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import os
import librosa
import commons
import sys
import re
import numpy as np
import torch
import xml.etree.ElementTree as ET
import config
import logging
from torch import no_grad, LongTensor, inference_mode, FloatTensor
from io import BytesIO
from graiax import silkcoder
from utils.nlp import cut, sentence_split
from scipy.io.wavfile import write
from mel_processing import spectrogram_torch
from text import text_to_sequence, _clean_text
from models import SynthesizerTrn
from utils import utils
# torch.set_num_threads(1) # 设置torch线程为1
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
class vits:
def __init__(self, model, config, model_=None, model_type=None):
self.model_type = model_type
self.hps_ms = utils.get_hparams_from_file(config)
self.n_speakers = self.hps_ms.data.n_speakers if 'n_speakers' in self.hps_ms.data.keys() else 0
self.n_symbols = len(self.hps_ms.symbols) if 'symbols' in self.hps_ms.keys() else 0
self.speakers = self.hps_ms.speakers if 'speakers' in self.hps_ms.keys() else ['0']
self.use_f0 = self.hps_ms.data.use_f0 if 'use_f0' in self.hps_ms.data.keys() else False
self.emotion_embedding = self.hps_ms.data.emotion_embedding if 'emotion_embedding' in self.hps_ms.data.keys() else False
self.net_g_ms = SynthesizerTrn(
self.n_symbols,
self.hps_ms.data.filter_length // 2 + 1,
self.hps_ms.train.segment_size // self.hps_ms.data.hop_length,
n_speakers=self.n_speakers,
emotion_embedding=self.emotion_embedding,
**self.hps_ms.model)
_ = self.net_g_ms.eval()
# load model
self.load_model(model, model_)
def load_model(self, model, model_=None):
utils.load_checkpoint(model, self.net_g_ms)
self.net_g_ms.to(device)
if self.model_type == "hubert":
self.hubert = model_
elif self.model_type == "w2v2":
self.emotion_reference = model_
def get_cleaned_text(self, text, hps, cleaned=False):
if cleaned:
text_norm = text_to_sequence(text, hps.symbols, [])
else:
text_norm = text_to_sequence(text, hps.symbols, hps.data.text_cleaners)
if hps.data.add_blank:
text_norm = commons.intersperse(text_norm, 0)
text_norm = LongTensor(text_norm)
return text_norm
def get_label_value(self, label, default, warning_name='value', text=""):
value = re.search(rf'\[{label}=(.+?)\]', text)
if value:
try:
text = re.sub(rf'\[{label}=(.+?)\]', '', text, 1)
value = float(value.group(1))
except:
print(f'Invalid {warning_name}!')
sys.exit(1)
else:
value = default
if text == "":
return value
else:
return value, text
def get_label(self, text, label):
if f'[{label}]' in text:
return True, text.replace(f'[{label}]', '')
else:
return False, text
def get_cleaner(self):
return getattr(self.hps_ms.data, 'text_cleaners', [None])[0]
def return_speakers(self, escape=False):
return self.speakers
def infer(self, params):
emotion = params.get("emotion", None)
with no_grad():
x_tst = params.get("stn_tst").unsqueeze(0)
x_tst_lengths = LongTensor([params.get("stn_tst").size(0)])
audio = self.net_g_ms.infer(x_tst.to(device), x_tst_lengths.to(device), sid=params.get("sid").to(device),
noise_scale=params.get("noise_scale"),
noise_scale_w=params.get("noise_scale_w"),
length_scale=params.get("length_scale"),
emotion_embedding=emotion.to(device) if emotion != None else None)[0][
0, 0].data.float().cpu().numpy()
torch.cuda.empty_cache()
return audio
def get_infer_param(self, length, noise, noisew, text=None, speaker_id=None, audio_path=None,
emotion=None):
emo = None
if self.model_type != "hubert":
length_scale, text = self.get_label_value('LENGTH', length, 'length scale', text)
