DEFAULT_TIMEOUT,
+        )
 
         if not response.is_success:
             raise ValueError(f"Failed to get voices. Error: {response.text}")
 
         voices = response.json()
-        # TODO: Update the API to return the embedding as a list of floats rather than string.
-        if not skip_embeddings:
-            for voice in voices:
+        for voice in voices:
+            if "embedding" in voice and isinstance(voice["embedding"], str):
                 voice["embedding"] = json.loads(voice["embedding"])
         return {voice["name"]: voice for voice in voices}
 
+    @retry_on_connection_error(max_retries=MAX_RETRIES, backoff_factor=BACKOFF_FACTOR, logger=logger)
     def get_voice_embedding(
         self, *, voice_id: str = None, filepath: str = None, link: str = None
     ) -> Embedding:
@@ -178,18 +188,18 @@ class CartesiaTTS:
 
         if voice_id:
             url = f"{self._http_url()}/voices/embedding/{voice_id}"
-            response = httpx.get(url, headers=self.headers)
+            response = httpx.get(url, headers=self.headers, timeout=DEFAULT_TIMEOUT)
         elif filepath:
             url = f"{self._http_url()}/voices/clone/clip"
             files = {"clip": open(filepath, "rb")}
             headers = self.headers.copy()
             # The default content type of JSON is incorrect for file uploads
             headers.pop("Content-Type")
-            response = httpx.post(url, headers=headers, files=files)
+            response = httpx.post(url, headers=headers, files=files, timeout=DEFAULT_TIMEOUT)
         elif link:
             url = f"{self._http_url()}/voices/clone/url"
             params = {"link": link}
-            response = httpx.post(url, headers=self.headers, params=params)
+            response = httpx.post(url, headers=self.headers, params=params, timeout=DEFAULT_TIMEOUT)
 
         if not response.is_success:
             raise ValueError(
@@ -199,9 +209,10 @@ class CartesiaTTS:
 
         # Handle successful response
         out = response.json()
-        if isinstance(out["embedding"], str):
-            out["embedding"] = json.loads(out["embedding"])
-        return out["embedding"]
+        embedding = out["embedding"]
+        if isinstance(embedding, str):
+            embedding = json.loads(embedding)
+        return embedding
 
     def refresh_websocket(self):
         """Refresh the websocket connection.
@@ -209,15 +220,11 @@ class CartesiaTTS:
         Note:
             The connection is synchronous.
         """
-        if self.websocket and not self._is_websocket_closed():
-            self.websocket.close()
-        route = "audio/websocket"
-        if self.experimental_ws_handle_interrupts:
-            route = f"experimental/{route}"
-        self.websocket = connect(
-            f"{self._ws_url()}/{route}?api_key={self.api_key}",
-            close_timeout=None,
-        )
+        if self.websocket is None or self._is_websocket_closed():
+            route = "audio/websocket"
+            if self.experimental_ws_handle_interrupts:
+                route = f"experimental/{route}"
+            self.websocket = connect(f"{self._ws_url()}/{route}?api_key={self.api_key}")
 
     def _is_websocket_closed(self):
         return self.websocket.socket.fileno() == -1
@@ -240,20 +247,23 @@ class CartesiaTTS:
         self,
         *,
         transcript: str,
+        voice: Embedding,
+        model_id: str,
+        output_format: str,
         duration: int = None,
         chunk_time: float = None,
-        voice: Embedding = None,
     ) -> Dict[str, Any]:
+        """Create the request body for a stream request.
+
+        Note that anything that's not provided will use a default if available or be
+        filtered out otherwise.
         """
-        Create the request body for a stream request.
-        Note that anything that's not provided will use a default if available or be filtered out otherwise.
-        """
-        body = dict(transcript=transcript, model_id=DEFAULT_MODEL_ID, voice=voice)
+        body = dict(transcript=transcript, model_id=model_id, voice=voice)
 
         optional_body = dict(
             duration=duration,
             chunk_time=chunk_time,
-            voice=voice,
+            output_format=output_format,
         )
         body.update({k: v for k, v in optional_body.items() if v is not None})
 
@@ -263,25 +273,26 @@ class CartesiaTTS:
         self,
         *,
         transcript: str,
+        voice: Embedding,
+        model_id: str = DEFAULT_MODEL_ID,
         duration: int = None,
         chunk_time: float = None,
-        voice: Embedding = None,
         stream: bool = False,
         websocket: bool = True,
+        output_format: str = "fp32",
     ) -> Union[AudioOutput, Generator[AudioOutput, None, None]]:
         """Generate audio from a transcript.
 
