dv-pipecat-ai 0.0.82.dev815__py3-none-any.whl → 0.0.82.dev857__py3-none-any.whl

pipecat/services/sarvam/tts.py

@@ -6,25 +6,40 @@
 
 """Sarvam AI text-to-speech service implementation."""
 
+import asyncio
 import base64
-from typing import AsyncGenerator, Optional
+import json
+from typing import Any, AsyncGenerator, Mapping, Optional
 
 import aiohttp
 from loguru import logger
 from pydantic import BaseModel, Field
 
 from pipecat.frames.frames import (
+    CancelFrame,
+    EndFrame,
     ErrorFrame,
     Frame,
+    LLMFullResponseEndFrame,
     StartFrame,
+    StartInterruptionFrame,
     TTSAudioRawFrame,
     TTSStartedFrame,
     TTSStoppedFrame,
 )
-from pipecat.services.tts_service import TTSService
+from pipecat.processors.frame_processor import FrameDirection
+from pipecat.services.tts_service import InterruptibleTTSService, TTSService
 from pipecat.transcriptions.language import Language
 from pipecat.utils.tracing.service_decorators import traced_tts
 
+try:
+    from websockets.asyncio.client import connect as websocket_connect
+    from websockets.protocol import State
+except ModuleNotFoundError as e:
+    logger.error(f"Exception: {e}")
+    logger.error("In order to use Sarvam, you need to `pip install pipecat-ai[sarvam]`.")
+    raise Exception(f"Missing module: {e}")
+
 
 def language_to_sarvam_language(language: Language) -> Optional[str]:
     """Convert Pipecat Language enum to Sarvam AI language codes.
@@ -52,7 +67,7 @@ def language_to_sarvam_language(language: Language) -> Optional[str]:
     return LANGUAGE_MAP.get(language)
 
 
-class SarvamTTSService(TTSService):
+class SarvamHttpTTSService(TTSService):
     """Text-to-Speech service using Sarvam AI's API.
 
     Converts text to speech using Sarvam AI's TTS models with support for multiple
@@ -95,9 +110,9 @@ class SarvamTTSService(TTSService):
         self,
         *,
         api_key: str,
+        aiohttp_session: aiohttp.ClientSession,
         voice_id: str = "anushka",
         model: str = "bulbul:v2",
-        aiohttp_session: aiohttp.ClientSession,
         base_url: str = "https://api.sarvam.ai",
         sample_rate: Optional[int] = None,
         params: Optional[InputParams] = None,
@@ -107,9 +122,9 @@ class SarvamTTSService(TTSService):
 
         Args:
             api_key: Sarvam AI API subscription key.
+            aiohttp_session: Shared aiohttp session for making requests.
             voice_id: Speaker voice ID (e.g., "anushka", "meera"). Defaults to "anushka".
             model: TTS model to use ("bulbul:v1" or "bulbul:v2"). Defaults to "bulbul:v2".
-            aiohttp_session: Shared aiohttp session for making requests.
             base_url: Sarvam AI API base URL. Defaults to "https://api.sarvam.ai".
             sample_rate: Audio sample rate in Hz (8000, 16000, 22050, 24000). If None, uses default.
             params: Additional voice and preprocessing parameters. If None, uses defaults.
@@ -117,16 +132,16 @@ class SarvamTTSService(TTSService):
         """
         super().__init__(sample_rate=sample_rate, **kwargs)
 
-        params = params or SarvamTTSService.InputParams()
+        params = params or SarvamHttpTTSService.InputParams()
 
         self._api_key = api_key
         self._base_url = base_url
         self._session = aiohttp_session
 