noise_scale, text = self.get_label_value('NOISE', noise, 'noise scale', text)
noise_scale_w, text = self.get_label_value('NOISEW', noisew, 'deviation of noise', text)
cleaned, text = self.get_label(text, 'CLEANED')
stn_tst = self.get_cleaned_text(text, self.hps_ms, cleaned=cleaned)
sid = LongTensor([speaker_id])
if self.model_type == "w2v2":
# if emotion_reference.endswith('.npy'):
# emotion = np.load(emotion_reference)
# emotion = FloatTensor(emotion).unsqueeze(0)
# else:
# audio16000, sampling_rate = librosa.load(
# emotion_reference, sr=16000, mono=True)
# emotion = self.w2v2(audio16000, sampling_rate)[
# 'hidden_states']
# emotion_reference = re.sub(
# r'\..*$', '', emotion_reference)
# np.save(emotion_reference, emotion.squeeze(0))
# emotion = FloatTensor(emotion)
emo = torch.FloatTensor(self.emotion_reference[emotion]).unsqueeze(0)
elif self.model_type == "hubert":
if self.use_f0:
audio, sampling_rate = librosa.load(
audio_path, sr=self.hps_ms.data.sampling_rate, mono=True)
audio16000 = librosa.resample(
audio, orig_sr=sampling_rate, target_sr=16000)
else:
audio16000, sampling_rate = librosa.load(
audio_path, sr=16000, mono=True)
length_scale = self.get_label_value('LENGTH', length, 'length scale')
noise_scale = self.get_label_value('NOISE', noise, 'noise scale')
noise_scale_w = self.get_label_value('NOISEW', noisew, 'deviation of noise')
with inference_mode():
units = self.hubert.units(FloatTensor(audio16000).unsqueeze(0).unsqueeze(0)).squeeze(0).numpy()
if self.use_f0:
f0_scale = self.get_label_value('F0', 1, 'f0 scale')
f0 = librosa.pyin(audio,
sr=sampling_rate,
fmin=librosa.note_to_hz('C0'),
fmax=librosa.note_to_hz('C7'),
frame_length=1780)[0]
target_length = len(units[:, 0])
f0 = np.nan_to_num(np.interp(np.arange(0, len(f0) * target_length, len(f0)) / target_length,
np.arange(0, len(f0)), f0)) * f0_scale
units[:, 0] = f0 / 10
stn_tst = FloatTensor(units)
sid = LongTensor([speaker_id])
params = {"length_scale": length_scale, "noise_scale": noise_scale,
"noise_scale_w": noise_scale_w, "stn_tst": stn_tst,
"sid": sid, "emotion": emo}
return params
def get_audio(self, voice, auto_break=False):
text = voice.get("text", None)
speaker_id = voice.get("id", 0)
length = voice.get("length", 1)
noise = voice.get("noise", 0.667)
noisew = voice.get("noisew", 0.8)
max = voice.get("max", 50)
lang = voice.get("lang", "auto")
speaker_lang = voice.get("speaker_lang", None)
audio_path = voice.get("audio_path", None)
emotion = voice.get("emotion", 0)
# 去除所有多余的空白字符
if text is not None: text = re.sub(r'\s+', ' ', text).strip()
# 停顿0.75s,避免语音分段合成再拼接后的连接突兀
brk = np.zeros(int(0.75 * 22050), dtype=np.int16)
tasks = []
if self.model_type == "vits":
sentence_list = sentence_split(text, max, lang, speaker_lang)
for sentence in sentence_list:
tasks.append(
self.get_infer_param(text=sentence, speaker_id=speaker_id, length=length, noise=noise,
noisew=noisew))
audios = []
for task in tasks:
audios.append(self.infer(task))
if auto_break:
audios.append(brk)
audio = np.concatenate(audios, axis=0)
elif self.model_type == "hubert":
params = self.get_infer_param(speaker_id=speaker_id, length=length, noise=noise, noisew=noisew,
audio_path=audio_path)
audio = self.infer(params)
elif self.model_type == "w2v2":
sentence_list = sentence_split(text, max, lang, speaker_lang)
for sentence in sentence_list:
tasks.append(
self.get_infer_param(text=sentence, speaker_id=speaker_id, length=length, noise=noise,
noisew=noisew, emotion=emotion))
audios = []
for task in tasks:
audios.append(self.infer(task))
if auto_break:
audios.append(brk)
