update dev branch

This commit is contained in:
Leon 2024-07-20 05:29:03 +08:00
parent 2d2e3b0a07
commit 8f3d3ef006
2 changed files with 477 additions and 1 deletions

3
.gitignore vendored
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reference
GPT_weights
SoVITS_weights
TEMP
TEMP
outputs

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quick_inference.py Normal file
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import os, re, logging
import LangSegment
import pdb
import torch
import gradio as gr
from transformers import AutoModelForMaskedLM, AutoTokenizer
import numpy as np
import librosa
from feature_extractor import cnhubert
from module.models import SynthesizerTrn
from AR.models.t2s_lightning_module import Text2SemanticLightningModule
from text import cleaned_text_to_sequence
from text.cleaner import clean_text
from time import time as ttime
from module.mel_processing import spectrogram_torch
from tools.my_utils import load_audio
from tools.i18n.i18n import I18nAuto
import scipy.io.wavfile as wavfile
device = "cuda" if torch.cuda.is_available() else "cpu"
i18n = I18nAuto()
dict_language = {
i18n("中文"): "all_zh", # 全部按中文识别
i18n("英文"): "en", # 全部按英文识别#######不变
i18n("日文"): "all_ja", # 全部按日文识别
i18n("中英混合"): "zh", # 按中英混合识别####不变
i18n("日英混合"): "ja", # 按日英混合识别####不变
i18n("多语种混合"): "auto", # 多语种启动切分识别语种
}
is_share = os.environ.get("is_share", "False")
is_share = eval(is_share)
if "_CUDA_VISIBLE_DEVICES" in os.environ:
os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"]
half_precision = True
is_half = half_precision and torch.cuda.is_available()
splits = {"", "", "", "", ",", ".", "?", "!", "~", ":", "", "", "", }
punctuation = set(['!', '?', '', ',', '.', '-', " "])
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 replace_consecutive_punctuation(text):
punctuations = ''.join(re.escape(p) for p in punctuation)
pattern = f'([{punctuations}])([{punctuations}])+'
result = re.sub(pattern, r'\1', text)
return result
def get_first(text):
pattern = "[" + "".join(re.escape(sep) for sep in splits) + "]"
text = re.split(pattern, text)[0].strip()
return text
def split(todo_text):
todo_text = todo_text.replace("……", "").replace("——", "")
if todo_text[-1] not in splits:
todo_text += ""
i_split_head = i_split_tail = 0
len_text = len(todo_text)
todo_texts = []
while 1:
if i_split_head >= len_text:
break # 结尾一定有标点,所以直接跳出即可,最后一段在上次已加入
if todo_text[i_split_head] in splits:
i_split_head += 1
todo_texts.append(todo_text[i_split_tail:i_split_head])
i_split_tail = i_split_head
else:
i_split_head += 1
return todo_texts
# 四句一切
def cut1(inp):
inp = inp.strip("\n")
inps = split(inp)
split_idx = list(range(0, len(inps), 4))
split_idx[-1] = None
if len(split_idx) > 1:
opts = []
for idx in range(len(split_idx) - 1):
opts.append("".join(inps[split_idx[idx]: split_idx[idx + 1]]))
else:
opts = [inp]
opts = [item for item in opts if not set(item).issubset(punctuation)]
return "\n".join(opts)
# 句号切
def cut3(inp):
inp = inp.strip("\n")
opts = ["%s" % item for item in inp.strip("").split("")]
opts = [item for item in opts if not set(item).issubset(punctuation)]
return "\n".join(opts)
def process_text(texts):
_text = []
if all(text in [None, " ", "\n", ""] for text in texts):
raise ValueError(i18n("请输入有效文本"))
for text in texts:
if text in [None, " ", ""]:
pass
else:
_text.append(text)
return _text
def merge_short_text_in_array(texts, threshold):
if (len(texts)) < 2:
return texts
result = []
text = ""
for ele in texts:
text += ele
if len(text) >= threshold:
result.append(text)
text = ""
if (len(text) > 0):
if len(result) == 0:
result.append(text)
else:
result[len(result) - 1] += text
return result
def clean_text_inf(text, language):
phones, word2ph, norm_text = clean_text(text, language)
phones = cleaned_text_to_sequence(phones)
return phones, word2ph, norm_text
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)
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)
return phone_level_feature.T
dtype = torch.float16 if is_half else torch.float32
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 else torch.float32,
).to(device)
return bert
def get_spepc(hps, filename):
audio = load_audio(filename, int(hps.data.sampling_rate))
