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GPT-SoVITS/api.py
Spr_Aachen dffefeab84 Update api.py 
- Address parameter assignment issue for DefaultRefer class
- Standarlize parameter passing in initialization section
2025-04-01 10:50:40 +08:00

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"""
# api.py usage
` python api.py -dr "123.wav" -dt "一二三。" -dl "zh" `
## 执行参数:
`-s` - `SoVITS模型路径, 可在 config.py 中指定`
`-g` - `GPT模型路径, 可在 config.py 中指定`
调用请求缺少参考音频时使用
`-dr` - `默认参考音频路径`
`-dt` - `默认参考音频文本`
`-dl` - `默认参考音频语种, "中文","英文","日文","韩文","粤语,"zh","en","ja","ko","yue"`
`-d` - `推理设备, "cuda","cpu"`
`-a` - `绑定地址, 默认"127.0.0.1"`
`-p` - `绑定端口, 默认9880, 可在 config.py 中指定`
`-fp` - `覆盖 config.py 使用全精度`
`-hp` - `覆盖 config.py 使用半精度`
`-sm` - `流式返回模式, 默认不启用, "close","c", "normal","n", "keepalive","k"`
·-mt` - `返回的音频编码格式, 流式默认ogg, 非流式默认wav, "wav", "ogg", "aac"`
·-st` - `返回的音频数据类型, 默认int16, "int16", "int32"`
·-cp` - `文本切分符号设定, 默认为空, 以",.，。"字符串的方式传入`
`-hb` - `cnhubert路径`
`-b` - `bert路径`
## 调用:
### 推理
endpoint: `/`
使用执行参数指定的参考音频:
GET:
`http://127.0.0.1:9880?text=先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。&text_language=zh`
POST:
```json
{
"text": "先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。",
"text_language": "zh"
}
```
使用执行参数指定的参考音频并设定分割符号:
GET:
`http://127.0.0.1:9880?text=先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。&text_language=zh&cut_punc=，。`
POST:
```json
{
"text": "先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。",
"text_language": "zh",
"cut_punc": "，。",
}
```
手动指定当次推理所使用的参考音频:
GET:
`http://127.0.0.1:9880?refer_wav_path=123.wav&prompt_text=一二三。&prompt_language=zh&text=先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。&text_language=zh`
POST:
```json
{
"refer_wav_path": "123.wav",
"prompt_text": "一二三。",
"prompt_language": "zh",
"text": "先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。",
"text_language": "zh"
}
```
RESP:
成功: 直接返回 wav 音频流， http code 200
失败: 返回包含错误信息的 json, http code 400
手动指定当次推理所使用的参考音频，并提供参数:
GET:
`http://127.0.0.1:9880?refer_wav_path=123.wav&prompt_text=一二三。&prompt_language=zh&text=先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。&text_language=zh&top_k=20&top_p=0.6&temperature=0.6&speed=1&inp_refs="456.wav"&inp_refs="789.wav"`
POST:
```json
{
"refer_wav_path": "123.wav",
"prompt_text": "一二三。",
"prompt_language": "zh",
"text": "先帝创业未半而中道崩殂，今天下三分，益州疲弊，此诚危急存亡之秋也。",
"text_language": "zh",
"top_k": 20,
"top_p": 0.6,
"temperature": 0.6,
"speed": 1,
"inp_refs": ["456.wav","789.wav"]
}
```
RESP:
成功: 直接返回 wav 音频流， http code 200
失败: 返回包含错误信息的 json, http code 400
### 更换默认参考音频
endpoint: `/change_refer`
key与推理端一样
GET:
`http://127.0.0.1:9880/change_refer?refer_wav_path=123.wav&prompt_text=一二三。&prompt_language=zh`
POST:
```json
{
"refer_wav_path": "123.wav",
"prompt_text": "一二三。",
"prompt_language": "zh"
}
```
RESP:
成功: json, http code 200
失败: json, 400
### 命令控制
endpoint: `/control`
command:
"restart": 重新运行
"exit": 结束运行
GET:
`http://127.0.0.1:9880/control?command=restart`
POST:
```json
{
"command": "restart"
}
```
RESP: 无
"""
import argparse
import os,re
import sys
now_dir = os.getcwd()
sys.path.append(now_dir)
sys.path.append("%s/GPT_SoVITS" % (now_dir))
import signal
from text.LangSegmenter import LangSegmenter
from time import time as ttime
import torch, torchaudio
import librosa
import soundfile as sf
from fastapi import FastAPI, Request, Query, HTTPException
from fastapi.responses import StreamingResponse, JSONResponse
import uvicorn
from transformers import AutoModelForMaskedLM, AutoTokenizer
import numpy as np
from feature_extractor import cnhubert
from io import BytesIO
from module.models import SynthesizerTrn, SynthesizerTrnV3
from peft import LoraConfig, PeftModel, get_peft_model
from AR.models.t2s_lightning_module import Text2SemanticLightningModule
from text import cleaned_text_to_sequence
from text.cleaner import clean_text
from module.mel_processing import spectrogram_torch