         Args:
-            transcript: The text to generate audio for.
-            duration: The maximum duration of the audio in seconds.
-            chunk_time: How long each audio segment should be in seconds.
+            transcript (str): The text to generate audio for.
+            voice (Embedding (List[float])): The voice to use for generating audio.
+            duration (int, optional): The maximum duration of the audio in seconds.
+            chunk_time (float, optional): How long each audio segment should be in seconds.
                 This should not need to be adjusted.
-            voice: The voice to use for generating audio.
-                This can either be a voice id (string) or an embedding vector (List[float]).
-            stream: Whether to stream the audio or not.
-                If ``True`` this function returns a generator.
-            websocket: Whether to use a websocket for streaming audio.
-                Using the websocket reduces latency by pre-poning the handshake.
+            stream (bool, optional): Whether to stream the audio or not.
+                If True this function returns a generator. False by default.
+            websocket (bool, optional): Whether to use a websocket for streaming audio.
+                Using the websocket reduces latency by pre-poning the handshake. True by default.
 
         Returns:
             A generator if `stream` is True, otherwise a dictionary.
@@ -292,13 +303,18 @@ class CartesiaTTS:
         self._check_inputs(transcript, duration, chunk_time)
 
         body = self._generate_request_body(
-            transcript=transcript, duration=duration, chunk_time=chunk_time, voice=voice
+            transcript=transcript, 
+            voice=voice, 
+            model_id=model_id,
+            duration=duration, 
+            chunk_time=chunk_time,
+            output_format=output_format,
         )
 
         if websocket:
             generator = self._generate_ws(body)
         else:
-            generator = self._generate_http(body)
+            generator = self._generate_http_wrapper(body)
 
         if stream:
             return generator
@@ -312,12 +328,23 @@ class CartesiaTTS:
 
         return {"audio": b"".join(chunks), "sampling_rate": sampling_rate}
 
+    @retry_on_connection_error(max_retries=MAX_RETRIES, backoff_factor=BACKOFF_FACTOR, logger=logger)
+    def _generate_http_wrapper(self, body: Dict[str, Any]):
+        """Need to wrap the http generator in a function for the retry decorator to work."""
+        try:
+            for chunk in self._generate_http(body):
+                yield chunk
+        except Exception as e:
+            logger.error(f"Failed to generate audio. {e}")
+            raise e
+
     def _generate_http(self, body: Dict[str, Any]):
         response = requests.post(
-            f"{self._http_url()}/audio/stream",
+            f"{self._http_url()}/audio/sse",
             stream=True,
             data=json.dumps(body),
             headers=self.headers,
+            timeout=(DEFAULT_TIMEOUT, DEFAULT_TIMEOUT),
         )
         if not response.ok:
             raise ValueError(f"Failed to generate audio. {response.text}")
@@ -356,6 +383,8 @@ class CartesiaTTS:
         try:
             while True:
                 response = json.loads(self.websocket.recv())
+                if "error" in response:
+                    raise RuntimeError(f"Error generating audio:\n{response['error']}")
                 if response["done"]:
                     break
 