audio = torch.FloatTensor(audio)
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 get_phones_and_bert(text, language):
if language in {"en", "all_zh", "all_ja"}:
language = language.replace("all_", "")
if language == "en":
LangSegment.setfilters(["en"])
formattext = " ".join(tmp["text"] for tmp in LangSegment.getTexts(text))
else:
# 因无法区别中日文汉字,以用户输入为准
formattext = text
while " " in formattext:
formattext = formattext.replace(" ", " ")
phones, word2ph, norm_text = clean_text_inf(formattext, language)
if language == "zh":
bert = get_bert_feature(norm_text, word2ph).to(device)
else:
bert = torch.zeros(
(1024, len(phones)),
dtype=torch.float16 if is_half == True else torch.float32,
).to(device)
elif language in {"zh", "ja", "auto"}:
textlist = []
langlist = []
LangSegment.setfilters(["zh", "ja", "en", "ko"])
if language == "auto":
for tmp in LangSegment.getTexts(text):
if tmp["lang"] == "ko":
langlist.append("zh")
textlist.append(tmp["text"])
else:
langlist.append(tmp["lang"])
textlist.append(tmp["text"])
else:
for tmp in LangSegment.getTexts(text):
if tmp["lang"] == "en":
langlist.append(tmp["lang"])
else:
# 因无法区别中日文汉字,以用户输入为准
langlist.append(language)
textlist.append(tmp["text"])
print(textlist)
print(langlist)
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)
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)
return phones, bert.to(dtype), norm_text
def set_gpt_weights(gpt_path):
global hz, max_sec, t2s_model, config
hz = 50
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:
t2s_model = t2s_model.half()
t2s_model = t2s_model.to(device)
t2s_model.eval()
total = sum([param.nelement() for param in t2s_model.parameters()])
print("Number of parameter: %.2fM" % (total / 1e6))
with open("./gweight.txt", "w", encoding="utf-8") as f: f.write(gpt_path)
def set_sovits_weights(sovits_path):
global vq_model, hps
dict_s2 = torch.load(sovits_path, map_location="cpu")
hps = dict_s2["config"]
hps = DictToAttrRecursive(hps)
hps.model.semantic_frame_rate = "25hz"
vq_model = SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**hps.model
)
if "pretrained" not in sovits_path:
del vq_model.enc_q
if is_half:
vq_model = vq_model.half().to(device)
else:
vq_model = vq_model.to(device)
vq_model.eval()
print(vq_model.load_state_dict(dict_s2["weight"], strict=False))
with open("./sweight.txt", "w", encoding="utf-8") as f:
f.write(sovits_path)
def gen_audio(ref_wav_path, prompt_text, text_to_speak, output_file, top_k=20, top_p=0.6, temperature=0.6, ref_free=False):
if prompt_text is None or len(prompt_text) == 0:
ref_free = True
t0 = ttime()
prompt_language = "zh"
text_language = "zh"
if not ref_free:
prompt_text = prompt_text.strip("\n")
if prompt_text[-1] not in splits:
prompt_text += "" if prompt_language != "en" else "."
print(i18n("实际输入的参考文本:"), prompt_text)
text_to_speak = text_to_speak.strip("\n")
text_to_speak = replace_consecutive_punctuation(text_to_speak)
if text_to_speak[0] not in splits and len(get_first(text_to_speak)) < 4:
text_to_speak = "" + text_to_speak if text_language != "en" else "." + text_to_speak
print(i18n("实际输入的目标文本:"), text_to_speak)
zero_wav = np.zeros(
int(hps.data.sampling_rate * 0.3),
dtype=np.float16 if is_half == True else np.float32,
)
if not ref_free:
with torch.no_grad():
wav16k, sr = librosa.load(ref_wav_path, sr=16000)
if wav16k.shape[0] > 160000 or wav16k.shape[0] < 48000:
raise OSError(i18n("参考音频在3~10秒范围外请更换"))
wav16k = torch.from_numpy(wav16k)
zero_wav_torch = torch.from_numpy(zero_wav)
if is_half:
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)
t1 = ttime()
# text_to_speak = cut1(text_to_speak)
text_to_speak = cut3(text_to_speak)
while "\n\n" in text_to_speak:
text_to_speak = text_to_speak.replace("\n\n", "\n")
print(i18n("实际输入的目标文本(切句后):"), text_to_speak)
texts = text_to_speak.split("\n")
texts = process_text(texts)
texts = merge_short_text_in_array(texts, 5)
audio_opt = []
if not ref_free:
phones1, bert1, norm_text1 = get_phones_and_bert(prompt_text, prompt_language)
for text_to_speak in texts:
# 解决输入目标文本的空行导致报错的问题
if len(text_to_speak.strip()) == 0:
continue
if text_to_speak[-1] not in splits:
text_to_speak += "" if text_language != "en" else "."