         self._settings = {
-            "language": self.language_to_service_language(params.language)
-            if params.language
-            else "en-IN",
+            "language": (
+                self.language_to_service_language(params.language) if params.language else "en-IN"
+            ),
             "pitch": params.pitch,
             "pace": params.pace,
             "loudness": params.loudness,
@@ -186,7 +201,7 @@ class SarvamTTSService(TTSService):
                 "pitch": self._settings["pitch"],
                 "pace": self._settings["pace"],
                 "loudness": self._settings["loudness"],
-                "speech_sample_rate": self.sample_rate,
+                "sample_rate": self.sample_rate,
                 "enable_preprocessing": self._settings["enable_preprocessing"],
                 "model": self._model_name,
             }
@@ -240,3 +255,393 @@ class SarvamTTSService(TTSService):
         finally:
             await self.stop_ttfb_metrics()
             yield TTSStoppedFrame()
+
+
+class SarvamTTSService(InterruptibleTTSService):
+    """WebSocket-based text-to-speech service using Sarvam AI.
+
+    Provides streaming TTS with real-time audio generation for multiple Indian languages.
+    Supports voice control parameters like pitch, pace, and loudness adjustment.
+
+    Example::
+
+        tts = SarvamTTSService(
+            api_key="your-api-key",
+            voice_id="anushka",
+            model="bulbul:v2",
+            params=SarvamTTSService.InputParams(
+                language=Language.HI,
+                pitch=0.1,
+                pace=1.2
+            )
+        )
+    """
+
+    class InputParams(BaseModel):
+        """Configuration parameters for Sarvam TTS.
+
+        Parameters:
+            pitch: Voice pitch adjustment (-0.75 to 0.75). Defaults to 0.0.
+            pace: Speech pace multiplier (0.3 to 3.0). Defaults to 1.0.
+            loudness: Volume multiplier (0.1 to 3.0). Defaults to 1.0.
+            enable_preprocessing: Enable text preprocessing. Defaults to False.
+            min_buffer_size: Minimum number of characters to buffer before generating audio.
+                Lower values reduce latency but may affect quality. Defaults to 50.
+            max_chunk_length: Maximum number of characters processed in a single chunk.
+                Controls memory usage and processing efficiency. Defaults to 200.
+            output_audio_codec: Audio codec format. Defaults to "linear16".
+            output_audio_bitrate: Audio bitrate. Defaults to "128k".
+            language: Target language for synthesis. Supports Bengali (bn-IN), English (en-IN),
+                Gujarati (gu-IN), Hindi (hi-IN), Kannada (kn-IN), Malayalam (ml-IN),
+                Marathi (mr-IN), Odia (od-IN), Punjabi (pa-IN), Tamil (ta-IN),
+                Telugu (te-IN). Defaults to en-IN.
+
+                Available Speakers:
+            Female: anushka, manisha, vidya, arya
+            Male: abhilash, karun, hitesh
+        """
+
+        pitch: Optional[float] = Field(default=0.0, ge=-0.75, le=0.75)
+        pace: Optional[float] = Field(default=1.0, ge=0.3, le=3.0)
+        loudness: Optional[float] = Field(default=1.0, ge=0.1, le=3.0)
+        enable_preprocessing: Optional[bool] = False
+        min_buffer_size: Optional[int] = 50
+        max_chunk_length: Optional[int] = 200
+        output_audio_codec: Optional[str] = "linear16"
+        output_audio_bitrate: Optional[str] = "128k"
+        language: Optional[Language] = Language.EN
+
+    def __init__(
+        self,
+        *,
+        api_key: str,
+        model: str = "bulbul:v2",
+        voice_id: str = "anushka",
+        url: str = "wss://api.sarvam.ai/text-to-speech/ws",
+        aiohttp_session: Optional[aiohttp.ClientSession] = None,
+        aggregate_sentences: Optional[bool] = True,
+        sample_rate: Optional[int] = None,
+        params: Optional[InputParams] = None,
+        **kwargs,
+    ):
+        """Initialize the Sarvam TTS service with voice and transport configuration.
+
+        Args:
+            api_key: Sarvam API key for authenticating TTS requests.
+            model: Identifier of the Sarvam speech model (default "bulbul:v2").
+            voice_id: Voice identifier for synthesis (default "anushka").
+            url: WebSocket URL for connecting to the TTS backend (default production URL).
+            aiohttp_session: Optional shared aiohttp session. To maintain backward compatibility.
+
+                .. deprecated:: 0.0.81
+                    aiohttp_session is no longer used. This parameter will be removed in a future version.
+
+            aggregate_sentences: Whether to merge multiple sentences into one audio chunk (default True).
+            sample_rate: Desired sample rate for the output audio in Hz (overrides default if set).
+            params: Optional input parameters to override global configuration.
+            **kwargs: Optional keyword arguments forwarded to InterruptibleTTSService (such as
+                `push_stop_frames`, `sample_rate`, task manager parameters, event hooks, etc.)
+                to customize transport behavior or enable metrics support.
+
+        This method sets up the internal TTS configuration mapping, constructs the WebSocket
+        URL based on the chosen model, and initializes state flags before connecting.
+        """
+        # Initialize parent class first
+        super().__init__(
+            aggregate_sentences=aggregate_sentences,
+            push_text_frames=True,
+            pause_frame_processing=True,
+            push_stop_frames=True,
+            sample_rate=sample_rate,
