cartesia 2.0.4__py3-none-any.whl → 2.0.6__py3-none-any.whl

cartesia/__init__.py

@@ -1,6 +1,6 @@
 # This file was auto-generated by Fern from our API Definition.
 
-from . import api_status, auth, datasets, embedding, infill, tts, voice_changer, voices
+from . import api_status, auth, datasets, embedding, infill, stt, tts, voice_changer, voices
 from .api_status import ApiInfo, ApiInfoParams
 from .auth import TokenGrant, TokenGrantParams, TokenRequest, TokenRequestParams, TokenResponse, TokenResponseParams
 from .client import AsyncCartesia, Cartesia
@@ -19,6 +19,32 @@ from .datasets import (
 )
 from .embedding import Embedding
 from .environment import CartesiaEnvironment
+from .stt import (
+    DoneMessage,
+    DoneMessageParams,
+    ErrorMessage,
+    ErrorMessageParams,
+    FlushDoneMessage,
+    FlushDoneMessageParams,
+    StreamingTranscriptionResponse,
+    StreamingTranscriptionResponseParams,
+    StreamingTranscriptionResponse_Done,
+    StreamingTranscriptionResponse_DoneParams,
+    StreamingTranscriptionResponse_Error,
+    StreamingTranscriptionResponse_ErrorParams,
+    StreamingTranscriptionResponse_FlushDone,
+    StreamingTranscriptionResponse_FlushDoneParams,
+    StreamingTranscriptionResponse_Transcript,
+    StreamingTranscriptionResponse_TranscriptParams,
+    SttEncoding,
+    TimestampGranularity,
+    TranscriptMessage,
+    TranscriptMessageParams,
+    TranscriptionResponse,
+    TranscriptionResponseParams,
+    TranscriptionWord,
+    TranscriptionWordParams,
+)
 from .tts import (
     CancelContextRequest,
     CancelContextRequestParams,
@@ -49,6 +75,8 @@ from .tts import (
     RawOutputFormatParams,
     Speed,
     SpeedParams,
+    SseOutputFormat,
+    SseOutputFormatParams,
     SupportedLanguage,
     TtsRequest,
     TtsRequestEmbeddingSpecifier,
@@ -58,6 +86,8 @@ from .tts import (
     TtsRequestParams,
     TtsRequestVoiceSpecifier,
     TtsRequestVoiceSpecifierParams,
+    TtssseRequest,
+    TtssseRequestParams,
     WavOutputFormat,
     WavOutputFormatParams,
     WebSocketBaseResponse,
@@ -173,13 +203,19 @@ __all__ = [
     "DatasetFile",
     "DatasetFileParams",
     "DatasetParams",
+    "DoneMessage",
+    "DoneMessageParams",
     "Embedding",
     "EmbeddingResponse",
     "EmbeddingResponseParams",
     "EmbeddingSpecifier",
     "EmbeddingSpecifierParams",
     "Emotion",
+    "ErrorMessage",
+    "ErrorMessageParams",
     "FilePurpose",
+    "FlushDoneMessage",
+    "FlushDoneMessageParams",
     "FlushId",
     "Gender",
     "GenderPresentation",
@@ -227,6 +263,8 @@ __all__ = [
     "RawOutputFormatParams",
     "Speed",
     "SpeedParams",
+    "SseOutputFormat",
+    "SseOutputFormatParams",
     "StreamingResponse",
     "StreamingResponseParams",
     "StreamingResponse_Chunk",
@@ -235,13 +273,31 @@ __all__ = [
     "StreamingResponse_DoneParams",
     "StreamingResponse_Error",
     "StreamingResponse_ErrorParams",
+    "StreamingTranscriptionResponse",
+    "StreamingTranscriptionResponseParams",
+    "StreamingTranscriptionResponse_Done",
+    "StreamingTranscriptionResponse_DoneParams",
+    "StreamingTranscriptionResponse_Error",
+    "StreamingTranscriptionResponse_ErrorParams",
+    "StreamingTranscriptionResponse_FlushDone",
+    "StreamingTranscriptionResponse_FlushDoneParams",
+    "StreamingTranscriptionResponse_Transcript",
+    "StreamingTranscriptionResponse_TranscriptParams",
+    "SttEncoding",
     "SupportedLanguage",
+    "TimestampGranularity",
     "TokenGrant",
     "TokenGrantParams",
     "TokenRequest",
     "TokenRequestParams",
     "TokenResponse",
     "TokenResponseParams",
+    "TranscriptMessage",
+    "TranscriptMessageParams",
+    "TranscriptionResponse",
+    "TranscriptionResponseParams",
+    "TranscriptionWord",
+    "TranscriptionWordParams",
     "TtsRequest",
     "TtsRequestEmbeddingSpecifier",
     "TtsRequestEmbeddingSpecifierParams",
@@ -250,6 +306,8 @@ __all__ = [
     "TtsRequestParams",
     "TtsRequestVoiceSpecifier",
     "TtsRequestVoiceSpecifierParams",
+    "TtssseRequest",
+    "TtssseRequestParams",
     "UpdateVoiceRequest",
     "UpdateVoiceRequestParams",
     "Voice",
@@ -307,6 +365,7 @@ __all__ = [
     "datasets",
     "embedding",
     "infill",
+    "stt",
     "tts",
     "voice_changer",
     "voices",