print(i18n("实际输入的目标文本(每句):"), text_to_speak)
phones2, bert2, norm_text2 = get_phones_and_bert(text_to_speak, text_language)
print(i18n("前端处理后的文本(每句):"), norm_text2)
if not ref_free:
bert = torch.cat([bert1, bert2], 1)
all_phoneme_ids = torch.LongTensor(phones1 + phones2).to(device).unsqueeze(0)
else:
bert = bert2
all_phoneme_ids = torch.LongTensor(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 = t2s_model.model.infer(
pred_semantic, idx = t2s_model.model.infer_panel(
all_phoneme_ids,
all_phoneme_len,
None if ref_free else prompt,
bert,
# prompt_phone_len=ph_offset,
top_k=top_k,
top_p=top_p,
temperature=temperature,
early_stop_num=hz * max_sec,
)
t3 = ttime()
# print(pred_semantic.shape,idx)
pred_semantic = pred_semantic[:, -idx:].unsqueeze(
0
) # .unsqueeze(0)#mq要多unsqueeze一次
refer = get_spepc(hps, ref_wav_path) # .to(device)
if is_half:
refer = refer.half().to(device)
else:
refer = refer.to(device)
# audio = vq_model.decode(pred_semantic, all_phoneme_ids, refer).detach().cpu().numpy()[0, 0]
audio = (
vq_model.decode(
pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer
)
.detach()
.cpu()
.numpy()[0, 0]
) # 试试重建不带上prompt部分
max_audio = np.abs(audio).max() # 简单防止16bit爆音
if max_audio > 1:
audio /= max_audio
audio_opt.append(audio)
audio_opt.append(zero_wav)
t4 = ttime()
# 将音频数据合并
audio_data = np.concatenate(audio_opt, 0) * 32768
audio_data = audio_data.astype(np.int16)
wavfile.write(output_file, hps.data.sampling_rate, audio_data)
print("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3))
cnhubert_base_path = os.environ.get(
"cnhubert_base_path", "GPT_SoVITS/pretrained_models/chinese-hubert-base"
)
bert_path = os.environ.get(
"bert_path", "GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large"
)
cnhubert.cnhubert_base_path = cnhubert_base_path
ssl_model = cnhubert.get_model()
if is_half:
ssl_model = ssl_model.half().to(device)
else:
ssl_model = ssl_model.to(device)
tokenizer = AutoTokenizer.from_pretrained(bert_path)
bert_model = AutoModelForMaskedLM.from_pretrained(bert_path)
if is_half:
bert_model = bert_model.half().to(device)
else:
bert_model = bert_model.to(device)
def speak(text_to_speak):
sovits_path = "SoVITS_weights/阿贝多_e12_s2748.pth"
set_sovits_weights(sovits_path)
gpt_path = "GPT_weights/阿贝多-e10.ckpt"
set_gpt_weights(gpt_path)
ref_wav_path = "audio/首先,先看看这不明来源的元素力,究竟是如何对外流动的.wav"
prompt_text = "首先,先看看这不明来源的元素力,究竟是如何对外流动的。"
# text_to_speak = "我...我...我不知道你在说什么,我们之间没有秘密呀。可能你弄错了,我们平时关系很好的,请不要误会。"
# 创建一个时间戳的文件名
output_file = "outputs/" + str(int(ttime())) + ".wav"
gen_audio(ref_wav_path, prompt_text, text_to_speak, output_file)
return output_file
def main():
speak("放学了,我该回家了,你叫我留下来干什么?")
if __name__ == '__main__':
main()