from tools.my_utils import load_audio
import config as global_config
import logging
import subprocess
class DefaultRefer:
def __init__(self, path, text, language):
self.path = path
self.text = text
self.language = language
def is_ready(self) -> bool:
return is_full(self.path, self.text, self.language)
def is_empty(*items): # 任意一项不为空返回False
for item in items:
if item is not None and item != "":
return False
return True
def is_full(*items): # 任意一项为空返回False
for item in items:
if item is None or item == "":
return False
return True
def init_bigvgan():
global bigvgan_model
from BigVGAN import bigvgan
bigvgan_model = bigvgan.BigVGAN.from_pretrained("%s/GPT_SoVITS/pretrained_models/models--nvidia--bigvgan_v2_24khz_100band_256x" % (now_dir,), use_cuda_kernel=False) # if True, RuntimeError: Ninja is required to load C++ extensions
# remove weight norm in the model and set to eval mode
bigvgan_model.remove_weight_norm()
bigvgan_model = bigvgan_model.eval()
if is_half == True:
bigvgan_model = bigvgan_model.half().to(device)
else:
bigvgan_model = bigvgan_model.to(device)
resample_transform_dict={}
def resample(audio_tensor, sr0):
global resample_transform_dict
if sr0 not in resample_transform_dict:
resample_transform_dict[sr0] = torchaudio.transforms.Resample(
sr0, 24000
).to(device)
return resample_transform_dict[sr0](audio_tensor)
from module.mel_processing import spectrogram_torch,mel_spectrogram_torch
spec_min = -12
spec_max = 2
def norm_spec(x):
return (x - spec_min) / (spec_max - spec_min) * 2 - 1
def denorm_spec(x):
return (x + 1) / 2 * (spec_max - spec_min) + spec_min
mel_fn=lambda x: mel_spectrogram_torch(x, **{
"n_fft": 1024,
"win_size": 1024,
"hop_size": 256,
"num_mels": 100,
"sampling_rate": 24000,
"fmin": 0,
"fmax": None,
"center": False
})
sr_model=None
def audio_sr(audio,sr):
global sr_model
if sr_model==None:
from tools.audio_sr import AP_BWE
try:
sr_model=AP_BWE(device,DictToAttrRecursive)
except FileNotFoundError:
logger.info("你没有下载超分模型的参数，因此不进行超分。如想超分请先参照教程把文件下载")
return audio.cpu().detach().numpy(),sr
return sr_model(audio,sr)
class Speaker:
def __init__(self, name, gpt, sovits, phones = None, bert = None, prompt = None):
self.name = name
self.sovits = sovits
self.gpt = gpt
self.phones = phones
self.bert = bert
self.prompt = prompt
speaker_list = {}
class Sovits:
def __init__(self, vq_model, hps):
self.vq_model = vq_model
self.hps = hps
from process_ckpt import get_sovits_version_from_path_fast,load_sovits_new
def get_sovits_weights(sovits_path):
path_sovits_v3="GPT_SoVITS/pretrained_models/s2Gv3.pth"
is_exist_s2gv3=os.path.exists(path_sovits_v3)
version, model_version, if_lora_v3=get_sovits_version_from_path_fast(sovits_path)
if if_lora_v3==True and is_exist_s2gv3==False:
logger.info("SoVITS V3 底模缺失，无法加载相应 LoRA 权重")
dict_s2 = load_sovits_new(sovits_path)
hps = dict_s2["config"]
hps = DictToAttrRecursive(hps)
hps.model.semantic_frame_rate = "25hz"
if 'enc_p.text_embedding.weight' not in dict_s2['weight']:
hps.model.version = "v2"#v3model,v2sybomls
elif dict_s2['weight']['enc_p.text_embedding.weight'].shape[0] == 322:
hps.model.version = "v1"
else:
hps.model.version = "v2"
if model_version == "v3":
hps.model.version = "v3"
model_params_dict = vars(hps.model)
if model_version!="v3":
vq_model = SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**model_params_dict
)
else:
vq_model = SynthesizerTrnV3(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**model_params_dict
)
init_bigvgan()
model_version=hps.model.version
logger.info(f"模型版本: {model_version}")
if ("pretrained" not in sovits_path):
try:
del vq_model.enc_q
except:pass
if is_half == True:
vq_model = vq_model.half().to(device)
else:
vq_model = vq_model.to(device)
vq_model.eval()
if if_lora_v3 == False:
vq_model.load_state_dict(dict_s2["weight"], strict=False)
else:
vq_model.load_state_dict(load_sovits_new(path_sovits_v3)["weight"], strict=False)
lora_rank=dict_s2["lora_rank"]
lora_config = LoraConfig(
