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重新划分目录 (#214)
* 🦄 refactor(web): 修改zhikongtaiweb到web * 🦄 refactor: 重写前端 路由守护尚未写完 * 🦄 refactor: 标准化路由 * update:前端重写,保留后端 * update:添加前端代码 * update:pip转成poetry启动 * update:增加mem0ai包依赖 * update:文档增加mem0ai的描述 * feat: play online mp3 (#181) Co-authored-by: 欣南科技 <huangrongzhuang@xin-nan.com> * 修改前端代码 * update:调整项目目录 * update:优化 * update:配置文件去除8002端口 * update:增加开发说明 * update:更新开发协议 --------- Co-authored-by: kalicyh <34980061+kaliCYH@users.noreply.github.com> Co-authored-by: hrz <1710360675@qq.com> Co-authored-by: freshlife001 <talent@mises.site> Co-authored-by: CGD <3030332422@qq.com>
This commit is contained in:
co-authored by
kalicyh
hrz
freshlife001
CGD
parent
8b151d07c2
commit
0e43748fdc
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import base64
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import os
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import uuid
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import requests
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import ormsgpack
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from pathlib import Path
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from pydantic import BaseModel, Field, conint, model_validator
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from typing_extensions import Annotated
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from datetime import datetime
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from typing import Literal
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from core.providers.tts.base import TTSProviderBase
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from config.logger import setup_logging
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TAG = __name__
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logger = setup_logging()
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class ServeReferenceAudio(BaseModel):
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audio: bytes
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text: str
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@model_validator(mode="before")
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def decode_audio(cls, values):
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audio = values.get("audio")
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if (
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isinstance(audio, str) and len(audio) > 255
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): # Check if audio is a string (Base64)
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try:
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values["audio"] = base64.b64decode(audio)
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except Exception as e:
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# If the audio is not a valid base64 string, we will just ignore it and let the server handle it
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pass
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return values
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def __repr__(self) -> str:
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return f"ServeReferenceAudio(text={self.text!r}, audio_size={len(self.audio)})"
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class ServeTTSRequest(BaseModel):
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text: str
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chunk_length: Annotated[int, conint(ge=100, le=300, strict=True)] = 200
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# Audio format
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format: Literal["wav", "pcm", "mp3"] = "wav"
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# References audios for in-context learning
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references: list[ServeReferenceAudio] = []
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# Reference id
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# For example, if you want use https://fish.audio/m/7f92f8afb8ec43bf81429cc1c9199cb1/
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# Just pass 7f92f8afb8ec43bf81429cc1c9199cb1
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reference_id: str | None = None
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seed: int | None = None
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use_memory_cache: Literal["on", "off"] = "off"
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# Normalize text for en & zh, this increase stability for numbers
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normalize: bool = True
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# not usually used below
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streaming: bool = False
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max_new_tokens: int = 1024
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top_p: Annotated[float, Field(ge=0.1, le=1.0, strict=True)] = 0.7
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repetition_penalty: Annotated[float, Field(ge=0.9, le=2.0, strict=True)] = 1.2
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temperature: Annotated[float, Field(ge=0.1, le=1.0, strict=True)] = 0.7
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class Config:
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# Allow arbitrary types for pytorch related types
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arbitrary_types_allowed = True
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def audio_to_bytes(file_path):
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if not file_path or not Path(file_path).exists():
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return None
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with open(file_path, "rb") as wav_file:
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wav = wav_file.read()
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return wav
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def read_ref_text(ref_text):
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path = Path(ref_text)
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if path.exists() and path.is_file():
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with path.open("r", encoding="utf-8") as file:
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return file.read()
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return ref_text
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class TTSProvider(TTSProviderBase):
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def __init__(self, config, delete_audio_file):
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super().__init__(config, delete_audio_file)
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self.reference_id = config.get("reference_id")
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self.reference_audio = config.get("reference_audio",[])
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self.reference_text = config.get("reference_text",[])
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self.format = config.get("format","wav")
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self.channels = config.get("channels",1)
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self.rate = config.get("rate",44100)
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self.api_key = config.get("api_key","YOUR_API_KEY")
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if "你" in self.api_key:
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logger.bind(tag=TAG).error("你还没配置FishSpeech TTS的密钥,请在配置文件中配置密钥,否则无法正常工作")
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return
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self.normalize = config.get("normalize",True)
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self.max_new_tokens = config.get("max_new_tokens",1024)
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self.chunk_length = config.get("chunk_length",200)
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self.top_p = config.get("top_p",0.7)
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self.repetition_penalty = config.get("repetition_penalty",1.2)
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self.temperature = config.get("temperature",0.7)
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self.streaming = config.get("streaming",False)
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self.use_memory_cache = config.get("use_memory_cache","on")
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self.seed = config.get("seed")
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self.api_url = config.get("api_url","http://127.0.0.1:8080/v1/tts")
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def generate_filename(self, extension=".wav"):
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return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
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async def text_to_speak(self, text, output_file):
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# Prepare reference data
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byte_audios = [audio_to_bytes(ref_audio) for ref_audio in self.reference_audio]
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ref_texts = [read_ref_text(ref_text) for ref_text in self.reference_text]
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data = {
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"text": text,
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"references": [
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ServeReferenceAudio(
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audio=audio if audio else b"", text=text
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)
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for text, audio in zip(ref_texts, byte_audios)
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],
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"reference_id": self.reference_id,
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"normalize": self.normalize,
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"format": self.format,
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"max_new_tokens": self.max_new_tokens,
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"chunk_length": self.chunk_length,
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"top_p": self.top_p,
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"repetition_penalty": self.repetition_penalty,
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"temperature": self.temperature,
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"streaming": self.streaming,
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"use_memory_cache": self.use_memory_cache,
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"seed": self.seed,
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}
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pydantic_data = ServeTTSRequest(**data)
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response = requests.post(
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self.api_url,
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data=ormsgpack.packb(pydantic_data, option=ormsgpack.OPT_SERIALIZE_PYDANTIC),
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headers={
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"Authorization": f"Bearer {self.api_key}",
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"Content-Type": "application/msgpack",
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},
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)
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if response.status_code == 200:
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audio_content = response.content
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with open(output_file, "wb") as audio_file:
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audio_file.write(audio_content)
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else:
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print(f"Request failed with status code {response.status_code}")
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print(response.json())
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