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xiaozhi-esp32-server/core/providers/tts/fishspeech.py
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2025-03-05 17:26:32 +08:00

266 lines
10 KiB
Python

import base64
import os
import traceback
import uuid
import queue
import io
import numpy as np
import requests
import ormsgpack
from pathlib import Path
import torch
import torchaudio
from pydantic import BaseModel, Field, conint, model_validator
from pydub import AudioSegment
from typing_extensions import Annotated
from datetime import datetime
from typing import Literal
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")
if "你" in self.api_key:
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")
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())
def _get_audio_from_tts(self, data_bytes):
tts_speech = torch.from_numpy(np.array(np.frombuffer(data_bytes, dtype=np.int16))).unsqueeze(dim=0)
with io.BytesIO() as bf:
torchaudio.save(bf, tts_speech, 44100, format="wav")
audio = AudioSegment.from_file(bf, format="wav")
audio = audio.set_channels(1).set_frame_rate(16000)
return audio
async def text_to_speak_stream(self, text, queue: queue.Queue, text_index=0):
try:
# 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)
audio_buff = None
chunk_total = b''
last_raw = b''
audio_raw = b''
print("请求tts")
with 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",
},
) as response:
if response.status_code == 200:
for chunk in response.iter_content():
# 拼接当前块和上一块数据
chunk_total += chunk
# 最后一个是静音,说明是一个完整的音频
if len(chunk_total) % 2 == 0 and chunk_total[-2:] == b'\x00\x00':
audio = self._get_audio_from_tts(chunk_total)
audio_raw = audio_raw + audio.raw_data
#长度凑够2贞开始发送,60ms*4=240ms
if len(audio_raw) >= 7680:
duration = 60 * len(audio_raw) // 1920
if (len(audio_raw) % 1920) > 0:
duration += 60
duration = duration / 1000.0
logger.bind(tag=TAG).info(f'发送数据长度:{len(audio_raw)}')
opus_datas = self.wav_to_opus_data_audio_raw(audio_raw)
queue.put({
"data": opus_datas,
"duration": duration,
"end": False,
"text_index": text_index
})
audio_raw = b''
chunk_total = b''
if len(chunk_total) > 0:
audio = self._get_audio_from_tts(chunk_total)
audio_raw = audio_raw + audio.raw_data
duration = 60 * len(audio_raw) // 1920
if (len(audio_raw) % 1920) > 0:
duration += 60
duration = duration / 1000.0
# 把 audio 转成 opus
logger.bind(tag=TAG).info(f'发送数据长度:{len(audio_raw)}')
opus_datas = self.wav_to_opus_data_audio_raw(audio_raw)
queue.put({
"data": opus_datas,
"duration": duration,
"end": False
})
else:
print('请求失败:', response.status_code, response.text)
queue.put({
"data": None,
"end": True
})
except Exception as e:
logger.bind(tag=TAG).error("tts发生错误")
traceback.print_exc()
raise e