target_modules=["to_k", "to_q", "to_v", "to_out.0"],
r=lora_rank,
lora_alpha=lora_rank,
init_lora_weights=True,
)
vq_model.cfm = get_peft_model(vq_model.cfm, lora_config)
vq_model.load_state_dict(dict_s2["weight"], strict=False)
vq_model.cfm = vq_model.cfm.merge_and_unload()
# torch.save(vq_model.state_dict(),"merge_win.pth")
vq_model.eval()
sovits = Sovits(vq_model, hps)
return sovits
class Gpt:
def __init__(self, max_sec, t2s_model):
self.max_sec = max_sec
self.t2s_model = t2s_model
global hz
hz = 50
def get_gpt_weights(gpt_path):
dict_s1 = torch.load(gpt_path, map_location="cpu")
config = dict_s1["config"]
max_sec = config["data"]["max_sec"]
t2s_model = Text2SemanticLightningModule(config, "****", is_train=False)
t2s_model.load_state_dict(dict_s1["weight"])
if is_half == True:
t2s_model = t2s_model.half()
t2s_model = t2s_model.to(device)
t2s_model.eval()
# total = sum([param.nelement() for param in t2s_model.parameters()])
# logger.info("Number of parameter: %.2fM" % (total / 1e6))
gpt = Gpt(max_sec, t2s_model)
return gpt
def change_gpt_sovits_weights(gpt_path,sovits_path):
try:
gpt = get_gpt_weights(gpt_path)
sovits = get_sovits_weights(sovits_path)
except Exception as e:
return JSONResponse({"code": 400, "message": str(e)}, status_code=400)
speaker_list["default"] = Speaker(name="default", gpt=gpt, sovits=sovits)
return JSONResponse({"code": 0, "message": "Success"}, status_code=200)
def get_bert_feature(text, word2ph):
with torch.no_grad():
inputs = tokenizer(text, return_tensors="pt")
for i in inputs:
inputs[i] = inputs[i].to(device) #####输入是long不用管精度问题，精度随bert_model
res = bert_model(**inputs, output_hidden_states=True)
res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1]
assert len(word2ph) == len(text)
phone_level_feature = []
for i in range(len(word2ph)):
repeat_feature = res[i].repeat(word2ph[i], 1)
phone_level_feature.append(repeat_feature)
phone_level_feature = torch.cat(phone_level_feature, dim=0)
# if(is_half==True):phone_level_feature=phone_level_feature.half()
return phone_level_feature.T
def clean_text_inf(text, language, version):
language = language.replace("all_","")
phones, word2ph, norm_text = clean_text(text, language, version)
phones = cleaned_text_to_sequence(phones, version)
return phones, word2ph, norm_text
def get_bert_inf(phones, word2ph, norm_text, language):
language=language.replace("all_","")
if language == "zh":
bert = get_bert_feature(norm_text, word2ph).to(device)#.to(dtype)
else:
bert = torch.zeros(
(1024, len(phones)),
dtype=torch.float16 if is_half == True else torch.float32,
).to(device)
return bert
from text import chinese
def get_phones_and_bert(text,language,version,final=False):
if language in {"en", "all_zh", "all_ja", "all_ko", "all_yue"}:
formattext = text
while " " in formattext:
formattext = formattext.replace(" ", " ")
if language == "all_zh":
if re.search(r'[A-Za-z]', formattext):
formattext = re.sub(r'[a-z]', lambda x: x.group(0).upper(), formattext)
formattext = chinese.mix_text_normalize(formattext)
return get_phones_and_bert(formattext,"zh",version)
else:
phones, word2ph, norm_text = clean_text_inf(formattext, language, version)
bert = get_bert_feature(norm_text, word2ph).to(device)
elif language == "all_yue" and re.search(r'[A-Za-z]', formattext):
formattext = re.sub(r'[a-z]', lambda x: x.group(0).upper(), formattext)
formattext = chinese.mix_text_normalize(formattext)
return get_phones_and_bert(formattext,"yue",version)
else:
phones, word2ph, norm_text = clean_text_inf(formattext, language, version)
bert = torch.zeros(
(1024, len(phones)),
dtype=torch.float16 if is_half == True else torch.float32,
).to(device)
elif language in {"zh", "ja", "ko", "yue", "auto", "auto_yue"}:
textlist=[]
langlist=[]
if language == "auto":
for tmp in LangSegmenter.getTexts(text):
langlist.append(tmp["lang"])
textlist.append(tmp["text"])
elif language == "auto_yue":
for tmp in LangSegmenter.getTexts(text):
