774 lines
29 KiB
Python
774 lines
29 KiB
Python
#
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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 base64
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import json
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import uuid
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from typing import Any, AsyncGenerator, Dict, List, Literal, Mapping, Optional, Tuple, Union
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import aiohttp
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from loguru import logger
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from pydantic import BaseModel
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from pipecat.frames.frames import (
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CancelFrame,
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EndFrame,
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ErrorFrame,
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Frame,
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LLMFullResponseEndFrame,
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StartFrame,
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StartInterruptionFrame,
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TTSAudioRawFrame,
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TTSStartedFrame,
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TTSStoppedFrame,
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)
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from pipecat.processors.frame_processor import FrameDirection
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from pipecat.services.tts_service import (
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AudioContextWordTTSService,
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WordTTSService,
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)
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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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except ModuleNotFoundError as e:
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logger.error(f"Exception: {e}")
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logger.error("In order to use ElevenLabs, you need to `pip install pipecat-ai[elevenlabs]`.")
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raise Exception(f"Missing module: {e}")
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ElevenLabsOutputFormat = Literal["pcm_16000", "pcm_22050", "pcm_24000", "pcm_44100"]
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# Models that support language codes
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# The following models are excluded as they don't support language codes:
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# - eleven_flash_v2
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# - eleven_turbo_v2
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# - eleven_multilingual_v2
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ELEVENLABS_MULTILINGUAL_MODELS = {
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"eleven_flash_v2_5",
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"eleven_turbo_v2_5",
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}
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def language_to_elevenlabs_language(language: Language) -> Optional[str]:
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BASE_LANGUAGES = {
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Language.AR: "ar",
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Language.BG: "bg",
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Language.CS: "cs",
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Language.DA: "da",
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Language.DE: "de",
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Language.EL: "el",
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Language.EN: "en",
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Language.ES: "es",
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Language.FI: "fi",
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Language.FIL: "fil",
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Language.FR: "fr",
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Language.HI: "hi",
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Language.HR: "hr",
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Language.HU: "hu",
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Language.ID: "id",
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Language.IT: "it",
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Language.JA: "ja",
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Language.KO: "ko",
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Language.MS: "ms",
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Language.NL: "nl",
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Language.NO: "no",
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Language.PL: "pl",
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Language.PT: "pt",
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Language.RO: "ro",
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Language.RU: "ru",
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Language.SK: "sk",
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Language.SV: "sv",
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Language.TA: "ta",
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Language.TR: "tr",
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Language.UK: "uk",
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Language.VI: "vi",
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Language.ZH: "zh",
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}
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result = BASE_LANGUAGES.get(language)
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# If not found in base languages, try to find the base language from a variant
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if not result:
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# Convert enum value to string and get the base language part (e.g. es-ES -> es)
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lang_str = str(language.value)
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base_code = lang_str.split("-")[0].lower()
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# Look up the base code in our supported languages
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result = base_code if base_code in BASE_LANGUAGES.values() else None
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return result
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def output_format_from_sample_rate(sample_rate: int) -> str:
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match sample_rate:
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case 8000:
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return "pcm_8000"
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case 16000:
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return "pcm_16000"
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case 22050:
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return "pcm_22050"
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case 24000:
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return "pcm_24000"
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case 44100:
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return "pcm_44100"
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logger.warning(
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f"ElevenLabsTTSService: No output format available for {sample_rate} sample rate"
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)
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return "pcm_24000"
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def build_elevenlabs_voice_settings(
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settings: Dict[str, Any],
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) -> Optional[Dict[str, Union[float, bool]]]:
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"""Build voice settings dictionary for ElevenLabs based on provided settings.
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Args:
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settings: Dictionary containing voice settings parameters
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Returns:
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Dictionary of voice settings or None if no valid settings are provided
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"""
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voice_setting_keys = ["stability", "similarity_boost", "style", "use_speaker_boost", "speed"]
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voice_settings = {}
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for key in voice_setting_keys:
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if key in settings and settings[key] is not None:
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voice_settings[key] = settings[key]
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return voice_settings or None
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def calculate_word_times(
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alignment_info: Mapping[str, Any], cumulative_time: float
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) -> List[Tuple[str, float]]:
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zipped_times = list(zip(alignment_info["chars"], alignment_info["charStartTimesMs"]))
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words = "".join(alignment_info["chars"]).split(" ")
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# Calculate start time for each word. We do this by finding a space character
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# and using the previous word time, also taking into account there might not
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# be a space at the end.