@@ -370,7 +399,43 @@ class CartesiaTTS:
             if self.experimental_ws_handle_interrupts:
                 self.websocket.send(json.dumps({"context_id": context_id, "action": "cancel"}))
         except Exception as e:
+            # Close the websocket connection if an error occurs.
+            if self.websocket and not self._is_websocket_closed():
+                self.websocket.close()
             raise RuntimeError(f"Failed to generate audio. {response}") from e
+        finally:
+            # Ensure the websocket is ultimately closed.
+            if self.websocket and not self._is_websocket_closed():
+                self.websocket.close()
+
+    @retry_on_connection_error(max_retries=MAX_RETRIES, backoff_factor=BACKOFF_FACTOR, logger=logger)
+    def transcribe(self, raw_audio: Union[bytes, str]) -> str:
+        raw_audio_bytes, headers = self.prepare_audio_and_headers(raw_audio)
+        response = httpx.post(
+            f"{self._http_url()}/audio/transcriptions",
+            headers=headers,
+            files={"clip": ("input.wav", raw_audio_bytes)},
+            timeout=DEFAULT_TIMEOUT,
+        )
+
+        if not response.is_success:
+            raise ValueError(f"Failed to transcribe audio. Error: {response.text()}")
+
+        transcript = response.json()
+        return transcript["text"]
+
+
+    def prepare_audio_and_headers(
+        self, raw_audio: Union[bytes, str]
+    ) -> Tuple[bytes, Dict[str, Any]]:
+        if isinstance(raw_audio, str):
+            with open(raw_audio, "rb") as f:
+                raw_audio_bytes = f.read()
+        else:
+            raw_audio_bytes = raw_audio
+        # application/json is not the right content type for this request
+        headers = {k: v for k, v in self.headers.items() if k != "Content-Type"}
+        return raw_audio_bytes, headers
 
     def _http_url(self):
         prefix = "http" if "localhost" in self.base_url else "https"
@@ -380,63 +445,119 @@ class CartesiaTTS:
         prefix = "ws" if "localhost" in self.base_url else "wss"
         return f"{prefix}://{self.base_url}/{self.api_version}"
 
-    def __del__(self):
-        if self.websocket.socket.fileno() > -1:
+    def close(self):
+        if self.websocket and not self._is_websocket_closed():
             self.websocket.close()
 
+    def __del__(self):
+        self.close()
+
+    def __enter__(self):
+        self.refresh_websocket()
+        return self
+
+    def __exit__(
+        self,
+        exc_type: Union[type, None],
+        exc: Union[BaseException, None],
+        exc_tb: Union[TracebackType, None],
+    ):
+        self.close()
+
 
 class AsyncCartesiaTTS(CartesiaTTS):
     def __init__(self, *, api_key: str = None, experimental_ws_handle_interrupts: bool = False):
-        self.timeout = aiohttp.ClientTimeout(total=DEFAULT_TIMEOUT)
-        self.connector = aiohttp.TCPConnector(limit=DEFAULT_NUM_CONNECTIONS)
-        self._session = aiohttp.ClientSession(timeout=self.timeout, connector=self.connector)
+        self._session = None
+        self._loop = None
         super().__init__(
             api_key=api_key, experimental_ws_handle_interrupts=experimental_ws_handle_interrupts
         )
-
-    def refresh_websocket(self):
-        pass  # do not load the websocket for the client until asynchronously when it is needed
-
-    async def _async_refresh_websocket(self):
+    
+    async def _get_session(self):
+        current_loop = asyncio.get_event_loop()
+        if self._loop is not current_loop:
+            # If the loop has changed, close the session and create a new one.
+            await self.close()
+        if self._session is None or self._session.closed:
+            timeout = aiohttp.ClientTimeout(total=DEFAULT_TIMEOUT)
+            connector = aiohttp.TCPConnector(limit=DEFAULT_NUM_CONNECTIONS)
+            self._session = aiohttp.ClientSession(
+                timeout=timeout, connector=connector
+            )
+            self._loop = current_loop
+        return self._session
+    
+    async def refresh_websocket(self):
         """Refresh the websocket connection."""
-        if self.websocket and not self._is_websocket_closed():
-            self.websocket.close()
-        route = "audio/websocket"
-        if self.experimental_ws_handle_interrupts:
-            route = f"experimental/{route}"
-        self.websocket = await self._session.ws_connect(
-            f"{self._ws_url()}/{route}?api_key={self.api_key}"
-        )
+        if self.websocket is None or self._is_websocket_closed():
+            route = "audio/websocket"
+            if self.experimental_ws_handle_interrupts:
+                route = f"experimental/{route}"
+            session = await self._get_session()
+            self.websocket = await session.ws_connect(
+                f"{self._ws_url()}/{route}?api_key={self.api_key}"
+            )
+    
+    def _is_websocket_closed(self):
+        return self.websocket.closed
+
+    async def close(self):
+        """This method closes the websocket and the session.
+        
+        It is *strongly* recommended to call this method when you are done using the client.
+        """
+        if self.websocket is not None and not self._is_websocket_closed():
+            await self.websocket.close()
+        if self._session is not None and not self._session.closed:
+            await self._session.close()
 