if tmp["lang"] == "zh":
tmp["lang"] = "yue"
langlist.append(tmp["lang"])
textlist.append(tmp["text"])
else:
for tmp in LangSegmenter.getTexts(text):
if tmp["lang"] == "en":
langlist.append(tmp["lang"])
else:
# 因无法区别中日韩文汉字,以用户输入为准
langlist.append(language)
textlist.append(tmp["text"])
phones_list = []
bert_list = []
norm_text_list = []
for i in range(len(textlist)):
lang = langlist[i]
phones, word2ph, norm_text = clean_text_inf(textlist[i], lang, version)
bert = get_bert_inf(phones, word2ph, norm_text, lang)
phones_list.append(phones)
norm_text_list.append(norm_text)
bert_list.append(bert)
bert = torch.cat(bert_list, dim=1)
phones = sum(phones_list, [])
norm_text = ''.join(norm_text_list)
if not final and len(phones) < 6:
return get_phones_and_bert("." + text,language,version,final=True)
return phones,bert.to(torch.float16 if is_half == True else torch.float32),norm_text
class DictToAttrRecursive(dict):
def __init__(self, input_dict):
super().__init__(input_dict)
for key, value in input_dict.items():
if isinstance(value, dict):
value = DictToAttrRecursive(value)
self[key] = value
setattr(self, key, value)
def __getattr__(self, item):
try:
return self[item]
except KeyError:
raise AttributeError(f"Attribute {item} not found")
def __setattr__(self, key, value):
if isinstance(value, dict):
value = DictToAttrRecursive(value)
super(DictToAttrRecursive, self).__setitem__(key, value)
super().__setattr__(key, value)
def __delattr__(self, item):
try:
del self[item]
except KeyError:
raise AttributeError(f"Attribute {item} not found")
def get_spepc(hps, filename):
audio,_ = librosa.load(filename, int(hps.data.sampling_rate))
audio = torch.FloatTensor(audio)
maxx=audio.abs().max()
if(maxx>1):
audio/=min(2,maxx)
audio_norm = audio
audio_norm = audio_norm.unsqueeze(0)
spec = spectrogram_torch(audio_norm, hps.data.filter_length, hps.data.sampling_rate, hps.data.hop_length,
hps.data.win_length, center=False)
return spec
def pack_audio(audio_bytes, data, rate):
if media_type == "ogg":
audio_bytes = pack_ogg(audio_bytes, data, rate)
elif media_type == "aac":
audio_bytes = pack_aac(audio_bytes, data, rate)
else:
# wav无法流式, 先暂存raw
audio_bytes = pack_raw(audio_bytes, data, rate)
return audio_bytes
def pack_ogg(audio_bytes, data, rate):
# Author: AkagawaTsurunaki
# Issue:
# Stack overflow probabilistically occurs
# when the function `sf_writef_short` of `libsndfile_64bit.dll` is called
# using the Python library `soundfile`
# Note:
# This is an issue related to `libsndfile`, not this project itself.
# It happens when you generate a large audio tensor (about 499804 frames in my PC)
# and try to convert it to an ogg file.
# Related:
# https://github.com/RVC-Boss/GPT-SoVITS/issues/1199
# https://github.com/libsndfile/libsndfile/issues/1023
# https://github.com/bastibe/python-soundfile/issues/396
# Suggestion:
# Or split the whole audio data into smaller audio segment to avoid stack overflow?
def handle_pack_ogg():
with sf.SoundFile(audio_bytes, mode='w', samplerate=rate, channels=1, format='ogg') as audio_file:
audio_file.write(data)
import threading
# See: https://docs.python.org/3/library/threading.html
# The stack size of this thread is at least 32768
# If stack overflow error still occurs, just modify the `stack_size`.
# stack_size = n * 4096, where n should be a positive integer.
# Here we chose n = 4096.
stack_size = 4096 * 4096
try:
threading.stack_size(stack_size)
pack_ogg_thread = threading.Thread(target=handle_pack_ogg)
pack_ogg_thread.start()
pack_ogg_thread.join()
except RuntimeError as e:
# If changing the thread stack size is unsupported, a RuntimeError is raised.
print("RuntimeError: {}".format(e))
print("Changing the thread stack size is unsupported.")
except ValueError as e:
# If the specified stack size is invalid, a ValueError is raised and the stack size is unmodified.
print("ValueError: {}".format(e))
print("The specified stack size is invalid.")