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times = []
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for i, (a, b) in enumerate(zipped_times):
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if a == " " or i == len(zipped_times) - 1:
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t = cumulative_time + (zipped_times[i - 1][1] / 1000.0)
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times.append(t)
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word_times = list(zip(words, times))
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return word_times
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class ElevenLabsTTSService(AudioContextWordTTSService):
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class InputParams(BaseModel):
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language: Optional[Language] = 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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speed: Optional[float] = None
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auto_mode: Optional[bool] = True
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enable_ssml_parsing: Optional[bool] = None
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enable_logging: 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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url: str = "wss://api.elevenlabs.io",
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sample_rate: Optional[int] = None,
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params: InputParams = InputParams(),
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**kwargs,
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):
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# Aggregating sentences still gives cleaner-sounding results and fewer
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# artifacts than streaming one word at a time. On average, waiting for a
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# full sentence should only "cost" us 15ms or so with GPT-4o or a Llama
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# 3 model, and it's worth it for the better audio quality.
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#
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# We also don't want to automatically push LLM response text frames,
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# because the context aggregators will add them to the LLM context even
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# if we're interrupted. ElevenLabs gives us word-by-word timestamps. We
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# can use those to generate text frames ourselves aligned with the
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# playout timing of the audio!
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#
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# Finally, ElevenLabs doesn't provide information on when the bot stops
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# speaking for a while, so we want the parent class to send TTSStopFrame
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# after a short period not receiving any audio.
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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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pause_frame_processing=True,
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sample_rate=sample_rate,
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**kwargs,
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)
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self._api_key = api_key
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self._url = url
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self._settings = {
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"language": self.language_to_service_language(params.language)
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if params.language
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else None,
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"stability": params.stability,
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"similarity_boost": params.similarity_boost,
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"style": params.style,
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"use_speaker_boost": params.use_speaker_boost,
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"speed": params.speed,
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"auto_mode": str(params.auto_mode).lower(),
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"enable_ssml_parsing": params.enable_ssml_parsing,
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"enable_logging": params.enable_logging,
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}
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self.set_model_name(model)
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self.set_voice(voice_id)
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self._output_format = "" # initialized in start()
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self._voice_settings = self._set_voice_settings()
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# Indicates if we have sent TTSStartedFrame. It will reset to False when
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# there's an interruption or TTSStoppedFrame.
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self._started = False
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self._cumulative_time = 0
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# Context management for v1 multi API
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self._context_id = None
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self._receive_task = None
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self._keepalive_task = None
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def can_generate_metrics(self) -> bool:
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return True
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def language_to_service_language(self, language: Language) -> Optional[str]:
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return language_to_elevenlabs_language(language)
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def _set_voice_settings(self):
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return build_elevenlabs_voice_settings(self._settings)
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async def set_model(self, model: str):
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await super().set_model(model)
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logger.info(f"Switching TTS model to: [{model}]")
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# No need to disconnect/reconnect for model changes with multi-context API
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async def _update_settings(self, settings: Mapping[str, Any]):
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prev_voice = self._voice_id
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await super()._update_settings(settings)
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# If voice changes, we don't need to reconnect, just use a new context
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if not prev_voice == self._voice_id:
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logger.info(f"Switching TTS voice to: [{self._voice_id}]")
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async def start(self, frame: StartFrame):
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await super().start(frame)
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self._output_format = output_format_from_sample_rate(self.sample_rate)
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await self._connect()
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async def stop(self, frame: EndFrame):
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await super().stop(frame)
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await self._disconnect()
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async def cancel(self, frame: CancelFrame):
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await super().cancel(frame)
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await self._disconnect()
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async def flush_audio(self):
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if self._websocket and self._context_id:
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msg = {"context_id": self._context_id, "flush": True}
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await self._websocket.send(json.dumps(msg))
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async def push_frame(self, frame: Frame, direction: FrameDirection = FrameDirection.DOWNSTREAM):