audio = np.concatenate(audios, axis=0)
return audio
def voice_conversion(self, voice):
audio_path = voice.get("audio_path")
original_id = voice.get("original_id")
target_id = voice.get("target_id")
audio = utils.load_audio_to_torch(
audio_path, self.hps_ms.data.sampling_rate)
y = audio.unsqueeze(0)
spec = spectrogram_torch(y, self.hps_ms.data.filter_length,
self.hps_ms.data.sampling_rate, self.hps_ms.data.hop_length,
self.hps_ms.data.win_length,
center=False)
spec_lengths = LongTensor([spec.size(-1)])
sid_src = LongTensor([original_id])
with no_grad():
sid_tgt = LongTensor([target_id])
audio = self.net_g_ms.voice_conversion(spec.to(device),
spec_lengths.to(device),
sid_src=sid_src.to(device),
sid_tgt=sid_tgt.to(device))[0][0, 0].data.cpu().float().numpy()
torch.cuda.empty_cache()
return audio
class TTS:
def __init__(self, voice_obj, voice_speakers):
self._voice_obj = voice_obj
self._voice_speakers = voice_speakers
self._strength_dict = {"x-weak": 0.25, "weak": 0.5, "Medium": 0.75, "Strong": 1, "x-strong": 1.25}
self._speakers_count = sum([len(self._voice_speakers[i]) for i in self._voice_speakers])
self._vits_speakers_count = len(self._voice_speakers["VITS"])
self._hubert_speakers_count = len(self._voice_speakers["HUBERT-VITS"])
self._w2v2_speakers_count = len(self._voice_speakers["W2V2-VITS"])
self.dem = None
if getattr(config, "DIMENSIONAL_EMOTION_MODEL", None) != None:
try:
import audonnx
root = os.path.dirname(config.DIMENSIONAL_EMOTION_MODEL)
model_file = config.DIMENSIONAL_EMOTION_MODEL
self.dem = audonnx.load(root=root, model_file=model_file)
except Exception as e:
self.logger.warning(f"Load DIMENSIONAL_EMOTION_MODEL failed {e}")
# Initialization information
self.logger = logging.getLogger("vits-simple-api")
self.logger.info(f"torch:{torch.__version__} cuda_available:{torch.cuda.is_available()}")
self.logger.info(f'device:{device} device.type:{device.type}')
if self._vits_speakers_count != 0: self.logger.info(f"[VITS] {self._vits_speakers_count} speakers")
if self._hubert_speakers_count != 0: self.logger.info(f"[hubert] {self._hubert_speakers_count} speakers")
if self._w2v2_speakers_count != 0: self.logger.info(f"[w2v2] {self._w2v2_speakers_count} speakers")
self.logger.info(f"{self._speakers_count} speakers in total")
if self._speakers_count == 0:
self.logger.warning(f"No model was loaded")
@property
def voice_speakers(self):
return self._voice_speakers
@property
def speakers_count(self):
return self._speakers_count
@property
def vits_speakers_count(self):
return self._vits_speakers_count
@property
def hubert_speakers_count(self):
return self._hubert_speakers_count
@property
def w2v2_speakers_count(self):
return self._w2v2_speakers_count
def encode(self, sampling_rate, audio, format):
with BytesIO() as f:
write(f, sampling_rate, audio)
if format.upper() == 'OGG':
with BytesIO() as o:
utils.wav2ogg(f, o)
return BytesIO(o.getvalue())
elif format.upper() == 'SILK':
return BytesIO(silkcoder.encode(f))
elif format.upper() == 'MP3':
with BytesIO() as o:
utils.wav2mp3(f, o)
return BytesIO(o.getvalue())
elif format.upper() == 'WAV':
return BytesIO(f.getvalue())
def convert_time_string(self, time_string):
time_value = float(re.findall(r'\d+\.?\d*', time_string)[0])
time_unit = re.findall(r'[a-zA-Z]+', time_string)[0].lower()
if time_unit.upper() == 'MS':
return time_value / 1000
elif time_unit.upper() == 'S':
return time_value
elif time_unit.upper() == 'MIN':
return time_value * 60
elif time_unit.upper() == 'H':
return time_value * 3600
elif time_unit.upper() == 'D':
return time_value * 24 * 3600 # 不会有人真写D吧?