return audio_bytes
def pack_raw(audio_bytes, data, rate):
audio_bytes.write(data.tobytes())
return audio_bytes
def pack_wav(audio_bytes, rate):
if is_int32:
data = np.frombuffer(audio_bytes.getvalue(),dtype=np.int32)
wav_bytes = BytesIO()
sf.write(wav_bytes, data, rate, format='WAV', subtype='PCM_32')
else:
data = np.frombuffer(audio_bytes.getvalue(),dtype=np.int16)
wav_bytes = BytesIO()
sf.write(wav_bytes, data, rate, format='WAV')
return wav_bytes
def pack_aac(audio_bytes, data, rate):
if is_int32:
pcm = 's32le'
bit_rate = '256k'
else:
pcm = 's16le'
bit_rate = '128k'
process = subprocess.Popen([
'ffmpeg',
'-f', pcm, # 输入16位有符号小端整数PCM
'-ar', str(rate), # 设置采样率
'-ac', '1', # 单声道
'-i', 'pipe:0', # 从管道读取输入
'-c:a', 'aac', # 音频编码器为AAC
'-b:a', bit_rate, # 比特率
'-vn', # 不包含视频
'-f', 'adts', # 输出AAC数据流格式
'pipe:1' # 将输出写入管道
], stdin=subprocess.PIPE, stdout=subprocess.PIPE, stderr=subprocess.PIPE)
out, _ = process.communicate(input=data.tobytes())
audio_bytes.write(out)
return audio_bytes
def read_clean_buffer(audio_bytes):
audio_chunk = audio_bytes.getvalue()
audio_bytes.truncate(0)
audio_bytes.seek(0)
return audio_bytes, audio_chunk
def cut_text(text, punc):
punc_list = [p for p in punc if p in {",", ".", ";", "?", "!", "、", "，", "。", "？", "！", "；", "：", "…"}]
if len(punc_list) > 0:
punds = r"[" + "".join(punc_list) + r"]"
text = text.strip("\n")
items = re.split(f"({punds})", text)
mergeitems = ["".join(group) for group in zip(items[::2], items[1::2])]
# 在句子不存在符号或句尾无符号的时候保证文本完整
if len(items)%2 == 1:
mergeitems.append(items[-1])
text = "\n".join(mergeitems)
while "\n\n" in text:
text = text.replace("\n\n", "\n")
return text
def only_punc(text):
return not any(t.isalnum() or t.isalpha() for t in text)
splits = {"，", "。", "？", "！", ",", ".", "?", "!", "~", ":", "：", "—", "…", }
def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language, top_k= 15, top_p = 0.6, temperature = 0.6, speed = 1, inp_refs = None, sample_steps = 32, if_sr = False, spk = "default"):
infer_sovits = speaker_list[spk].sovits
vq_model = infer_sovits.vq_model
hps = infer_sovits.hps
version = vq_model.version
infer_gpt = speaker_list[spk].gpt
t2s_model = infer_gpt.t2s_model
max_sec = infer_gpt.max_sec
t0 = ttime()
prompt_text = prompt_text.strip("\n")
if (prompt_text[-1] not in splits): prompt_text += "。" if prompt_language != "en" else "."
prompt_language, text = prompt_language, text.strip("\n")
dtype = torch.float16 if is_half == True else torch.float32
zero_wav = np.zeros(int(hps.data.sampling_rate * 0.3), dtype=np.float16 if is_half == True else np.float32)
with torch.no_grad():
wav16k, sr = librosa.load(ref_wav_path, sr=16000)
wav16k = torch.from_numpy(wav16k)
zero_wav_torch = torch.from_numpy(zero_wav)
if (is_half == True):
wav16k = wav16k.half().to(device)
zero_wav_torch = zero_wav_torch.half().to(device)
else:
wav16k = wav16k.to(device)
zero_wav_torch = zero_wav_torch.to(device)
wav16k = torch.cat([wav16k, zero_wav_torch])
ssl_content = ssl_model.model(wav16k.unsqueeze(0))["last_hidden_state"].transpose(1, 2) # .float()
codes = vq_model.extract_latent(ssl_content)
prompt_semantic = codes[0, 0]
prompt = prompt_semantic.unsqueeze(0).to(device)
if version != "v3":
refers=[]
if(inp_refs):
for path in inp_refs:
try:
refer = get_spepc(hps, path).to(dtype).to(device)
refers.append(refer)
except Exception as e:
logger.error(e)
if(len(refers)==0):
refers = [get_spepc(hps, ref_wav_path).to(dtype).to(device)]
else:
refer = get_spepc(hps, ref_wav_path).to(device).to(dtype)
t1 = ttime()
# os.environ['version'] = version
prompt_language = dict_language[prompt_language.lower()]
text_language = dict_language[text_language.lower()]
phones1, bert1, norm_text1 = get_phones_and_bert(prompt_text, prompt_language, version)
texts = text.split("\n")
audio_bytes = BytesIO()
for text in texts:
# 简单防止纯符号引发参考音频泄露
if only_punc(text):
continue
audio_opt = []
if (text[-1] not in splits): text += "。" if text_language != "en" else "."
