Files
xiaozhi-esp32-server/main/xiaozhi-server/core/providers/tts/fishspeech.py
T
+2 dd5fecdf5a Merge old pr (#235)
* 新增openai tts (#137)

On branch openai_tts
Changes to be committed:
	modified:   config.yaml
	new file:   core/providers/tts/openai_tts.py

Co-authored-by: 欣南科技 <huangrongzhuang@xin-nan.com>

* Update config.yaml (#201)

* 修改了coze的实现方式,使用了v3接口,修复了只能对话一次的bug

* 使用mem0ai api实现记忆功能

* Fix docker space (#179)

* update:调试docker编译缓存

* update:调试docker编译

* update:调试docker编译容量

* update:调试docker编译

---------

Co-authored-by: hrz <1710360675@qq.com>

* TTS处理增加index信息,完善TTS出错时的处理 (#177)

* TTS处理增加index信息,完善TTS出错时的处理

* 第一句语音index处理,歌曲播放index处理

* update:去掉了无用变量,修复continuebug

---------

Co-authored-by: hrz <1710360675@qq.com>

* update:更新demo (#180)

Co-authored-by: hrz <1710360675@qq.com>

* 优化chat中异步调用方式

* 记忆增加时间,以便大模型判断先后关系

* 记忆对时间戳排序,便于梳理前后关系

* add: 不跟踪music下的文件的更改 (#187)

* Mem0ai (#199)

* update:增加mem0ai包依赖

* update:文档增加mem0ai的描述

---------

Co-authored-by: hrz <1710360675@qq.com>

* Update config.yaml

* Update config.yaml

* Update config.yaml

* Update config.yaml

---------

Co-authored-by: strix214 <a1425699467@gmail.com>
Co-authored-by: 玄凤科技 <eric230308@gmail.com>
Co-authored-by: HonestQiao <honestqiao@163.com>
Co-authored-by: 欣南科技 <huangrongzhuang@xin-nan.com>
Co-authored-by: hrz <1710360675@qq.com>
Co-authored-by: TOM88812 <ttangxs0808@163.com>
Co-authored-by: 香草味的纳西妲 <151599587+NyaOH-Nahida@users.noreply.github.com>

* update:添加忽略文件

* feat: 增加fastgpt适配:支持think、variables参数 (#213)

* 修改了coze的实现方式,使用了v3接口,修复了只能对话一次的bug

* 使用mem0ai api实现记忆功能

* Fix docker space (#179)

* update:调试docker编译缓存

* update:调试docker编译

* update:调试docker编译容量

* update:调试docker编译

---------

Co-authored-by: hrz <1710360675@qq.com>

* TTS处理增加index信息,完善TTS出错时的处理 (#177)

* TTS处理增加index信息,完善TTS出错时的处理

* 第一句语音index处理,歌曲播放index处理

* update:去掉了无用变量,修复continuebug

---------

Co-authored-by: hrz <1710360675@qq.com>

* update:更新demo (#180)

Co-authored-by: hrz <1710360675@qq.com>

* 优化chat中异步调用方式

* 记忆增加时间,以便大模型判断先后关系

* 记忆对时间戳排序,便于梳理前后关系

* add: 不跟踪music下的文件的更改 (#187)

* Mem0ai (#199)

* update:增加mem0ai包依赖

* update:文档增加mem0ai的描述

---------

Co-authored-by: hrz <1710360675@qq.com>

* feat: 增加fastgpt适配:支持think、variables参数

---------

Co-authored-by: strix214 <a1425699467@gmail.com>
Co-authored-by: 玄凤科技 <eric230308@gmail.com>
Co-authored-by: HonestQiao <honestqiao@163.com>
Co-authored-by: 欣南科技 <huangrongzhuang@xin-nan.com>
Co-authored-by: hrz <1710360675@qq.com>
Co-authored-by: TOM88812 <ttangxs0808@163.com>
Co-authored-by: 香草味的纳西妲 <151599587+NyaOH-Nahida@users.noreply.github.com>
Co-authored-by: pt <ptsghs@163.com>

* update:合并最新代码

* update:合并群友pr

* update:openaitts 测试通过

* update:ACGNTTS测试完成

---------

Co-authored-by: Chris <119588753+Chris-websketch@users.noreply.github.com>
Co-authored-by: ikun441 <3497689533@qq.com>
Co-authored-by: strix214 <a1425699467@gmail.com>
Co-authored-by: 玄凤科技 <eric230308@gmail.com>
Co-authored-by: HonestQiao <honestqiao@163.com>
Co-authored-by: hrz <1710360675@qq.com>
Co-authored-by: TOM88812 <ttangxs0808@163.com>
Co-authored-by: 香草味的纳西妲 <151599587+NyaOH-Nahida@users.noreply.github.com>
Co-authored-by: ptisnoob <49125305+ptisnoob@users.noreply.github.com>
Co-authored-by: pt <ptsghs@163.com>
2025-03-07 18:25:18 +08:00