cartesia/auth/client.py

@@ -22,7 +22,7 @@ class AuthClient:
     def access_token(
         self,
         *,
-        grants: TokenGrantParams,
+        grants: typing.Optional[TokenGrantParams] = OMIT,
         expires_in: typing.Optional[int] = OMIT,
         request_options: typing.Optional[RequestOptions] = None,
     ) -> TokenResponse:
@@ -31,8 +31,8 @@ class AuthClient:
 
         Parameters
         ----------
-        grants : TokenGrantParams
-            The permissions to be granted via the token.
+        grants : typing.Optional[TokenGrantParams]
+            The permissions to be granted via the token. Both TTS and STT grants are optional - specify only the capabilities you need.
 
         expires_in : typing.Optional[int]
             The number of seconds the token will be valid for since the time of generation. The maximum is 1 hour (3600 seconds).
@@ -52,7 +52,7 @@ class AuthClient:
             api_key="YOUR_API_KEY",
         )
         client.auth.access_token(
-            grants={"tts": True},
+            grants={"tts": True, "stt": True},
             expires_in=60,
         )
         """
@@ -90,7 +90,7 @@ class AsyncAuthClient:
     async def access_token(
         self,
         *,
-        grants: TokenGrantParams,
+        grants: typing.Optional[TokenGrantParams] = OMIT,
         expires_in: typing.Optional[int] = OMIT,
         request_options: typing.Optional[RequestOptions] = None,
     ) -> TokenResponse:
@@ -99,8 +99,8 @@ class AsyncAuthClient:
 
         Parameters
         ----------
-        grants : TokenGrantParams
-            The permissions to be granted via the token.
+        grants : typing.Optional[TokenGrantParams]
+            The permissions to be granted via the token. Both TTS and STT grants are optional - specify only the capabilities you need.
 
         expires_in : typing.Optional[int]
             The number of seconds the token will be valid for since the time of generation. The maximum is 1 hour (3600 seconds).
@@ -125,7 +125,7 @@ class AsyncAuthClient:
 
         async def main() -> None:
             await client.auth.access_token(
-                grants={"tts": True},
+                grants={"tts": True, "stt": True},
                 expires_in=60,
             )
 

cartesia/auth/requests/token_grant.py

@@ -1,10 +1,16 @@
 # This file was auto-generated by Fern from our API Definition.
 
 import typing_extensions
+import typing_extensions
 
 
 class TokenGrantParams(typing_extensions.TypedDict):
-    tts: bool
+    tts: typing_extensions.NotRequired[bool]
     """
     The `tts` grant allows the token to be used to access any TTS endpoint.
     """
+
+    stt: typing_extensions.NotRequired[bool]
+    """
+    The `stt` grant allows the token to be used to access any STT endpoint.
+    """

cartesia/auth/requests/token_request.py

@@ -1,14 +1,14 @@
 # This file was auto-generated by Fern from our API Definition.
 
 import typing_extensions
-from .token_grant import TokenGrantParams
 import typing_extensions
+from .token_grant import TokenGrantParams
 
 
 class TokenRequestParams(typing_extensions.TypedDict):
-    grants: TokenGrantParams
+    grants: typing_extensions.NotRequired[TokenGrantParams]
     """
-    The permissions to be granted via the token.
+    The permissions to be granted via the token. Both TTS and STT grants are optional - specify only the capabilities you need.
     """
 
     expires_in: typing_extensions.NotRequired[int]

cartesia/auth/types/token_grant.py

@@ -1,17 +1,22 @@
 # This file was auto-generated by Fern from our API Definition.
 