phones2, bert2, norm_text2 = get_phones_and_bert(text, text_language, version)
bert = torch.cat([bert1, bert2], 1)
all_phoneme_ids = torch.LongTensor(phones1 + phones2).to(device).unsqueeze(0)
bert = bert.to(device).unsqueeze(0)
all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device)
t2 = ttime()
with torch.no_grad():
pred_semantic, idx = t2s_model.model.infer_panel(
all_phoneme_ids,
all_phoneme_len,
prompt,
bert,
# prompt_phone_len=ph_offset,
top_k = top_k,
top_p = top_p,
temperature = temperature,
early_stop_num=hz * max_sec)
pred_semantic = pred_semantic[:, -idx:].unsqueeze(0)
t3 = ttime()
if version != "v3":
audio = \
vq_model.decode(pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0),
refers,speed=speed).detach().cpu().numpy()[
0, 0] ###试试重建不带上prompt部分
else:
phoneme_ids0=torch.LongTensor(phones1).to(device).unsqueeze(0)
phoneme_ids1=torch.LongTensor(phones2).to(device).unsqueeze(0)
# print(11111111, phoneme_ids0, phoneme_ids1)
fea_ref,ge = vq_model.decode_encp(prompt.unsqueeze(0), phoneme_ids0, refer)
ref_audio, sr = torchaudio.load(ref_wav_path)
ref_audio=ref_audio.to(device).float()
if (ref_audio.shape[0] == 2):
ref_audio = ref_audio.mean(0).unsqueeze(0)
if sr!=24000:
ref_audio=resample(ref_audio,sr)
# print("ref_audio",ref_audio.abs().mean())
mel2 = mel_fn(ref_audio)
mel2 = norm_spec(mel2)
T_min = min(mel2.shape[2], fea_ref.shape[2])
mel2 = mel2[:, :, :T_min]
fea_ref = fea_ref[:, :, :T_min]
if (T_min > 468):
mel2 = mel2[:, :, -468:]
fea_ref = fea_ref[:, :, -468:]
T_min = 468
chunk_len = 934 - T_min
# print("fea_ref",fea_ref,fea_ref.shape)
# print("mel2",mel2)
mel2=mel2.to(dtype)
fea_todo, ge = vq_model.decode_encp(pred_semantic, phoneme_ids1, refer, ge,speed)
# print("fea_todo",fea_todo)
# print("ge",ge.abs().mean())
cfm_resss = []
idx = 0
while (1):
fea_todo_chunk = fea_todo[:, :, idx:idx + chunk_len]
if (fea_todo_chunk.shape[-1] == 0): break
idx += chunk_len
fea = torch.cat([fea_ref, fea_todo_chunk], 2).transpose(2, 1)
# set_seed(123)
cfm_res = vq_model.cfm.inference(fea, torch.LongTensor([fea.size(1)]).to(fea.device), mel2, sample_steps, inference_cfg_rate=0)
cfm_res = cfm_res[:, :, mel2.shape[2]:]
mel2 = cfm_res[:, :, -T_min:]
# print("fea", fea)
# print("mel2in", mel2)
fea_ref = fea_todo_chunk[:, :, -T_min:]
cfm_resss.append(cfm_res)
cmf_res = torch.cat(cfm_resss, 2)
cmf_res = denorm_spec(cmf_res)
if bigvgan_model==None:init_bigvgan()
with torch.inference_mode():
wav_gen = bigvgan_model(cmf_res)
audio=wav_gen[0][0].cpu().detach().numpy()
max_audio=np.abs(audio).max()
if max_audio>1:
audio/=max_audio
audio_opt.append(audio)
audio_opt.append(zero_wav)
audio_opt = np.concatenate(audio_opt, 0)
t4 = ttime()
sr = hps.data.sampling_rate if version != "v3" else 24000
if if_sr and sr == 24000:
audio_opt = torch.from_numpy(audio_opt).float().to(device)
audio_opt,sr=audio_sr(audio_opt.unsqueeze(0),sr)
max_audio=np.abs(audio_opt).max()
if max_audio > 1: audio_opt /= max_audio
sr = 48000
if is_int32:
audio_bytes = pack_audio(audio_bytes,(audio_opt * 2147483647).astype(np.int32),sr)
else:
audio_bytes = pack_audio(audio_bytes,(audio_opt * 32768).astype(np.int16),sr)
# logger.info("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3))
if stream_mode == "normal":
audio_bytes, audio_chunk = read_clean_buffer(audio_bytes)
yield audio_chunk
if not stream_mode == "normal":
if media_type == "wav":
sr = 48000 if if_sr else 24000
sr = hps.data.sampling_rate if version != "v3" else sr
audio_bytes = pack_wav(audio_bytes,sr)
yield audio_bytes.getvalue()
def handle_control(command):
if command == "restart":
os.execl(g_config.python_exec, g_config.python_exec, *sys.argv)
elif command == "exit":
os.kill(os.getpid(), signal.SIGTERM)
exit(0)
def handle_change(path, text, language):
if is_empty(path, text, language):
return JSONResponse({"code": 400, "message": '缺少任意一项以下参数: "path", "text", "language"'}, status_code=400)
if path != "" or path is not None:
default_refer.path = path
if text != "" or text is not None:
default_refer.text = text
if language != "" or language is not None:
default_refer.language = language
logger.info(f"当前默认参考音频路径: {default_refer.path}")