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await super().push_frame(frame, direction)
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if isinstance(frame, (TTSStoppedFrame, StartInterruptionFrame)):
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self._started = False
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if isinstance(frame, TTSStoppedFrame):
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await self.add_word_timestamps([("Reset", 0)])
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async def _connect(self):
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await self._connect_websocket()
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if self._websocket and not self._receive_task:
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self._receive_task = self.create_task(self._receive_task_handler(self._report_error))
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if self._websocket and not self._keepalive_task:
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self._keepalive_task = self.create_task(self._keepalive_task_handler())
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async def _disconnect(self):
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if self._receive_task:
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await self.cancel_task(self._receive_task)
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self._receive_task = None
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if self._keepalive_task:
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await self.cancel_task(self._keepalive_task)
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self._keepalive_task = None
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await self._disconnect_websocket()
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async def _connect_websocket(self):
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try:
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if self._websocket and self._websocket.open:
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return
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logger.debug("Connecting to ElevenLabs")
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voice_id = self._voice_id
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model = self.model_name
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output_format = self._output_format
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url = f"{self._url}/v1/text-to-speech/{voice_id}/multi-stream-input?model_id={model}&output_format={output_format}&auto_mode={self._settings['auto_mode']}"
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if self._settings["enable_ssml_parsing"]:
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url += f"&enable_ssml_parsing={self._settings['enable_ssml_parsing']}"
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if self._settings["enable_logging"]:
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url += f"&enable_logging={self._settings['enable_logging']}"
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# Language can only be used with the ELEVENLABS_MULTILINGUAL_MODELS
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language = self._settings["language"]
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if model in ELEVENLABS_MULTILINGUAL_MODELS and language is not None:
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url += f"&language_code={language}"
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logger.debug(f"Using language code: {language}")
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elif language is not None:
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logger.warning(
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f"Language code [{language}] not applied. Language codes can only be used with multilingual models: {', '.join(sorted(ELEVENLABS_MULTILINGUAL_MODELS))}"
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)
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# Set max websocket message size to 16MB for large audio responses
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self._websocket = await websockets.connect(url, max_size=16 * 1024 * 1024)
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except Exception as e:
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logger.error(f"{self} initialization error: {e}")
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self._websocket = None
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await self._call_event_handler("on_connection_error", f"{e}")
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async def _disconnect_websocket(self):
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try:
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await self.stop_all_metrics()
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if self._websocket:
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logger.debug("Disconnecting from ElevenLabs")
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# Close all contexts and the socket
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if self._context_id:
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await self._websocket.send(json.dumps({"close_socket": True}))
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await self._websocket.close()
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except Exception as e:
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logger.error(f"{self} error closing websocket: {e}")
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finally:
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self._started = False
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self._context_id = None
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self._websocket = None
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def _get_websocket(self):
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if self._websocket:
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return self._websocket
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raise Exception("Websocket not connected")
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async def _handle_interruption(self, frame: StartInterruptionFrame, direction: FrameDirection):
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await super()._handle_interruption(frame, direction)
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# Close the current context when interrupted without closing the websocket
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if self._context_id and self._websocket:
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logger.trace(f"Closing context {self._context_id} due to interruption")
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try:
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await self._websocket.send(
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json.dumps({"context_id": self._context_id, "close_context": True})
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)
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except Exception as e:
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logger.error(f"Error closing context on interruption: {e}")
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self._context_id = None
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self._started = False
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async def _receive_messages(self):
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async for message in self._get_websocket():
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msg = json.loads(message)
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# Check if this message belongs to the current context
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# The default context may return null/None for context_id
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received_ctx_id = msg.get("context_id")
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if (
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self._context_id is not None
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and received_ctx_id is not None
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and received_ctx_id != self._context_id
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):
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logger.trace(f"Ignoring message from different context: {received_ctx_id}")
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continue
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if msg.get("audio"):
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await self.stop_ttfb_metrics()
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self.start_word_timestamps()
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audio = base64.b64decode(msg["audio"])