+            **kwargs,
+        )
+        params = params or SarvamTTSService.InputParams()
+        if aiohttp_session is not None:
+            import warnings
+
+            with warnings.catch_warnings():
+                warnings.simplefilter("always")
+                warnings.warn(
+                    "The 'aiohttp_session' parameter is deprecated and will be removed in a future version. ",
+                    DeprecationWarning,
+                    stacklevel=2,
+                )
+        # WebSocket endpoint URL
+        self._websocket_url = f"{url}?model={model}"
+        self._api_key = api_key
+        self.set_model_name(model)
+        self.set_voice(voice_id)
+        # Configuration parameters
+        self._settings = {
+            "target_language_code": (
+                self.language_to_service_language(params.language) if params.language else "en-IN"
+            ),
+            "pitch": params.pitch,
+            "pace": params.pace,
+            "speaker": voice_id,
+            "loudness": params.loudness,
+            "speech_sample_rate": 0,
+            "enable_preprocessing": params.enable_preprocessing,
+            "min_buffer_size": params.min_buffer_size,
+            "max_chunk_length": params.max_chunk_length,
+            "output_audio_codec": params.output_audio_codec,
+            "output_audio_bitrate": params.output_audio_bitrate,
+        }
+        self._started = False
+
+        self._receive_task = None
+        self._keepalive_task = None
+        self._disconnecting = False
+
+    def can_generate_metrics(self) -> bool:
+        """Check if this service can generate processing metrics.
+
+        Returns:
+            True, as Sarvam service supports metrics generation.
+        """
+        return True
+
+    def language_to_service_language(self, language: Language) -> Optional[str]:
+        """Convert a Language enum to Sarvam AI language format.
+
+        Args:
+            language: The language to convert.
+
+        Returns:
+            The Sarvam AI-specific language code, or None if not supported.
+        """
+        return language_to_sarvam_language(language)
+
+    async def start(self, frame: StartFrame):
+        """Start the Sarvam TTS service.
+
+        Args:
+            frame: The start frame containing initialization parameters.
+        """
+        await super().start(frame)
+
+        self._settings["speech_sample_rate"] = self.sample_rate
+        await self._connect()
+
+    async def stop(self, frame: EndFrame):
+        """Stop the Sarvam TTS service.
+
+        Args:
+            frame: The end frame.
+        """
+        await super().stop(frame)
+        await self._disconnect()
+
+    async def cancel(self, frame: CancelFrame):
+        """Cancel the Sarvam TTS service.
+
+        Args:
+            frame: The cancel frame.
+        """
+        await super().cancel(frame)
+        await self._disconnect()
+
+    async def flush_audio(self):
+        """Flush any pending audio synthesis by sending stop command."""
+        if self._websocket:
+            msg = {"type": "flush"}
+            await self._websocket.send(json.dumps(msg))
+
+    async def push_frame(self, frame: Frame, direction: FrameDirection = FrameDirection.DOWNSTREAM):
+        """Push a frame downstream with special handling for stop conditions.
+
+        Args:
+            frame: The frame to push.
+            direction: The direction to push the frame.
+        """
+        await super().push_frame(frame, direction)
+        if isinstance(frame, (TTSStoppedFrame, StartInterruptionFrame)):
+            self._started = False
+
+    async def process_frame(self, frame: Frame, direction: FrameDirection):
+        """Process a frame and flush audio if it's the end of a full response."""
+        if isinstance(frame, LLMFullResponseEndFrame):
+            await self.flush_audio()
+        return await super().process_frame(frame, direction)
+
+    async def _update_settings(self, settings: Mapping[str, Any]):
+        """Update service settings and reconnect if voice changed."""
+        prev_voice = self._voice_id
+        await super()._update_settings(settings)
+        if not prev_voice == self._voice_id:
+            logger.info(f"Switching TTS voice to: [{self._voice_id}]")
+            await self._send_config()
+
+    async def _connect(self):
+        """Connect to Sarvam WebSocket and start background tasks."""
+        await self._connect_websocket()
+
+        if self._websocket and not self._receive_task:
+            self._receive_task = self.create_task(self._receive_task_handler(self._report_error))
+
+        if self._websocket and not self._keepalive_task:
+            self._keepalive_task = self.create_task(
+                self._keepalive_task_handler(),
+            )
+
+    async def _disconnect(self):
+        """Disconnect from Sarvam WebSocket and clean up tasks."""
+        try:
+            # First, set a flag to prevent new operations
+            self._disconnecting = True
+
+            # Cancel background tasks BEFORE closing websocket
+            if self._receive_task:
+                await self.cancel_task(self._receive_task, timeout=2.0)
+                self._receive_task = None
+
+            if self._keepalive_task:
+                await self.cancel_task(self._keepalive_task, timeout=2.0)
+                self._keepalive_task = None
+
+            # Now close the websocket
+            await self._disconnect_websocket()
+
+        except Exception as e:
+            logger.error(f"Error during disconnect: {e}")
+        finally:
+            # Reset state only after everything is cleaned up
+            self._started = False
+            self._websocket = None
+            self._disconnecting = False
+
+    async def _connect_websocket(self):
+        """Establish WebSocket connection to Sarvam API."""
+        try:
+            if self._websocket and self._websocket.state is State.OPEN:
+                return
+
+            self._websocket = await websocket_connect(
+                self._websocket_url,
+                additional_headers={
+                    "api-subscription-key": self._api_key,
+                },
+            )
+            logger.debug("Connected to Sarvam TTS Websocket")
+            await self._send_config()
+
+        except Exception as e:
+            logger.error(f"{self} initialization error: {e}")
+            self._websocket = None
+            await self._call_event_handler("on_connection_error", f"{e}")
+
+    async def _send_config(self):
+        """Send initial configuration message."""
+        if not self._websocket:
+            raise Exception("WebSocket not connected")
+        self._settings["speaker"] = self._voice_id
+        logger.debug(f"Config being sent is {self._settings}")
+        config_message = {"type": "config", "data": self._settings}
+
+        try:
+            await self._websocket.send(json.dumps(config_message))
+            logger.debug("Configuration sent successfully")
+        except Exception as e:
+            logger.error(f"Failed to send config: {str(e)}")
+            await self.push_frame(ErrorFrame(f"Failed to send config: {str(e)}"))
+            raise
+
+    async def _disconnect_websocket(self):
+        """Close WebSocket connection and clean up state."""
+        try:
+            await self.stop_all_metrics()
+
+            if self._websocket:
+                logger.debug("Disconnecting from Sarvam")
+                await self._websocket.close()
+        except Exception as e:
+            logger.error(f"{self} error closing websocket: {e}")
+
+    def _get_websocket(self):
+        if self._websocket:
+            return self._websocket
+        raise Exception("Websocket not connected")
+
+    async def _receive_messages(self):
+        """Receive and process messages from Sarvam WebSocket."""
+        async for message in self._get_websocket():
+            if isinstance(message, str):
+                msg = json.loads(message)
+                if msg.get("type") == "audio":
+                    # Check for interruption before processing audio
+                    await self.stop_ttfb_metrics()
+                    audio = base64.b64decode(msg["data"]["audio"])
+                    frame = TTSAudioRawFrame(audio, self.sample_rate, 1)
+                    await self.push_frame(frame)
+                elif msg.get("type") == "error":
+                    error_msg = msg["data"]["message"]
+                    logger.error(f"TTS Error: {error_msg}")
+
+                    # If it's a timeout error, the connection might need to be reset
+                    if "too long" in error_msg.lower() or "timeout" in error_msg.lower():
+                        logger.warning("Connection timeout detected, service may need restart")
+
+                    await self.push_frame(ErrorFrame(f"TTS Error: {error_msg}"))
+
+    async def _keepalive_task_handler(self):
+        """Handle keepalive messages to maintain WebSocket connection."""
+        KEEPALIVE_SLEEP = 20
+        while True:
+            await asyncio.sleep(KEEPALIVE_SLEEP)
+            await self._send_keepalive()
+
+    async def _send_keepalive(self):
+        """Send keepalive message to maintain connection."""
+        if self._disconnecting:
+            return
+
+        if self._websocket and self._websocket.state == State.OPEN:
+            msg = {"type": "ping"}
+            await self._websocket.send(json.dumps(msg))
+
+    async def _send_text(self, text: str):
+        """Send text to Sarvam WebSocket for synthesis."""
+        if self._disconnecting:
+            logger.warning("Service is disconnecting, ignoring text send")
+            return
+
+        if self._websocket and self._websocket.state == State.OPEN:
+            msg = {"type": "text", "data": {"text": text}}
+            await self._websocket.send(json.dumps(msg))
+        else:
+            logger.warning("WebSocket not ready, cannot send text")
+
+    @traced_tts
+    async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
+        """Generate speech audio frames from input text using Sarvam TTS.
+
+        Sends text over WebSocket for synthesis and yields corresponding audio or status frames.
+
+        Args:
+            text: The text input to synthesize.
+
+        Yields:
+            Frame objects including TTSStartedFrame, TTSAudioRawFrame(s), or TTSStoppedFrame.
+        """
+        logger.debug(f"Generating TTS: [{text}]")
+
+        try:
+            if not self._websocket or self._websocket.state is State.CLOSED:
+                await self._connect()
+
+            try:
+                if not self._started:
+                    await self.start_ttfb_metrics()
+                    yield TTSStartedFrame()
+                    self._started = True
+                await self._send_text(text)
+                await self.start_tts_usage_metrics(text)
+            except Exception as e:
+                logger.error(f"{self} error sending message: {e}")
+                yield TTSStoppedFrame()
+                await self._disconnect()
+                await self._connect()
+                return
+            yield None
+        except Exception as e:
+            logger.error(f"{self} exception: {e}")