logger.info(f"当前默认参考音频文本: {default_refer.text}")
logger.info(f"当前默认参考音频语种: {default_refer.language}")
logger.info(f"is_ready: {default_refer.is_ready()}")
return JSONResponse({"code": 0, "message": "Success"}, status_code=200)
def handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cut_punc, top_k, top_p, temperature, speed, inp_refs, sample_steps, if_sr):
if (
refer_wav_path == "" or refer_wav_path is None
or prompt_text == "" or prompt_text is None
or prompt_language == "" or prompt_language is None
):
refer_wav_path, prompt_text, prompt_language = (
default_refer.path,
default_refer.text,
default_refer.language,
)
if not default_refer.is_ready():
return JSONResponse({"code": 400, "message": "未指定参考音频且接口无预设"}, status_code=400)
if not sample_steps in [4,8,16,32]:
sample_steps = 32
if cut_punc == None:
text = cut_text(text,default_cut_punc)
else:
text = cut_text(text,cut_punc)
return StreamingResponse(get_tts_wav(refer_wav_path, prompt_text, prompt_language, text, text_language, top_k, top_p, temperature, speed, inp_refs, sample_steps, if_sr), media_type="audio/"+media_type)
# --------------------------------
# 初始化部分
# --------------------------------
dict_language = {
"中文": "all_zh",
"粤语": "all_yue",
"英文": "en",
"日文": "all_ja",
"韩文": "all_ko",
"中英混合": "zh",
"粤英混合": "yue",
"日英混合": "ja",
"韩英混合": "ko",
"多语种混合": "auto", #多语种启动切分识别语种
"多语种混合(粤语)": "auto_yue",
"all_zh": "all_zh",
"all_yue": "all_yue",
"en": "en",
"all_ja": "all_ja",
"all_ko": "all_ko",
"zh": "zh",
"yue": "yue",
"ja": "ja",
"ko": "ko",
"auto": "auto",
"auto_yue": "auto_yue",
}
# logger
logging.config.dictConfig(uvicorn.config.LOGGING_CONFIG)
logger = logging.getLogger('uvicorn')
# 获取配置
g_config = global_config.Config()
# 获取参数
parser = argparse.ArgumentParser(description="GPT-SoVITS api")
parser.add_argument("-s", "--sovits_path", type=str, default=g_config.sovits_path, help="SoVITS模型路径")
parser.add_argument("-g", "--gpt_path", type=str, default=g_config.gpt_path, help="GPT模型路径")
parser.add_argument("-dr", "--default_refer_path", type=str, default="", help="默认参考音频路径")
parser.add_argument("-dt", "--default_refer_text", type=str, default="", help="默认参考音频文本")
parser.add_argument("-dl", "--default_refer_language", type=str, default="", help="默认参考音频语种")
parser.add_argument("-d", "--device", type=str, default=g_config.infer_device, help="cuda / cpu")
parser.add_argument("-a", "--bind_addr", type=str, default="0.0.0.0", help="default: 0.0.0.0")
parser.add_argument("-p", "--port", type=int, default=g_config.api_port, help="default: 9880")
parser.add_argument("-fp", "--full_precision", action="store_true", default=False, help="覆盖config.is_half为False, 使用全精度")
parser.add_argument("-hp", "--half_precision", action="store_true", default=False, help="覆盖config.is_half为True, 使用半精度")
# bool值的用法为 `python ./api.py -fp ...`
# 此时 full_precision==True, half_precision==False
parser.add_argument("-sm", "--stream_mode", type=str, default="close", help="流式返回模式, close / normal / keepalive")
parser.add_argument("-mt", "--media_type", type=str, default="wav", help="音频编码格式, wav / ogg / aac")
parser.add_argument("-st", "--sub_type", type=str, default="int16", help="音频数据类型, int16 / int32")
parser.add_argument("-cp", "--cut_punc", type=str, default="", help="文本切分符号设定, 符号范围,.;?!、，。？！；：…")
# 切割常用分句符为 `python ./api.py -cp ".?!。？！"`
parser.add_argument("-hb", "--hubert_path", type=str, default=g_config.cnhubert_path, help="覆盖config.cnhubert_path")
parser.add_argument("-b", "--bert_path", type=str, default=g_config.bert_path, help="覆盖config.bert_path")
args = parser.parse_args()
sovits_path = args.sovits_path
gpt_path = args.gpt_path
default_refer_path = args.default_refer_path
default_refer_text = args.default_refer_text
default_refer_language = args.default_refer_language
device = args.device
port = args.port
host = args.bind_addr
full_precision = args.full_precision
half_precision = args.half_precision
stream_mode = args.stream_mode
media_type = args.media_type
sub_type = args.sub_type
default_cut_punc = args.cut_punc
cnhubert_base_path = args.hubert_path
bert_path = args.bert_path
# 应用参数配置
default_refer = DefaultRefer(default_refer_path, default_refer_text, default_refer_language)