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frame = TTSAudioRawFrame(audio, self.sample_rate, 1)
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await self.push_frame(frame)
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if msg.get("alignment"):
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word_times = calculate_word_times(msg["alignment"], self._cumulative_time)
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await self.add_word_timestamps(word_times)
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self._cumulative_time = word_times[-1][1]
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if msg.get("is_final"):
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logger.trace(f"Received final message for context {received_ctx_id}")
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# Context has finished
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if self._context_id == received_ctx_id:
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self._context_id = None
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self._started = False
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async def _keepalive_task_handler(self):
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while True:
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await asyncio.sleep(10)
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try:
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# Send an empty message to keep the connection alive
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if self._websocket and self._websocket.open:
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await self._websocket.send(json.dumps({}))
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except websockets.ConnectionClosed as e:
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logger.warning(f"{self} keepalive error: {e}")
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break
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async def _send_text(self, text: str):
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if self._websocket:
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if not self._context_id:
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# First message for a new context - need a space to initialize
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msg = {"text": " ", "context_id": str(uuid.uuid4()), "xi_api_key": self._api_key}
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# Add voice settings only in first message for a context
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if self._voice_settings:
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msg["voice_settings"] = self._voice_settings
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await self._websocket.send(json.dumps(msg))
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self._context_id = msg["context_id"]
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logger.trace(f"Created new context {self._context_id}")
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# Now send the actual text content
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msg = {"text": text, "context_id": self._context_id}
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await self._websocket.send(json.dumps(msg))
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else:
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# Continuing with an existing context
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msg = {"text": text, "context_id": self._context_id}
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await self._websocket.send(json.dumps(msg))
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async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
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logger.debug(f"{self}: Generating TTS [{text}]")
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try:
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if not self._websocket or self._websocket.closed:
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await self._connect()
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try:
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# Close previous context if there was one
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if self._context_id and not self._started:
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await self._websocket.send(
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json.dumps({"context_id": self._context_id, "close_context": True})
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)
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self._context_id = None
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if not self._started:
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await self.start_ttfb_metrics()
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yield TTSStartedFrame()
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self._started = True
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self._cumulative_time = 0
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await self._send_text(text)
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await self.start_tts_usage_metrics(text)
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except Exception as e:
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logger.error(f"{self} error sending message: {e}")
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yield TTSStoppedFrame()
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self._started = False
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self._context_id = None
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return
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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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"""ElevenLabs Text-to-Speech service using HTTP streaming with word timestamps.
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Args:
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api_key: ElevenLabs API key
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voice_id: ID of the voice to use
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aiohttp_session: aiohttp ClientSession
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model: Model ID (default: "eleven_flash_v2_5" for low latency)
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base_url: API base URL
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sample_rate: Output sample rate
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params: Additional parameters for voice configuration
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"""
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class InputParams(BaseModel):
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language: Optional[Language] = None
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optimize_streaming_latency: Optional[int] = 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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speed: Optional[float] = None
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||
def __init__(
|
||
self,
|
||
*,
|
||
api_key: str,
|
||
voice_id: str,
|
||
aiohttp_session: aiohttp.ClientSession,
|
||
model: str = "eleven_flash_v2_5",
|
||
base_url: str = "https://api.elevenlabs.io",
|
||
sample_rate: Optional[int] = None,
|
||
params: InputParams = InputParams(),
|
||
**kwargs,
|
||
):
|
||
super().__init__(
|
||
aggregate_sentences=True,
|
||
push_text_frames=False,
|
||
push_stop_frames=True,
|
||
sample_rate=sample_rate,
|
||
**kwargs,
|
||
)
|
||
|
||
self._api_key = api_key
|
||
self._base_url = base_url
|
||
self._params = params
|
||
self._session = aiohttp_session
|
||
|
||
self._settings = {
|
||
"language": self.language_to_service_language(params.language)
|
||
if params.language
|
||
else None,
|
||
"optimize_streaming_latency": params.optimize_streaming_latency,
|
||
"stability": params.stability,
|
||
"similarity_boost": params.similarity_boost,
|
||
"style": params.style,
|
||
"use_speaker_boost": params.use_speaker_boost,
|
||
"speed": params.speed,
|
||
}
|
||
self.set_model_name(model)
|
||
self.set_voice(voice_id)
|
||
self._output_format = "" # initialized in start()
|
||
self._voice_settings = self._set_voice_settings()
|
||
|
||
# Track cumulative time to properly sequence word timestamps across utterances
|
||
self._cumulative_time = 0
|
||
self._started = False
|
||
|
||
# Store previous text for context within a turn
|
||
self._previous_text = ""
|
||
|
||
def language_to_service_language(self, language: Language) -> Optional[str]:
|
||
"""Convert pipecat Language to ElevenLabs language code."""
|
||
return language_to_elevenlabs_language(language)
|
||
|
||
def can_generate_metrics(self) -> bool:
|
||
"""Indicate that this service can generate usage metrics."""