 from ...core.pydantic_utilities import UniversalBaseModel
+import typing
 import pydantic
 from ...core.pydantic_utilities import IS_PYDANTIC_V2
-import typing
 
 
 class TokenGrant(UniversalBaseModel):
-    tts: bool = pydantic.Field()
+    tts: typing.Optional[bool] = pydantic.Field(default=None)
     """
     The `tts` grant allows the token to be used to access any TTS endpoint.
     """
 
+    stt: typing.Optional[bool] = pydantic.Field(default=None)
+    """
+    The `stt` grant allows the token to be used to access any STT endpoint.
+    """
+
     if IS_PYDANTIC_V2:
         model_config: typing.ClassVar[pydantic.ConfigDict] = pydantic.ConfigDict(extra="allow", frozen=True)  # type: ignore # Pydantic v2
     else:

cartesia/auth/types/token_request.py

@@ -1,16 +1,16 @@
 # This file was auto-generated by Fern from our API Definition.
 
 from ...core.pydantic_utilities import UniversalBaseModel
+import typing
 from .token_grant import TokenGrant
 import pydantic
-import typing
 from ...core.pydantic_utilities import IS_PYDANTIC_V2
 
 
 class TokenRequest(UniversalBaseModel):
-    grants: TokenGrant = pydantic.Field()
+    grants: typing.Optional[TokenGrant] = pydantic.Field(default=None)
     """
-    The permissions to be granted via the token.
+    The permissions to be granted via the token. Both TTS and STT grants are optional - specify only the capabilities you need.
     """
 
     expires_in: typing.Optional[int] = pydantic.Field(default=None)

cartesia/base_client.py

@@ -7,6 +7,7 @@ from .core.client_wrapper import SyncClientWrapper
 from .api_status.client import ApiStatusClient
 from .auth.client import AuthClient
 from .infill.client import InfillClient
+from .stt.socket_client import AsyncSttClientWithWebsocket, SttClientWithWebsocket
 from .tts.client import TtsClient
 from .voice_changer.client import VoiceChangerClient
 from .voices.client import VoicesClient
@@ -80,6 +81,7 @@ class BaseCartesia:
         self.api_status = ApiStatusClient(client_wrapper=self._client_wrapper)
         self.auth = AuthClient(client_wrapper=self._client_wrapper)
         self.infill = InfillClient(client_wrapper=self._client_wrapper)
+        self.stt = SttClientWithWebsocket(client_wrapper=self._client_wrapper)
         self.tts = TtsClient(client_wrapper=self._client_wrapper)
         self.voice_changer = VoiceChangerClient(client_wrapper=self._client_wrapper)
         self.voices = VoicesClient(client_wrapper=self._client_wrapper)

cartesia/client.py

@@ -8,6 +8,7 @@ import httpx
 
 from .base_client import AsyncBaseCartesia, BaseCartesia
 from .environment import CartesiaEnvironment
+from .stt.socket_client import AsyncSttClientWithWebsocket, SttClientWithWebsocket
 from .tts.socket_client import AsyncTtsClientWithWebsocket, TtsClientWithWebsocket
 
 
@@ -66,6 +67,7 @@ class Cartesia(BaseCartesia):
             follow_redirects=follow_redirects,
             httpx_client=httpx_client,
         )
+        self.stt = SttClientWithWebsocket(client_wrapper=self._client_wrapper)
         self.tts = TtsClientWithWebsocket(client_wrapper=self._client_wrapper)
 
     def __enter__(self):
@@ -143,6 +145,9 @@ class AsyncCartesia(AsyncBaseCartesia):
         self._session = None
         self._loop = None
         self.max_num_connections = max_num_connections
+        self.stt = AsyncSttClientWithWebsocket(
+            client_wrapper=self._client_wrapper, get_session=self._get_session
+        )
         self.tts = AsyncTtsClientWithWebsocket(
             client_wrapper=self._client_wrapper, get_session=self._get_session
         )