# 模型路径检查
if sovits_path == "":
sovits_path = g_config.pretrained_sovits_path
logger.warn(f"未指定SoVITS模型路径, fallback后当前值: {sovits_path}")
if gpt_path == "":
gpt_path = g_config.pretrained_gpt_path
logger.warn(f"未指定GPT模型路径, fallback后当前值: {gpt_path}")
# 指定默认参考音频, 调用方 未提供/未给全 参考音频参数时使用
if default_refer.path == "" or default_refer.text == "" or default_refer.language == "":
default_refer.path, default_refer.text, default_refer.language = "", "", ""
logger.info("未指定默认参考音频")
else:
logger.info(f"默认参考音频路径: {default_refer.path}")
logger.info(f"默认参考音频文本: {default_refer.text}")
logger.info(f"默认参考音频语种: {default_refer.language}")
# 获取半精度
is_half = g_config.is_half
if full_precision:
is_half = False
if half_precision:
is_half = True
if full_precision and half_precision:
is_half = g_config.is_half # 炒饭fallback
logger.info(f"半精: {is_half}")
# 流式返回模式
if stream_mode.lower() in ["normal","n"]:
stream_mode = "normal"
logger.info("流式返回已开启")
else:
stream_mode = "close"
# 音频编码格式
if media_type.lower() in ["aac","ogg"]:
media_type = media_type.lower()
elif stream_mode == "close":
media_type = "wav"
else:
media_type = "ogg"
logger.info(f"编码格式: {media_type}")
# 音频数据类型
if sub_type.lower() == 'int32':
is_int32 = True
logger.info(f"数据类型: int32")
else:
is_int32 = False
logger.info(f"数据类型: int16")
# 初始化模型
cnhubert.cnhubert_base_path = cnhubert_base_path
tokenizer = AutoTokenizer.from_pretrained(bert_path)
bert_model = AutoModelForMaskedLM.from_pretrained(bert_path)
ssl_model = cnhubert.get_model()
if is_half:
bert_model = bert_model.half().to(device)
ssl_model = ssl_model.half().to(device)
else:
bert_model = bert_model.to(device)
ssl_model = ssl_model.to(device)
change_gpt_sovits_weights(gpt_path = gpt_path, sovits_path = sovits_path)
# --------------------------------
# 接口部分
# --------------------------------
app = FastAPI()
@app.post("/set_model")
async def set_model(request: Request):
json_post_raw = await request.json()
return change_gpt_sovits_weights(
gpt_path = json_post_raw.get("gpt_model_path"),
sovits_path = json_post_raw.get("sovits_model_path")
)
@app.get("/set_model")
async def set_model(
gpt_model_path: str = None,
sovits_model_path: str = None,
):
return change_gpt_sovits_weights(gpt_path = gpt_model_path, sovits_path = sovits_model_path)
@app.post("/control")
async def control(request: Request):
json_post_raw = await request.json()
return handle_control(json_post_raw.get("command"))
@app.get("/control")
async def control(command: str = None):
return handle_control(command)
@app.post("/change_refer")
async def change_refer(request: Request):
json_post_raw = await request.json()
return handle_change(
json_post_raw.get("refer_wav_path"),
json_post_raw.get("prompt_text"),
json_post_raw.get("prompt_language")
)
@app.get("/change_refer")
async def change_refer(
refer_wav_path: str = None,
prompt_text: str = None,
prompt_language: str = None
):
return handle_change(refer_wav_path, prompt_text, prompt_language)
@app.post("/")
async def tts_endpoint(request: Request):
json_post_raw = await request.json()
return handle(
json_post_raw.get("refer_wav_path"),
json_post_raw.get("prompt_text"),
json_post_raw.get("prompt_language"),
json_post_raw.get("text"),
json_post_raw.get("text_language"),
json_post_raw.get("cut_punc"),
json_post_raw.get("top_k", 15),
json_post_raw.get("top_p", 1.0),
json_post_raw.get("temperature", 1.0),
json_post_raw.get("speed", 1.0),
json_post_raw.get("inp_refs", []),
json_post_raw.get("sample_steps", 32),
json_post_raw.get("if_sr", False)
)
@app.get("/")
async def tts_endpoint(
refer_wav_path: str = None,
prompt_text: str = None,
prompt_language: str = None,
text: str = None,
text_language: str = None,
cut_punc: str = None,
top_k: int = 15,
top_p: float = 1.0,
temperature: float = 1.0,
speed: float = 1.0,
inp_refs: list = Query(default=[]),
sample_steps: int = 32,
if_sr: bool = False
):
return handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cut_punc, top_k, top_p, temperature, speed, inp_refs, sample_steps, if_sr)
if __name__ == "__main__":
uvicorn.run(app, host=host, port=port, workers=1)
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