     async def generate(
         self,
         *,
         transcript: str,
+        voice: Embedding,
+        model_id: str = DEFAULT_MODEL_ID,
         duration: int = None,
         chunk_time: float = None,
-        voice: Embedding = None,
         stream: bool = False,
         websocket: bool = True,
+        output_format: str = "fp32"
     ) -> Union[AudioOutput, AsyncGenerator[AudioOutput, None]]:
         """Asynchronously generate audio from a transcript.
         NOTE: This overrides the non-asynchronous generate method from the base class.
+
         Args:
-            transcript: The text to generate audio for.
-            voice: The embedding to use for generating audio.
-            options: The options to use for generating audio. See :class:`GenerateOptions`.
+            transcript (str): The text to generate audio for.
+            voice (Embedding (List[float])): The voice to use for generating audio.
+            duration (int, optional): The maximum duration of the audio in seconds.
+            chunk_time (float, optional): How long each audio segment should be in seconds.
+                This should not need to be adjusted.
+            stream (bool, optional): Whether to stream the audio or not.
+                If True this function returns a generator. False by default.
+            websocket (bool, optional): Whether to use a websocket for streaming audio.
+                Using the websocket reduces latency by pre-poning the handshake. True by default.
+
         Returns:
-            A dictionary containing the following:
-                * "audio": The audio as a 1D numpy array.
+            A generator if `stream` is True, otherwise a dictionary.
+            Dictionary from both generator and non-generator return types have the following keys:
+                * "audio": The audio as a bytes buffer.
                 * "sampling_rate": The sampling rate of the audio.
         """
+        self._check_inputs(transcript, duration, chunk_time)
+
         body = self._generate_request_body(
-            transcript=transcript, duration=duration, chunk_time=chunk_time, voice=voice
+            transcript=transcript, 
+            voice=voice,
+            model_id=model_id,
+            duration=duration, 
+            chunk_time=chunk_time,
+            output_format=output_format,
         )
 
         if websocket:
             generator = self._generate_ws(body)
         else:
-            generator = self._generate_http(body)
+            generator = self._generate_http_wrapper(body)
 
         if stream:
             return generator
@@ -450,12 +571,23 @@ class AsyncCartesiaTTS(CartesiaTTS):
 
         return {"audio": b"".join(chunks), "sampling_rate": sampling_rate}
 
+    @retry_on_connection_error_async(max_retries=MAX_RETRIES, backoff_factor=BACKOFF_FACTOR, logger=logger)
+    async def _generate_http_wrapper(self, body: Dict[str, Any]):
+        """Need to wrap the http generator in a function for the retry decorator to work."""
+        try:
+          async for chunk in self._generate_http(body):
+              yield chunk
+        except Exception as e:
+            logger.error(f"Failed to generate audio. {e}")
+            raise e
+
     async def _generate_http(self, body: Dict[str, Any]):
-        async with self._session.post(
-            f"{self._http_url()}/audio/stream", data=json.dumps(body), headers=self.headers
+        session = await self._get_session()
+        async with session.post(
+            f"{self._http_url()}/audio/sse", data=json.dumps(body), headers=self.headers
         ) as response:
-            if response.status < 200 or response.status >= 300:
-                raise ValueError(f"Failed to generate audio. {response.text}")
+            if not response.ok:
+                raise ValueError(f"Failed to generate audio. {await response.text()}")
 