cartesia/core/client_wrapper.py

@@ -16,7 +16,7 @@ class BaseClientWrapper:
         headers: typing.Dict[str, str] = {
             "X-Fern-Language": "Python",
             "X-Fern-SDK-Name": "cartesia",
-            "X-Fern-SDK-Version": "2.0.4",
+            "X-Fern-SDK-Version": "2.0.6",
         }
         headers["X-API-Key"] = self.api_key
         headers["Cartesia-Version"] = "2024-11-13"

cartesia/stt/__init__.py

@@ -0,0 +1,57 @@
+# This file was auto-generated by Fern from our API Definition.
+
+from .types import (
+    DoneMessage,
+    ErrorMessage,
+    FlushDoneMessage,
+    StreamingTranscriptionResponse,
+    StreamingTranscriptionResponse_Done,
+    StreamingTranscriptionResponse_Error,
+    StreamingTranscriptionResponse_FlushDone,
+    StreamingTranscriptionResponse_Transcript,
+    SttEncoding,
+    TimestampGranularity,
+    TranscriptMessage,
+    TranscriptionResponse,
+    TranscriptionWord,
+)
+from .requests import (
+    DoneMessageParams,
+    ErrorMessageParams,
+    FlushDoneMessageParams,
+    StreamingTranscriptionResponseParams,
+    StreamingTranscriptionResponse_DoneParams,
+    StreamingTranscriptionResponse_ErrorParams,
+    StreamingTranscriptionResponse_FlushDoneParams,
+    StreamingTranscriptionResponse_TranscriptParams,
+    TranscriptMessageParams,
+    TranscriptionResponseParams,
+    TranscriptionWordParams,
+)
+
+__all__ = [
+    "DoneMessage",
+    "DoneMessageParams",
+    "ErrorMessage",
+    "ErrorMessageParams",
+    "FlushDoneMessage",
+    "FlushDoneMessageParams",
+    "StreamingTranscriptionResponse",
+    "StreamingTranscriptionResponseParams",
+    "StreamingTranscriptionResponse_Done",
+    "StreamingTranscriptionResponse_DoneParams",
+    "StreamingTranscriptionResponse_Error",
+    "StreamingTranscriptionResponse_ErrorParams",
+    "StreamingTranscriptionResponse_FlushDone",
+    "StreamingTranscriptionResponse_FlushDoneParams",
+    "StreamingTranscriptionResponse_Transcript",
+    "StreamingTranscriptionResponse_TranscriptParams",
+    "SttEncoding",
+    "TimestampGranularity",
+    "TranscriptMessage",
+    "TranscriptMessageParams",
+    "TranscriptionResponse",
+    "TranscriptionResponseParams",
+    "TranscriptionWord",
+    "TranscriptionWordParams",
+]