160 lines
5.6 KiB
Python

import base64
import os
import uuid
import requests
import ormsgpack
from pathlib import Path
from pydantic import BaseModel, Field, conint, model_validator
from typing_extensions import Annotated
from datetime import datetime
from typing import Literal
from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class ServeReferenceAudio(BaseModel):
audio: bytes
text: str
@model_validator(mode="before")
def decode_audio(cls, values):
audio = values.get("audio")
if (
isinstance(audio, str) and len(audio) > 255
): # Check if audio is a string (Base64)
try:
values["audio"] = base64.b64decode(audio)
except Exception as e:
# If the audio is not a valid base64 string, we will just ignore it and let the server handle it
pass
return values
def __repr__(self) -> str:
return f"ServeReferenceAudio(text={self.text!r}, audio_size={len(self.audio)})"
class ServeTTSRequest(BaseModel):
text: str
chunk_length: Annotated[int, conint(ge=100, le=300, strict=True)] = 200
# Audio format
format: Literal["wav", "pcm", "mp3"] = "wav"
# References audios for in-context learning
references: list[ServeReferenceAudio] = []
# Reference id
# For example, if you want use https://fish.audio/m/7f92f8afb8ec43bf81429cc1c9199cb1/
# Just pass 7f92f8afb8ec43bf81429cc1c9199cb1
reference_id: str | None = None
seed: int | None = None
use_memory_cache: Literal["on", "off"] = "off"
# Normalize text for en & zh, this increase stability for numbers
normalize: bool = True
# not usually used below
streaming: bool = False
max_new_tokens: int = 1024
top_p: Annotated[float, Field(ge=0.1, le=1.0, strict=True)] = 0.7
repetition_penalty: Annotated[float, Field(ge=0.9, le=2.0, strict=True)] = 1.2
temperature: Annotated[float, Field(ge=0.1, le=1.0, strict=True)] = 0.7
class Config:
# Allow arbitrary types for pytorch related types
arbitrary_types_allowed = True
def audio_to_bytes(file_path):
if not file_path or not Path(file_path).exists():
return None
with open(file_path, "rb") as wav_file:
wav = wav_file.read()
return wav
def read_ref_text(ref_text):
path = Path(ref_text)
if path.exists() and path.is_file():
with path.open("r", encoding="utf-8") as file:
return file.read()
return ref_text
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.reference_id = config.get("reference_id")
self.reference_audio = config.get("reference_audio", [])
self.reference_text = config.get("reference_text", [])
self.format = config.get("format", "wav")
self.channels = config.get("channels", 1)
self.rate = config.get("rate", 44100)
self.api_key = config.get("api_key", "YOUR_API_KEY")
have_key = check_model_key("FishSpeech TTS", self.api_key)
if not have_key:
return
self.normalize = config.get("normalize", True)
self.max_new_tokens = config.get("max_new_tokens", 1024)
self.chunk_length = config.get("chunk_length", 200)
self.top_p = config.get("top_p", 0.7)
self.repetition_penalty = config.get("repetition_penalty", 1.2)
self.temperature = config.get("temperature", 0.7)
self.streaming = config.get("streaming", False)
self.use_memory_cache = config.get("use_memory_cache", "on")
self.seed = config.get("seed")
self.api_url = config.get("api_url", "http://127.0.0.1:8080/v1/tts")
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
# Prepare reference data
byte_audios = [audio_to_bytes(ref_audio) for ref_audio in self.reference_audio]
ref_texts = [read_ref_text(ref_text) for ref_text in self.reference_text]
data = {
"text": text,
"references": [
ServeReferenceAudio(
audio=audio if audio else b"", text=text
)
for text, audio in zip(ref_texts, byte_audios)
],
"reference_id": self.reference_id,
"normalize": self.normalize,
"format": self.format,
"max_new_tokens": self.max_new_tokens,
"chunk_length": self.chunk_length,
"top_p": self.top_p,
"repetition_penalty": self.repetition_penalty,
"temperature": self.temperature,
"streaming": self.streaming,
"use_memory_cache": self.use_memory_cache,
"seed": self.seed,
}
pydantic_data = ServeTTSRequest(**data)
response = requests.post(
self.api_url,
data=ormsgpack.packb(pydantic_data, option=ormsgpack.OPT_SERIALIZE_PYDANTIC),
headers={
"Authorization": f"Bearer {self.api_key}",
"Content-Type": "application/msgpack",
},
)
if response.status_code == 200:
audio_content = response.content
with open(output_file, "wb") as audio_file:
audio_file.write(audio_content)
else:
print(f"Request failed with status code {response.status_code}")
print(response.json())