pipecat/services/speechmatics/stt.py

@@ -10,7 +10,6 @@ import asyncio
 import datetime
 import os
 import re
-import warnings
 from dataclasses import dataclass, field
 from enum import Enum
 from typing import Any, AsyncGenerator
@@ -581,7 +580,6 @@ class SpeechmaticsSTTService(STTService):
             logger.debug(f"{self} Connected to Speechmatics STT service")
         except Exception as e:
             logger.error(f"{self} Error connecting to Speechmatics: {e}")
-        finally:
             self._client = None
 
     async def _disconnect(self) -> None:
@@ -1108,6 +1106,8 @@ def _check_deprecated_args(kwargs: dict, params: SpeechmaticsSTTService.InputPar
 
     # Show deprecation warnings
     def _deprecation_warning(old: str, new: str | None = None):
+        import warnings
+
         with warnings.catch_warnings():
             warnings.simplefilter("always")
             if new:

pipecat/services/tavus/video.py

@@ -34,7 +34,7 @@ from pipecat.frames.frames import (
 )
 from pipecat.processors.frame_processor import FrameDirection, FrameProcessorSetup
 from pipecat.services.ai_service import AIService
-from pipecat.transports.services.tavus import TavusCallbacks, TavusParams, TavusTransportClient
+from pipecat.transports.tavus.transport import TavusCallbacks, TavusParams, TavusTransportClient
 
 
 class TavusVideoService(AIService):

pipecat/services/tts_service.py

@@ -122,6 +122,9 @@ class TTSService(AIService):
         self._voice = None
         self._voice_clone_params = None
 
+        self._tracing_enabled: bool = False
+
+
         if text_filter:
             import warnings
 
@@ -283,11 +286,13 @@ class TTSService(AIService):
         """
         import warnings
 
-        warnings.warn(
-            "`TTSService.say()` is deprecated. Push a `TTSSpeakFrame` instead.",
-            DeprecationWarning,
-            stacklevel=2,
-        )
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "`TTSService.say()` is deprecated. Push a `TTSSpeakFrame` instead.",
+                DeprecationWarning,
+                stacklevel=2,
+            )
 
         await self.queue_frame(TTSSpeakFrame(text))
 
@@ -304,6 +309,11 @@ class TTSService(AIService):
         await super().process_frame(frame, direction)
 
         if (
+            isinstance(frame, (TextFrame, LLMFullResponseStartFrame, LLMFullResponseEndFrame))
+            and frame.skip_tts
+        ):
+            await self.push_frame(frame, direction)
+        elif (
             isinstance(frame, TextFrame)
             and not isinstance(frame, InterimTranscriptionFrame)
             and not isinstance(frame, TranscriptionFrame)

pipecat/services/vistaar/llm.py

@@ -13,8 +13,6 @@ from loguru import logger
 from pydantic import BaseModel, Field
 
 from pipecat.frames.frames import (
-    CancelFrame,
-    EndFrame,
     Frame,
     LLMFullResponseEndFrame,
     LLMFullResponseStartFrame,
@@ -22,7 +20,6 @@ from pipecat.frames.frames import (
     LLMTextFrame,
     LLMUpdateSettingsFrame,
     StartInterruptionFrame,
-    StopInterruptionFrame,
 )
 from pipecat.processors.aggregators.llm_response import (
     LLMAssistantAggregatorParams,
@@ -32,13 +29,13 @@ from pipecat.processors.aggregators.openai_llm_context import (
     OpenAILLMContext,
     OpenAILLMContextFrame,
 )
+from pipecat.processors.frame_processor import FrameDirection
+from pipecat.services.llm_service import LLMService
 from pipecat.services.openai.llm import (
     OpenAIAssistantContextAggregator,
     OpenAIContextAggregatorPair,
     OpenAIUserContextAggregator,
 )
-from pipecat.processors.frame_processor import FrameDirection
-from pipecat.services.llm_service import LLMService
 
 
 class VistaarLLMService(LLMService):