|
||
return True
|
||
|
||
def _set_voice_settings(self):
|
||
return build_elevenlabs_voice_settings(self._settings)
|
||
|
||
def _reset_state(self):
|
||
"""Reset internal state variables."""
|
||
self._cumulative_time = 0
|
||
self._started = False
|
||
self._previous_text = ""
|
||
logger.debug(f"{self}: Reset internal state")
|
||
|
||
async def start(self, frame: StartFrame):
|
||
"""Initialize the service upon receiving a StartFrame."""
|
||
await super().start(frame)
|
||
self._output_format = output_format_from_sample_rate(self.sample_rate)
|
||
self._reset_state()
|
||
|
||
async def push_frame(self, frame: Frame, direction: FrameDirection = FrameDirection.DOWNSTREAM):
|
||
await super().push_frame(frame, direction)
|
||
if isinstance(frame, (StartInterruptionFrame, TTSStoppedFrame)):
|
||
# Reset timing on interruption or stop
|
||
self._reset_state()
|
||
|
||
if isinstance(frame, TTSStoppedFrame):
|
||
await self.add_word_timestamps([("Reset", 0)])
|
||
|
||
elif isinstance(frame, LLMFullResponseEndFrame):
|
||
# End of turn - reset previous text
|
||
self._previous_text = ""
|
||
|
||
def calculate_word_times(self, alignment_info: Mapping[str, Any]) -> List[Tuple[str, float]]:
|
||
"""Calculate word timing from character alignment data.
|
||
|
||
Example input data:
|
||
{
|
||
"characters": [" ", "H", "e", "l", "l", "o", " ", "w", "o", "r", "l", "d"],
|
||
"character_start_times_seconds": [0.0, 0.1, 0.15, 0.2, 0.25, 0.3, 0.4, 0.5, 0.6, 0.7, 0.8, 0.9],
|
||
"character_end_times_seconds": [0.1, 0.15, 0.2, 0.25, 0.3, 0.4, 0.5, 0.6, 0.7, 0.8, 0.9, 1.0]
|
||
}
|
||
|
||
Would produce word times (with cumulative_time=0):
|
||
[("Hello", 0.1), ("world", 0.5)]
|
||
|
||
Args:
|
||
alignment_info: Character timing data from ElevenLabs
|
||
|
||
Returns:
|
||
List of (word, timestamp) pairs
|
||
"""
|
||
chars = alignment_info.get("characters", [])
|
||
char_start_times = alignment_info.get("character_start_times_seconds", [])
|
||
|
||
if not chars or not char_start_times or len(chars) != len(char_start_times):
|
||
logger.warning(
|
||
f"Invalid alignment data: chars={len(chars)}, times={len(char_start_times)}"
|
||
)
|
||
return []
|
||
|
||
# Build the words and find their start times
|
||
words = []
|
||
word_start_times = []
|
||
current_word = ""
|
||
first_char_idx = -1
|
||
|
||
for i, char in enumerate(chars):
|
||
if char == " ":
|
||
if current_word: # Only add non-empty words
|
||
words.append(current_word)
|
||
# Use time of the first character of the word, offset by cumulative time
|
||
word_start_times.append(
|
||
self._cumulative_time + char_start_times[first_char_idx]
|
||
)
|
||
current_word = ""
|
||
first_char_idx = -1
|
||
else:
|
||
if not current_word: # This is the first character of a new word
|
||
first_char_idx = i
|
||
current_word += char
|
||
|
||
# Don't forget the last word if there's no trailing space
|
||
if current_word and first_char_idx >= 0:
|
||
words.append(current_word)
|
||
word_start_times.append(self._cumulative_time + char_start_times[first_char_idx])
|
||
|
||
# Create word-time pairs
|
||
word_times = list(zip(words, word_start_times))
|
||
|
||
return word_times
|
||
|
||
async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
|
||
"""Generate speech from text using ElevenLabs streaming API with timestamps.
|
||
|
||
Makes a request to the ElevenLabs API to generate audio and timing data.
|
||
Tracks the duration of each utterance to ensure correct sequencing.
|
||
Includes previous text as context for better prosody continuity.