             buffer = ""
             async for chunk_bytes in response.content.iter_any():
@@ -478,7 +610,7 @@ class AsyncCartesiaTTS(CartesiaTTS):
             route = f"experimental/{route}"
 
         if not self.websocket or self._is_websocket_closed():
-            await self._async_refresh_websocket()
+            await self.refresh_websocket()
 
         ws = self.websocket
         if context_id is None:
@@ -502,17 +634,29 @@ class AsyncCartesiaTTS(CartesiaTTS):
             if self.experimental_ws_handle_interrupts:
                 await ws.send_json({"context_id": context_id, "action": "cancel"})
         except Exception as e:
-            raise RuntimeError(f"Failed to generate audio. {response}") from e
-
-    def _is_websocket_closed(self):
-        return self.websocket.closed
-
-    async def cleanup(self):
-        if self.websocket is not None and not self._is_websocket_closed():
-            await self.websocket.close()
-        if not self._session.closed:
-            await self._session.close()
+            if self.websocket and not self._is_websocket_closed():
+                await self.websocket.close()
+            raise RuntimeError(f"Failed to generate audio. {await response.text()}") from e
+        finally:
+            # Ensure the websocket is ultimately closed.
+            if self.websocket and not self._is_websocket_closed():
+                await self.websocket.close()
+
+    async def transcribe(self, raw_audio: Union[bytes, str]) -> str:
+        raw_audio_bytes, headers = self.prepare_audio_and_headers(raw_audio)
+        data = aiohttp.FormData()
+        data.add_field("clip", raw_audio_bytes, filename="input.wav", content_type="audio/wav")
+        session = await self._get_session()
+
+        async with session.post(
+            f"{self._http_url()}/audio/transcriptions", headers=headers, data=data
+        ) as response:
+            if not response.ok:
+                raise ValueError(f"Failed to transcribe audio. Error: {await response.text()}")
 
+            transcript = await response.json()
+            return transcript["text"]
+        
     def __del__(self):
         try:
             loop = asyncio.get_running_loop()
@@ -520,6 +664,18 @@ class AsyncCartesiaTTS(CartesiaTTS):
             loop = None
 
         if loop is None:
-            asyncio.run(self.cleanup())
+            asyncio.run(self.close())
         else:
-            loop.create_task(self.cleanup())
+            loop.create_task(self.close())
+
+    async def __aenter__(self):
+        await self.refresh_websocket()
+        return self
+
+    async def __aexit__(
+        self,
+        exc_type: Union[type, None],
+        exc: Union[BaseException, None],
+        exc_tb: Union[TracebackType, None],
+    ):
+        await self.close()