else:
raise ValueError("Unsupported time unit: {}".format(time_unit))
def parse_ssml(self, ssml):
root = ET.fromstring(ssml)
format = root.attrib.get("format", "wav")
voice_tasks = []
brk_count = 0
strength_dict = {"x-weak": 0.25, "weak": 0.5, "Medium": 0.75, "Strong": 1, "x-strong": 1.25}
for element in root.iter():
if element.tag == "voice":
id = int(element.attrib.get("id", root.attrib.get("id", config.ID)))
lang = element.attrib.get("lang", root.attrib.get("lang", config.LANG))
length = float(element.attrib.get("length", root.attrib.get("length", config.LENGTH)))
noise = float(element.attrib.get("noise", root.attrib.get("noise", config.NOISE)))
noisew = float(element.attrib.get("noisew", root.attrib.get("noisew", config.NOISEW)))
max = int(element.attrib.get("max", root.attrib.get("max", "0")))
# 不填写默认就是vits
model = element.attrib.get("model", root.attrib.get("model", "vits"))
# w2v2-vits/emotion-vits才有emotion
emotion = int(element.attrib.get("emotion", root.attrib.get("emotion", 0)))
voice_element = ET.tostring(element, encoding='unicode')
pattern_voice = r'<voice.*?>(.*?)</voice>'
pattern_break = r'<break\s*?(.*?)\s*?/>'
matches_voice = re.findall(pattern_voice, voice_element)[0]
matches_break = re.split(pattern_break, matches_voice)
for match in matches_break:
strength = re.search(r'\s*strength\s*=\s*[\'\"](.*?)[\'\"]', match)
time = re.search(r'\s*time\s*=\s*[\'\"](.*?)[\'\"]', match)
# break标签 strength属性
if strength:
brk = strength_dict[strength.group(1)]
voice_tasks.append({"break": brk})
brk_count += 1
# break标签 time属性
elif time:
brk = self.convert_time_string(time.group(1))
voice_tasks.append({"break": brk})
brk_count += 1
# break标签 为空说明只写了break,默认停顿0.75s
elif match == "":
voice_tasks.append({"break": 0.75})
brk_count += 1
# voice标签中除了break剩下的就是文本
else:
voice_tasks.append({"id": id,
"text": match,
"lang": lang,
"length": length,
"noise": noise,
"noisew": noisew,
"max": max,
"model": model,
"emotion": emotion
})
# 分段末尾停顿0.75s
voice_tasks.append({"break": 0.75})
elif element.tag == "break":
# brk_count大于0说明voice标签中有break
if brk_count > 0:
brk_count -= 1
continue
brk = strength_dict.get(element.attrib.get("strength"),
self.convert_time_string(element.attrib.get("time", "750ms")))
voice_tasks.append({"break": brk})
for i in voice_tasks:
self.logger.debug(i)
return voice_tasks, format
def create_ssml_infer_task(self, ssml):
voice_tasks, format = self.parse_ssml(ssml)
audios = []
for voice in voice_tasks:
if voice.get("break"):
audios.append(np.zeros(int(voice.get("break") * 22050), dtype=np.int16))
else:
model = voice.get("model").upper()
if model != "VITS" and model != "W2V2-VITS" and model != "EMOTION-VITS":
raise ValueError(f"Unsupported model: {voice.get('model')}")
voice_obj = self._voice_obj[model][voice.get("id")][1]
voice["id"] = self._voice_obj[model][voice.get("id")][0]
audios.append(voice_obj.get_audio(voice))
audio = np.concatenate(audios, axis=0)
return self.encode(voice_obj.hps_ms.data.sampling_rate, audio, format), format
def vits_infer(self, voice):
format = voice.get("format", "wav")
voice_obj = self._voice_obj["VITS"][voice.get("id")][1]
voice["id"] = self._voice_obj["VITS"][voice.get("id")][0]
audio = voice_obj.get_audio(voice, auto_break=True)
return self.encode(voice_obj.hps_ms.data.sampling_rate, audio, format)
def hubert_vits_infer(self, voice):
format = voice.get("format", "wav")
voice_obj = self._voice_obj["HUBERT-VITS"][voice.get("id")][1]
voice["id"] = self._voice_obj["HUBERT-VITS"][voice.get("id")][0]
audio = voice_obj.get_audio(voice)
return self.encode(voice_obj.hps_ms.data.sampling_rate, audio, format)
def w2v2_vits_infer(self, voice):
format = voice.get("format", "wav")
voice_obj = self._voice_obj["W2V2-VITS"][voice.get("id")][1]
voice["id"] = self._voice_obj["W2V2-VITS"][voice.get("id")][0]
audio = voice_obj.get_audio(voice, auto_break=True)
return self.encode(voice_obj.hps_ms.data.sampling_rate, audio, format)
def vits_voice_conversion(self, voice):
original_id = voice.get("original_id")
target_id = voice.get("target_id")
format = voice.get("format")
original_id_obj = int(self._voice_obj["VITS"][original_id][2])
target_id_obj = int(self._voice_obj["VITS"][target_id][2])
if original_id_obj != target_id_obj:
raise ValueError(f"speakers are in diffrent VITS Model")
voice["original_id"] = int(self._voice_obj["VITS"][original_id][0])
voice["target_id"] = int(self._voice_obj["VITS"][target_id][0])
voice_obj = self._voice_obj["VITS"][original_id][1]
audio = voice_obj.voice_conversion(voice)
return self.encode(voice_obj.hps_ms.data.sampling_rate, audio, format)
def get_dimensional_emotion_npy(self, audio):
if self.dem is None:
raise ValueError(f"Please configure DIMENSIONAL_EMOTION_MODEL path in config.py")
audio16000, sampling_rate = librosa.load(audio, sr=16000, mono=True)
emotion = self.dem(audio16000, sampling_rate)['hidden_states']
emotion_npy = BytesIO()
np.save(emotion_npy, emotion.squeeze(0))
emotion_npy.seek(0)
return emotion_npy
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