|
||
|
||
Args:
|
||
text: Text to convert to speech
|
||
|
||
Yields:
|
||
Audio and control frames
|
||
"""
|
||
logger.debug(f"{self}: Generating TTS [{text}]")
|
||
|
||
# Use the with-timestamps endpoint
|
||
url = f"{self._base_url}/v1/text-to-speech/{self._voice_id}/stream/with-timestamps"
|
||
|
||
payload: Dict[str, Union[str, Dict[str, Union[float, bool]]]] = {
|
||
"text": text,
|
||
"model_id": self._model_name,
|
||
}
|
||
|
||
# Include previous text as context if available
|
||
if self._previous_text:
|
||
payload["previous_text"] = self._previous_text
|
||
|
||
if self._voice_settings:
|
||
payload["voice_settings"] = self._voice_settings
|
||
|
||
language = self._settings["language"]
|
||
if self._model_name in ELEVENLABS_MULTILINGUAL_MODELS and language:
|
||
payload["language_code"] = language
|
||
logger.debug(f"Using language code: {language}")
|
||
elif language:
|
||
logger.warning(
|
||
f"Language code [{language}] not applied. Language codes can only be used with multilingual models: {', '.join(sorted(ELEVENLABS_MULTILINGUAL_MODELS))}"
|
||
)
|
||
|
||
headers = {
|
||
"xi-api-key": self._api_key,
|
||
"Content-Type": "application/json",
|
||
}
|
||
|
||
# Build query parameters
|
||
params = {
|
||
"output_format": self._output_format,
|
||
}
|
||
if self._settings["optimize_streaming_latency"] is not None:
|
||
params["optimize_streaming_latency"] = self._settings["optimize_streaming_latency"]
|
||
|
||
try:
|
||
await self.start_ttfb_metrics()
|
||
|
||
async with self._session.post(
|
||
url, json=payload, headers=headers, params=params
|
||
) as response:
|
||
if response.status != 200:
|
||
error_text = await response.text()
|
||
logger.error(f"{self} error: {error_text}")
|
||
yield ErrorFrame(error=f"ElevenLabs API error: {error_text}")
|
||
return
|
||
|
||
await self.start_tts_usage_metrics(text)
|
||
|
||
# Start TTS sequence if not already started
|
||
if not self._started:
|
||
self.start_word_timestamps()
|
||
yield TTSStartedFrame()
|
||
self._started = True
|
||
|
||
# Track the duration of this utterance based on the last character's end time
|
||
utterance_duration = 0
|
||
async for line in response.content:
|
||
line_str = line.decode("utf-8").strip()
|
||
if not line_str:
|
||
continue
|
||
|
||
try:
|
||
# Parse the JSON object
|
||
data = json.loads(line_str)
|
||
|
||
# Process audio if present
|
||
if data and "audio_base64" in data:
|
||
await self.stop_ttfb_metrics()
|
||
audio = base64.b64decode(data["audio_base64"])
|
||
yield TTSAudioRawFrame(audio, self.sample_rate, 1)
|
||
|
||
# Process alignment if present
|
||
if data and "alignment" in data:
|
||
alignment = data["alignment"]
|
||
if alignment: # Ensure alignment is not None
|
||
# Get end time of the last character in this chunk
|
||
char_end_times = alignment.get("character_end_times_seconds", [])
|
||
if char_end_times:
|
||
chunk_end_time = char_end_times[-1]
|
||
# Update to the longest end time seen so far
|
||
utterance_duration = max(utterance_duration, chunk_end_time)
|
||
|
||
# Calculate word timestamps
|
||
word_times = self.calculate_word_times(alignment)
|
||
if word_times:
|
||
await self.add_word_timestamps(word_times)
|
||
except json.JSONDecodeError as e:
|
||
logger.warning(f"Failed to parse JSON from stream: {e}")
|
||
continue
|
||
except Exception as e:
|
||
logger.error(f"Error processing response: {e}", exc_info=True)
|
||
continue
|
||
|
||
# After processing all chunks, add the total utterance duration
|
||
# to the cumulative time to ensure next utterance starts after this one
|
||
if utterance_duration > 0:
|
||
self._cumulative_time += utterance_duration
|
||
|
||
# Append the current text to previous_text for context continuity
|
||
# Only add a space if there's already text
|
||
if self._previous_text:
|
||
self._previous_text += " " + text
|
||
else:
|
||
self._previous_text = text
|
||
|
||
except Exception as e:
|
||
logger.error(f"Error in run_tts: {e}")
|
||
yield ErrorFrame(error=str(e))
|
||
finally:
|
||
await self.stop_ttfb_metrics()
|
||
# Let the parent class handle TTSStoppedFrame
|