cartesia/utils.py

@@ -0,0 +1,65 @@
+import time
+
+from aiohttp.client_exceptions import ServerDisconnectedError
+import asyncio
+from functools import wraps
+from http.client import RemoteDisconnected
+from httpx import TimeoutException
+from requests.exceptions import ConnectionError
+
+def retry_on_connection_error(max_retries=3, backoff_factor=1, logger=None):
+    """Retry a function if a ConnectionError, RemoteDisconnected, ServerDisconnectedError, or TimeoutException occurs.
+
+    Args:
+        max_retries (int): The maximum number of retries.
+        backoff_factor (int): The factor to increase the delay between retries.
+        logger (logging.Logger): The logger to use for logging.
+    """
+    def decorator(func):
+        @wraps(func)
+        def wrapper(*args, **kwargs):
+            retry_count = 0
+            while retry_count < max_retries:
+                try:
+                    return func(*args, **kwargs)
+                except (ConnectionError, RemoteDisconnected, ServerDisconnectedError, TimeoutException) as e:
+                    logger.info(f"Retrying after exception: {e}")
+                    retry_count += 1
+                    if retry_count < max_retries:
+                        delay = backoff_factor * (2 ** (retry_count - 1))
+                        logger.warn(f"Attempt {retry_count + 1}/{max_retries} in {delay} seconds...")
+                        time.sleep(delay)
+                    else:
+                        raise Exception(f"Exception occurred after {max_retries} tries.") from e
+        return wrapper
+    return decorator
+
+def retry_on_connection_error_async(max_retries=3, backoff_factor=1, logger=None):
+    """Retry an asynchronous function if a ConnectionError, RemoteDisconnected, ServerDisconnectedError, or TimeoutException occurs.
+
+    Args:
+        max_retries (int): The maximum number of retries.
+        backoff_factor (int): The factor to increase the delay between retries.
+        logger (logging.Logger): The logger to use for logging.
+    """
+    def decorator(func):
+        @wraps(func)
+        async def wrapper(*args, **kwargs):
+            retry_count = 0
+            while retry_count < max_retries:
+                try:
+                    async for chunk in func(*args, **kwargs):
+                        yield chunk
+                    # If the function completes without raising an exception return
+                    return
+                except (ConnectionError, RemoteDisconnected, ServerDisconnectedError, TimeoutException) as e:
+                    logger.info(f"Retrying after exception: {e}")
+                    retry_count += 1
+                    if retry_count < max_retries:
+                        delay = backoff_factor * (2 ** (retry_count - 1))
+                        logger.warn(f"Attempt {retry_count + 1}/{max_retries} in {delay} seconds...")
+                        await asyncio.sleep(delay)
+                    else:
+                        raise Exception(f"Exception occurred after {max_retries} tries.") from e
+        return wrapper
+    return decorator

cartesia/version.py

@@ -1 +1 @@
-__version__ = "0.0.5rc1"
+__version__ = "0.0.6"

{cartesia-0.0.5rc1.dist-info → cartesia-0.0.6.dist-info}/METADATA

@@ -1,6 +1,6 @@
 Metadata-Version: 2.1
 Name: cartesia
-Version: 0.0.5rc1
+Version: 0.0.6
 Summary: The official Python library for the Cartesia API.
 Home-page: 
 Author: Cartesia, Inc.
@@ -16,25 +16,17 @@ Requires-Dist: pytest-asyncio
 Requires-Dist: requests
 Requires-Dist: websockets
 Provides-Extra: all
-Requires-Dist: pre-commit ; extra == 'all'
-Requires-Dist: docformatter ; extra == 'all'
-Requires-Dist: black ==24.1.1 ; extra == 'all'
-Requires-Dist: isort ==5.13.2 ; extra == 'all'
-Requires-Dist: flake8 ==7.0.0 ; extra == 'all'
-Requires-Dist: flake8-bugbear ==24.2.6 ; extra == 'all'
 Requires-Dist: pytest >=8.0.2 ; extra == 'all'
 Requires-Dist: pytest-cov >=4.1.0 ; extra == 'all'
 Requires-Dist: twine ; extra == 'all'
+Requires-Dist: setuptools ; extra == 'all'
+Requires-Dist: wheel ; extra == 'all'
 Provides-Extra: dev
-Requires-Dist: pre-commit ; extra == 'dev'
-Requires-Dist: docformatter ; extra == 'dev'
-Requires-Dist: black ==24.1.1 ; extra == 'dev'
-Requires-Dist: isort ==5.13.2 ; extra == 'dev'
-Requires-Dist: flake8 ==7.0.0 ; extra == 'dev'
-Requires-Dist: flake8-bugbear ==24.2.6 ; extra == 'dev'
 Requires-Dist: pytest >=8.0.2 ; extra == 'dev'
 Requires-Dist: pytest-cov >=4.1.0 ; extra == 'dev'
 Requires-Dist: twine ; extra == 'dev'
+Requires-Dist: setuptools ; extra == 'dev'
+Requires-Dist: wheel ; extra == 'dev'
 