cartesia/stt/_async_websocket.py

@@ -0,0 +1,293 @@
+import asyncio
+import json
+import typing
+import uuid
+from typing import Any, Awaitable, AsyncGenerator, Callable, Dict, Optional, Union
+
+import aiohttp
+
+from cartesia.stt.types import (
+    StreamingTranscriptionResponse,
+    StreamingTranscriptionResponse_Error,
+    StreamingTranscriptionResponse_Transcript,
+)
+from cartesia.stt.types.stt_encoding import SttEncoding
+
+from ..core.pydantic_utilities import parse_obj_as
+from ._websocket import SttWebsocket
+
+
+class AsyncSttWebsocket(SttWebsocket):
+    """This class contains methods to transcribe audio using WebSocket asynchronously."""
+
+    def __init__(
+        self,
+        ws_url: str,
+        api_key: str,
+        cartesia_version: str,
+        get_session: Callable[[], Awaitable[Optional[aiohttp.ClientSession]]],
+        timeout: float = 30,
+    ):
+        """
+        Args:
+            ws_url: The WebSocket URL for the Cartesia API.
+            api_key: The API key to use for authorization.
+            cartesia_version: The version of the Cartesia API to use.
+            timeout: The timeout for responses on the WebSocket in seconds.
+            get_session: A function that returns an awaitable of aiohttp.ClientSession object.
+        """
+        super().__init__(ws_url, api_key, cartesia_version)
+        self.timeout = timeout
+        self._get_session = get_session
+        self.websocket: Optional[aiohttp.ClientWebSocketResponse] = None
+        self._default_model: str = "ink-whisper"
+        self._default_language: Optional[str] = "en"
+        self._default_encoding: SttEncoding = "pcm_s16le"
+        self._default_sample_rate: int = 16000
+        self._default_min_volume: Optional[float] = None
+        self._default_max_silence_duration_secs: Optional[float] = None
+
+    def __del__(self):
+        try:
+            loop = asyncio.get_running_loop()
+        except RuntimeError:
+            loop = None
+
+        if loop is None:
+            asyncio.run(self.close())
+        elif loop.is_running():
+            loop.create_task(self.close())
+
+    async def connect(
+        self,
+        *,
+        model: str = "ink-whisper",
+        language: Optional[str] = "en",
+        encoding: SttEncoding = "pcm_s16le",
+        sample_rate: int = 16000,
+        min_volume: Optional[float] = None,
+        max_silence_duration_secs: Optional[float] = None,
+    ):
+        """Connect to the STT WebSocket with the specified parameters.
+
+        Args:
+            model: ID of the model to use for transcription (required)
+            language: The language of the input audio in ISO-639-1 format (defaults to "en")
+            encoding: The encoding format of the audio data (required)
+            sample_rate: The sample rate of the audio in Hz (required)
+            min_volume: Volume threshold for voice activity detection (0.0-1.0)
+            max_silence_duration_secs: Maximum duration of silence before endpointing
+
+        Raises:
+            RuntimeError: If the connection to the WebSocket fails.
+        """
+        self._default_model = model
+        self._default_language = language
+        self._default_encoding = encoding
+        self._default_sample_rate = sample_rate
+        self._default_min_volume = min_volume
+        self._default_max_silence_duration_secs = max_silence_duration_secs
+        
+        if self.websocket is None or self._is_websocket_closed():
+            route = "stt/websocket"
+            session = await self._get_session()
+            
+            params = {
+                "model": model,
+                "api_key": self.api_key,
+                "cartesia_version": self.cartesia_version,
+                "encoding": encoding,
+                "sample_rate": str(sample_rate),
+            }
+            if language is not None:
+                params["language"] = language
+            if min_volume is not None:
+                params["min_volume"] = str(min_volume)
+            if max_silence_duration_secs is not None:
+                params["max_silence_duration_secs"] = str(max_silence_duration_secs)
+
+            query_string = "&".join([f"{k}={v}" for k, v in params.items()])
+            url = f"{self.ws_url}/{route}?{query_string}"
+            
+            try:
+                if session is None:
+                    raise RuntimeError("Session is not available")
+                self.websocket = await session.ws_connect(url)
+            except Exception as e:
+                status_code = None
+                error_message = str(e)
+                
+                if hasattr(e, 'status') and e.status is not None:
+                    status_code = e.status
+                
+                    if status_code == 402:
+                        error_message = "Payment required. Your API key may have insufficient credits or permissions."
+                    elif status_code == 401:
+                        error_message = "Unauthorized. Please check your API key."
+                    elif status_code == 403:
+                        error_message = "Forbidden. You don't have permission to access this resource."
+                    elif status_code == 404:
+                        error_message = "Not found. The requested resource doesn't exist."
+                    
+                    raise RuntimeError(f"Failed to connect to WebSocket.\nStatus: {status_code}. Error message: {error_message}")
+                else:
+                    raise RuntimeError(f"Failed to connect to WebSocket at {url}. {e}")
+
+    def _is_websocket_closed(self):
+        return self.websocket is None or self.websocket.closed
+
+    async def close(self):
+        """This method closes the websocket connection. Highly recommended to call this method when done."""
+        if self.websocket is not None and not self._is_websocket_closed():
+            await self.websocket.close()
+        self.websocket = None
+
+    async def send(self, data: Union[bytes, str]):
+        """Send audio data or control commands to the WebSocket.
+
+        Args:
+            data: Binary audio data or text command ("finalize" or "done")
+        """
+        if self.websocket is None or self._is_websocket_closed():
+            await self.connect(
