Add ElevenLabsHttpTTSService
This commit is contained in:
103
examples/foundational/07d-interruptible-elevenlabs-http.py
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103
examples/foundational/07d-interruptible-elevenlabs-http.py
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#
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# Copyright (c) 2024–2025, Daily
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#
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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import asyncio
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import os
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import sys
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import aiohttp
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from dotenv import load_dotenv
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from loguru import logger
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from runner import configure
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from pipecat.audio.vad.silero import SileroVADAnalyzer
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from pipecat.frames.frames import EndFrame
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from pipecat.pipeline.pipeline import Pipeline
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from pipecat.pipeline.runner import PipelineRunner
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from pipecat.pipeline.task import PipelineParams, PipelineTask
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from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
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from pipecat.services.elevenlabs import ElevenLabsHttpTTSService
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from pipecat.services.openai import OpenAILLMService
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from pipecat.transports.services.daily import DailyParams, DailyTransport
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load_dotenv(override=True)
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logger.remove(0)
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logger.add(sys.stderr, level="DEBUG")
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async def main():
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async with aiohttp.ClientSession() as session:
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(room_url, token) = await configure(session)
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transport = DailyTransport(
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room_url,
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token,
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"Respond bot",
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DailyParams(
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audio_out_enabled=True,
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transcription_enabled=True,
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vad_enabled=True,
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vad_analyzer=SileroVADAnalyzer(),
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),
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)
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tts = ElevenLabsHttpTTSService(
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api_key=os.getenv("ELEVENLABS_API_KEY", ""),
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voice_id=os.getenv("ELEVENLABS_VOICE_ID", ""),
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)
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llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
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messages = [
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{
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"role": "system",
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"content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
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},
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]
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context = OpenAILLMContext(messages)
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context_aggregator = llm.create_context_aggregator(context)
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pipeline = Pipeline(
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[
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transport.input(), # Transport user input
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context_aggregator.user(), # User responses
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llm, # LLM
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tts, # TTS
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transport.output(), # Transport bot output
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context_aggregator.assistant(), # Assistant spoken responses
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]
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)
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task = PipelineTask(
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pipeline,
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PipelineParams(
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allow_interruptions=True,
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enable_metrics=True,
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enable_usage_metrics=True,
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report_only_initial_ttfb=True,
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),
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)
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@transport.event_handler("on_first_participant_joined")
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async def on_first_participant_joined(transport, participant):
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await transport.capture_participant_transcription(participant["id"])
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# Kick off the conversation.
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messages.append({"role": "system", "content": "Please introduce yourself to the user."})
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await task.queue_frames([context_aggregator.user().get_context_frame()])
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@transport.event_handler("on_participant_left")
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async def on_participant_left(transport, participant, reason):
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await task.queue_frame(EndFrame())
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runner = PipelineRunner()
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await runner.run(task)
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if __name__ == "__main__":
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asyncio.run(main())
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@@ -6,9 +6,11 @@
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import asyncio
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import base64
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import io
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import json
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from typing import Any, AsyncGenerator, Dict, List, Literal, Mapping, Optional, Tuple
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import aiohttp
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from loguru import logger
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from pydantic import BaseModel, model_validator
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@@ -33,6 +35,8 @@ from pipecat.transcriptions.language import Language
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# See .env.example for ElevenLabs configuration needed
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try:
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import websockets
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from elevenlabs import Voice, VoiceSettings
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from elevenlabs.client import ElevenLabs
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except ModuleNotFoundError as e:
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logger.error(f"Exception: {e}")
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logger.error(
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@@ -418,3 +422,114 @@ class ElevenLabsTTSService(WordTTSService, WebsocketService):
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yield None
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except Exception as e:
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logger.error(f"{self} exception: {e}")
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class ElevenLabsHttpTTSService(WordTTSService):
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class InputParams(BaseModel):
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language: Optional[Language] = Language.EN
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optimize_streaming_latency: Optional[str] = None
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stability: Optional[float] = None
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similarity_boost: Optional[float] = None
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style: Optional[float] = None
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use_speaker_boost: Optional[bool] = None
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def __init__(
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self,
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*,
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api_key: str,
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voice_id: str,
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model: str = "eleven_flash_v2_5",
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output_format: ElevenLabsOutputFormat = "pcm_24000",
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params: InputParams = InputParams(),
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**kwargs,
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):
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sample_rate = self._sample_rate_from_output_format(output_format)
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super().__init__(
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aggregate_sentences=True,
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push_text_frames=False,
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push_stop_frames=True,
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stop_frame_timeout_s=2.0,
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sample_rate=sample_rate,
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**kwargs,
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)
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self._client = ElevenLabs(api_key=api_key)
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self._voice_id = voice_id
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self._model = model
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self._output_format = output_format
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# Create voice settings if provided
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self._voice_settings = None
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if params.stability is not None and params.similarity_boost is not None:
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self._voice_settings = VoiceSettings(
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stability=params.stability,
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similarity_boost=params.similarity_boost,
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style=params.style or 0.0,
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use_speaker_boost=params.use_speaker_boost or False,
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)
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logger.debug(f"Initialized with sample rate: {sample_rate}")
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@staticmethod
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def _sample_rate_from_output_format(output_format: str) -> int:
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return {
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"pcm_16000": 16000,
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"pcm_22050": 22050,
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"pcm_24000": 24000,
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"pcm_44100": 44100,
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}[output_format]
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async def start(self, frame: StartFrame):
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await super().start(frame)
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async def stop(self, frame: EndFrame):
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await super().stop(frame)
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async def cancel(self, frame: CancelFrame):
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await super().cancel(frame)
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async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
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def read_audio_stream(**kwargs):
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# Run the streaming in a separate thread
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audio_chunks = []
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stream = self._client.text_to_speech.convert_as_stream(**kwargs)
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for chunk in stream:
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if chunk:
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audio_chunks.append(chunk)
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return b"".join(audio_chunks)
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try:
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yield TTSStartedFrame()
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# Prepare parameters
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params = {
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"text": text,
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"voice_id": self._voice_id,
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"model_id": self._model,
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"output_format": self._output_format,
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"voice_settings": self._voice_settings,
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"optimize_streaming_latency": 4, # Maximum optimization + disabled text normalizer
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}
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# Get audio data in a separate thread
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audio_data = await asyncio.to_thread(read_audio_stream, **params)
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if not audio_data:
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logger.error(f"{self} No audio data returned")
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yield None
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return
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# Stream the audio data in chunks
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chunk_size = 4096 # Adjust this value as needed
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for i in range(0, len(audio_data), chunk_size):
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chunk = audio_data[i : i + chunk_size]
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if len(chunk) > 0:
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yield TTSAudioRawFrame(
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chunk, self._sample_rate_from_output_format(self._output_format), 1
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)
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yield TTSStoppedFrame()
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except Exception as e:
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logger.error(f"Error in run_tts: {e}")
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yield TTSStoppedFrame()
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