 
 # Cartesia Python API Library
@@ -60,13 +52,14 @@ client = CartesiaTTS(api_key=os.environ.get("CARTESIA_API_KEY"))
 voices = client.get_voices()
 voice = client.get_voice_embedding(voice_id=voices["Graham"]["id"])
 transcript = "Hello! Welcome to Cartesia"
+model_id = "genial-planet-1346" # (Optional) We'll specify a default if you don't have a specific model in mind
 
 p = pyaudio.PyAudio()
 
 stream = None
 
 # Generate and stream audio
-for output in client.generate(transcript=transcript, voice=voice, stream=True):
+for output in client.generate(transcript=transcript, voice=voice, model_id=model_id, stream=True):
     buffer = output["audio"]
     rate = output["sampling_rate"]
 
@@ -84,26 +77,68 @@ stream.close()
 p.terminate()
 ```
 
-If you are using Jupyter Notebook or JupyterLab, you can use IPython.display.Audio to play the generated audio directly in the notebook. Here's an example:
+You can also use the async client if you want to make asynchronous API calls:
+```python
+from cartesia.tts import AsyncCartesiaTTS
+import asyncio
+import pyaudio
+import os
+
+async def write_stream():
+    client = AsyncCartesiaTTS(api_key=os.environ.get("CARTESIA_API_KEY"))
+    voices = client.get_voices()
+    voice = client.get_voice_embedding(voice_id=voices["Graham"]["id"])
+    transcript = "Hello! Welcome to Cartesia"
+    model_id = "genial-planet-1346" # (Optional) We'll specify a default if you don't have a specific model in mind
+
+    p = pyaudio.PyAudio()
+
+    stream = None
+
+    # Generate and stream audio
+    async for output in await client.generate(transcript=transcript, voice=voice, model_id=model_id, stream=True):
+        buffer = output["audio"]
+        rate = output["sampling_rate"]
+
+        if not stream:
+            stream = p.open(format=pyaudio.paFloat32,
+                            channels=1,
+                            rate=rate,
+                            output=True)
+
+        # Write the audio data to the stream
+        stream.write(buffer)
+
+    stream.stop_stream()
+    stream.close()
+    p.terminate()
+
+asyncio.run(write_stream())
+```
+
+If you are using Jupyter Notebook or JupyterLab, you can use IPython.display.Audio to play the generated audio directly in the notebook.
+Additionally, in these notebook examples we show how to use the client as a context manager (though this is not required).
 
 ```python
-from cartesia.tts import CartesiaTTS
 from IPython.display import Audio
 import io
 import os
+import numpy as np
 
-client = CartesiaTTS(api_key=os.environ.get("CARTESIA_API_KEY"))
-voices = client.get_voices()
-voice = client.get_voice_embedding(voice_id=voices["Graham"]["id"])
-transcript = "Hello! Welcome to Cartesia"
+from cartesia.tts import CartesiaTTS
 
-# Create a BytesIO object to store the audio data
-audio_data = io.BytesIO()
+with CartesiaTTS(api_key=os.environ.get("CARTESIA_API_KEY")) as client:
+    voices = client.get_voices()
+    voice = client.get_voice_embedding(voice_id=voices["Graham"]["id"])
+    transcript = "Hello! Welcome to Cartesia"
 
-# Generate and stream audio
-for output in client.generate(transcript=transcript, voice=voice, stream=True):
-    buffer = output["audio"]
-    audio_data.write(buffer)
+    # Create a BytesIO object to store the audio data
+    audio_data = io.BytesIO()
+
+    # Generate and stream audio
+    for output in client.generate(transcript=transcript, voice=voice, stream=True):
+        buffer = output["audio"]
+        audio_data.write(buffer)
 
 # Set the cursor position to the beginning of the BytesIO object
 audio_data.seek(0)
@@ -115,25 +150,27 @@ audio = Audio(np.frombuffer(audio_data.read(), dtype=np.float32), rate=output["s
 display(audio)
 ```
 
-You can also use the async client if you want to make asynchronous API calls. The usage is very similar:
+Below is the same example using the async client:
 ```python
-from cartesia.tts import AsyncCartesiaTTS
 from IPython.display import Audio
 import io
 import os
+import numpy as np
 