+                model=self._default_model,
+                language=self._default_language,
+                encoding=self._default_encoding,
+                sample_rate=self._default_sample_rate,
+                min_volume=self._default_min_volume,
+                max_silence_duration_secs=self._default_max_silence_duration_secs,
+            )
+        
+        assert self.websocket is not None, "WebSocket should be connected after connect() call"
+        
+        if isinstance(data, bytes):
+            await self.websocket.send_bytes(data)
+        elif isinstance(data, str):
+            await self.websocket.send_str(data)
+        else:
+            raise TypeError("Data must be bytes (audio) or str (command)")
+
+    async def receive(self) -> AsyncGenerator[Dict[str, Any], None]:  # type: ignore[override]
+        """Receive transcription results from the WebSocket.
+
+        Yields:
+            Dictionary containing transcription results, flush_done, done, or error messages
+        """
+        if self.websocket is None or self._is_websocket_closed():
+            await self.connect(
+                model=self._default_model,
+                language=self._default_language,
+                encoding=self._default_encoding,
+                sample_rate=self._default_sample_rate,
+                min_volume=self._default_min_volume,
+                max_silence_duration_secs=self._default_max_silence_duration_secs,
+            )
+
+        assert self.websocket is not None, "WebSocket should be connected after connect() call"
+
+        try:
+            async for message in self.websocket:
+                if message.type == aiohttp.WSMsgType.TEXT:
+                    raw_data = json.loads(message.data)
+                    
+                    # Handle error responses
+                    if raw_data.get("type") == "error":
+                        raise RuntimeError(f"Error transcribing audio: {raw_data.get('message', 'Unknown error')}")
+                    
+                    # Handle transcript responses with flexible parsing
+                    if raw_data.get("type") == "transcript":
+                        # Provide defaults for missing required fields
+                        result = {
+                            "type": raw_data["type"],
+                            "request_id": raw_data.get("request_id", ""),
+                            "text": raw_data.get("text", ""),  # Default to empty string if missing
+                            "is_final": raw_data.get("is_final", False),  # Default to False if missing
+                        }
+                        
+                        # Add optional fields if present
+                        if "duration" in raw_data:
+                            result["duration"] = raw_data["duration"]
+                        if "language" in raw_data:
+                            result["language"] = raw_data["language"]
+                        if "words" in raw_data:
+                            result["words"] = raw_data["words"]
+                        
+                        yield result
+                    
+                    # Handle flush_done acknowledgment
+                    elif raw_data.get("type") == "flush_done":
+                        result = {
+                            "type": raw_data["type"],
+                            "request_id": raw_data.get("request_id", ""),
+                        }
+                        yield result
+                    
+                    # Handle done acknowledgment
+                    elif raw_data.get("type") == "done":
+                        result = {
+                            "type": raw_data["type"],
+                            "request_id": raw_data.get("request_id", ""),
+                        }
+                        yield result
+                        break  # Exit the loop when done
+                
+                elif message.type == aiohttp.WSMsgType.ERROR:
+                    error_message = f"WebSocket error: {self.websocket.exception()}"
+                    raise RuntimeError(error_message)
+                elif message.type == aiohttp.WSMsgType.CLOSE:
+                    break  # WebSocket was closed
+        except Exception as e:
+            await self.close()
+            raise e
+
+    async def transcribe(  # type: ignore[override]
+        self,
+        audio_chunks: typing.AsyncIterator[bytes],
+        *,
+        model: str = "ink-whisper",
+        language: Optional[str] = "en",
+        encoding: SttEncoding = "pcm_s16le",
+        sample_rate: int = 16000,
+        min_volume: Optional[float] = None,
+        max_silence_duration_secs: Optional[float] = None,
+    ) -> AsyncGenerator[Dict[str, Any], None]:
+        """Transcribe audio chunks using the WebSocket.
+
+        Args:
+            audio_chunks: Async iterator of audio chunks as bytes
+            model: ID of the model to use for transcription (required)
+            language: The language of the input audio in ISO-639-1 format (defaults to "en")
+            encoding: The encoding format of the audio data (required)
+            sample_rate: The sample rate of the audio in Hz (required)
+            min_volume: Volume threshold for voice activity detection (0.0-1.0)
+            max_silence_duration_secs: Maximum duration of silence before endpointing
+
+        Yields:
+            Dictionary containing transcription results, flush_done, done, or error messages
+        """
+        await self.connect(
+            model=model,
+            language=language,
+            encoding=encoding,
+            sample_rate=sample_rate,
+            min_volume=min_volume,
+            max_silence_duration_secs=max_silence_duration_secs,
+        )
+
+        try:
+            # Send all audio chunks
+            async for chunk in audio_chunks:
+                await self.send(chunk)
+            
+            # Send finalize command to flush remaining audio
+            await self.send("finalize")
+            
+            # Send done command to close session cleanly
+            await self.send("done")
+            
+            # Receive all responses until done
+            async for result in self.receive():
+                yield result
+            
+        finally:
+            await self.close()