-client = AsyncCartesiaTTS(api_key=os.environ.get("CARTESIA_API_KEY"))
-voices = client.get_voices()
-voice = client.get_voice_embedding(voice_id=voices["Graham"]["id"])
-transcript = "Hello! Welcome to Cartesia"
+from cartesia.tts import AsyncCartesiaTTS
 
-# Create a BytesIO object to store the audio data
-audio_data = io.BytesIO()
+async with AsyncCartesiaTTS(api_key=os.environ.get("CARTESIA_API_KEY")) as client:
+    voices = client.get_voices()
+    voice = client.get_voice_embedding(voice_id=voices["Graham"]["id"])
+    transcript = "Hello! Welcome to Cartesia"
 
-# Generate and stream audio
-async for output in client.generate(transcript=transcript, voice=voice, stream=True):
-    buffer = output["audio"]
-    audio_data.write(buffer)
+    # Create a BytesIO object to store the audio data
+    audio_data = io.BytesIO()
+
+    # Generate and stream audio
+    async for output in await client.generate(transcript=transcript, voice=voice, stream=True):
+        buffer = output["audio"]
+        audio_data.write(buffer)
 
 # Set the cursor position to the beginning of the BytesIO object
 audio_data.seek(0)

cartesia-0.0.6.dist-info/RECORD

@@ -0,0 +1,8 @@
+cartesia/__init__.py,sha256=uIc9xGNPs8_A6eAvbTUY1geazunYoEZVWFKhCwC9TRA,102
+cartesia/tts.py,sha256=YjOW8mlvvPbHblhcMUY71RsKn77K_WQi8ySok3ifeJg,26734
+cartesia/utils.py,sha256=GoTJe8LZ3WpS4hXkwoZauPYjo7Mbx7BvbBjAX5vEbwg,3024
+cartesia/version.py,sha256=QiiYsv0kcJaB8wCWyT-FnI2b6be87HA-CrrIUn8LQhg,22
+cartesia-0.0.6.dist-info/METADATA,sha256=yhq7LSvLrboBPI3IOcLTvaneisqhq-v1VMQ0sKBq8kk,5974
+cartesia-0.0.6.dist-info/WHEEL,sha256=DZajD4pwLWue70CAfc7YaxT1wLUciNBvN_TTcvXpltE,110
+cartesia-0.0.6.dist-info/top_level.txt,sha256=rTX4HnnCegMxl1FK9czpVC7GAvf3SwDzPG65qP-BS4w,9
+cartesia-0.0.6.dist-info/RECORD,,

{cartesia-0.0.5rc1.dist-info → cartesia-0.0.6.dist-info}/WHEEL

@@ -1,5 +1,5 @@
 Wheel-Version: 1.0
-Generator: bdist_wheel (0.41.2)
+Generator: bdist_wheel (0.43.0)
 Root-Is-Purelib: true
 Tag: py2-none-any
 Tag: py3-none-any

cartesia-0.0.5rc1.dist-info/RECORD

@@ -1,7 +0,0 @@
-cartesia/__init__.py,sha256=m8BX-qLjsMoI_JZtgf3jNi8R3cBZqYy-z4oEhYeJLdI,64
-cartesia/tts.py,sha256=yPLz41AR0oAYPUNW48mqmwEEbLBHCnbaK_wPT0iFBVk,20543
-cartesia/version.py,sha256=VkI5lk2CFatZR200RqGd8cBjTnMDmhtZW7DI6mPe6n4,25
-cartesia-0.0.5rc1.dist-info/METADATA,sha256=632D6iZ2IU3MLySAnMtwV2zQA38XkQv1rfFF4iRdAco,4893
-cartesia-0.0.5rc1.dist-info/WHEEL,sha256=iYlv5fX357PQyRT2o6tw1bN-YcKFFHKqB_LwHO5wP-g,110
-cartesia-0.0.5rc1.dist-info/top_level.txt,sha256=rTX4HnnCegMxl1FK9czpVC7GAvf3SwDzPG65qP-BS4w,9
-cartesia-0.0.5rc1.dist-info/RECORD,,