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import re
import yaml
from munch import Munch
import numpy as np
import librosa
import noisereduce as nr
from meldataset import TextCleaner
import torch
import torchaudio
from nltk.tokenize import word_tokenize
import nltk
nltk.download('punkt_tab')
from models import ProsodyPredictor, TextEncoder, StyleEncoder
from Modules.hifigan import Decoder
class Preprocess:
def __text_normalize(self, text):
punctuation = [",", "、", "،", ";", "(", ".", "。", "…", "!", "–", ":", "?"]
map_to = "."
punctuation_pattern = re.compile(f"[{''.join(re.escape(p) for p in punctuation)}]")
#replace punctuation that acts like a comma or period
text = punctuation_pattern.sub(map_to, text)
#replace consecutive whitespace chars with a single space and strip leading/trailing spaces
text = re.sub(r'\s+', ' ', text).strip()
return text
def __merge_fragments(self, texts, n):
merged = []
i = 0
while i < len(texts):
fragment = texts[i]
j = i + 1
while len(fragment.split()) < n and j < len(texts):
fragment += ", " + texts[j]
j += 1
merged.append(fragment)
i = j
if len(merged[-1].split()) < n and len(merged) > 1: #handle last sentence
merged[-2] = merged[-2] + ", " + merged[-1]
del merged[-1]
else:
merged[-1] = merged[-1]
return merged
def wave_preprocess(self, wave):
to_mel = torchaudio.transforms.MelSpectrogram(n_mels=80, n_fft=2048, win_length=1200, hop_length=300)
mean, std = -4, 4
wave_tensor = torch.from_numpy(wave).float()
mel_tensor = to_mel(wave_tensor)
mel_tensor = (torch.log(1e-5 + mel_tensor.unsqueeze(0)) - mean) / std
return mel_tensor
def text_preprocess(self, text, n_merge=12):
text_norm = self.__text_normalize(text).split(".")#split by sentences.
text_norm = [s.strip() for s in text_norm]
text_norm = list(filter(lambda x: x != '', text_norm)) #filter empty index
text_norm = self.__merge_fragments(text_norm, n=n_merge) #merge if a sentence has less that n
return text_norm
def length_to_mask(self, lengths):
mask = torch.arange(lengths.max()).unsqueeze(0).expand(lengths.shape[0], -1).type_as(lengths)
mask = torch.gt(mask+1, lengths.unsqueeze(1))
return mask
#For inference only
class StyleTTS2(torch.nn.Module):
def __init__(self, config_path, models_path):
super().__init__()
self.register_buffer("get_device", torch.empty(0))
self.preprocess = Preprocess()
self.ref_s = None
config = yaml.safe_load(open(config_path, "r", encoding="utf-8"))
try:
symbols = (
list(config['symbol']['pad']) +
list(config['symbol']['punctuation']) +
list(config['symbol']['letters']) +
list(config['symbol']['letters_ipa']) +
list(config['symbol']['extend'])
)
symbol_dict = {}
for i in range(len((symbols))):
symbol_dict[symbols[i]] = i
n_token = len(symbol_dict) + 1
print("\nFound:", n_token, "symbols")
except Exception as e:
print(f"\nERROR: Cannot find {e} in config file!\nYour config file is likely outdated, please download updated version from the repository.")
raise SystemExit(1)
args = self.__recursive_munch(config['model_params'])
args['n_token'] = n_token
self.cleaner = TextCleaner(symbol_dict, debug=False)
assert args.decoder.type in ['hifigan'], 'Decoder type unknown'
self.decoder = Decoder(dim_in=args.hidden_dim, style_dim=args.style_dim, dim_out=args.n_mels,
resblock_kernel_sizes = args.decoder.resblock_kernel_sizes,
upsample_rates = args.decoder.upsample_rates,
upsample_initial_channel=args.decoder.upsample_initial_channel,
resblock_dilation_sizes=args.decoder.resblock_dilation_sizes,
upsample_kernel_sizes=args.decoder.upsample_kernel_sizes)
self.predictor = ProsodyPredictor(style_dim=args.style_dim, d_hid=args.hidden_dim, nlayers=args.n_layer, max_dur=args.max_dur, dropout=args.dropout)
self.text_encoder = TextEncoder(channels=args.hidden_dim, kernel_size=5, depth=args.n_layer, n_symbols=args.n_token)
self.style_encoder = StyleEncoder(dim_in=args.dim_in, style_dim=args.style_dim, max_conv_dim=args.hidden_dim)# acoustic style encoder
self.__load_models(models_path)
def __recursive_munch(self, d):
if isinstance(d, dict):
return Munch((k, self.__recursive_munch(v)) for k, v in d.items())
elif isinstance(d, list):
return [self.__recursive_munch(v) for v in d]
else:
return d
def __replace_outliers_zscore(self, tensor, threshold=3.0, factor=0.95):
mean = tensor.mean()
std = tensor.std()
z = (tensor - mean) / std
# Identify outliers
outlier_mask = torch.abs(z) > threshold
# Compute replacement value, respecting sign
sign = torch.sign(tensor - mean)
replacement = mean + sign * (threshold * std * factor)
result = tensor.clone()
result[outlier_mask] = replacement[outlier_mask]
return result
def __load_models(self, models_path):
module_params = []
model = {'decoder':self.decoder, 'predictor':self.predictor, 'text_encoder':self.text_encoder, 'style_encoder':self.style_encoder}
params_whole = torch.load(models_path, map_location='cpu')
params = params_whole['net']
params = {key: value for key, value in params.items() if key in model.keys()}
for key in model:
try:
model[key].load_state_dict(params[key])
except:
from collections import OrderedDict
state_dict = params[key]
new_state_dict = OrderedDict()
for k, v in state_dict.items():
name = k[7:] # remove `module.`
new_state_dict[name] = v
model[key].load_state_dict(new_state_dict, strict=False)
total_params = sum(p.numel() for p in model[key].parameters())
print(key,":",total_params)
module_params.append(total_params)
print('\nTotal',":",sum(module_params))
def __compute_style(self, path, denoise, split_dur):
device = self.get_device.device
denoise = min(denoise, 1)
if split_dur != 0: split_dur = max(int(split_dur), 1)
max_samples = 24000*20 #max 20 seconds ref audio
print("Computing the style for:", path)
wave, sr = librosa.load(path, sr=24000)
audio, index = librosa.effects.trim(wave, top_db=30)
if sr != 24000:
audio = librosa.resample(audio, sr, 24000)
if len(audio) > max_samples:
audio = audio[:max_samples]
if denoise > 0.0:
audio_denoise = nr.reduce_noise(y=audio, sr=sr, n_fft=2048, win_length=1200, hop_length=300)
audio = audio*(1-denoise) + audio_denoise*denoise
with torch.no_grad():
if split_dur>0 and len(audio)/sr>=4: #Only effective if audio length is >= 4s
#This option will split the ref audio to multiple parts, calculate styles and average them
count = 0
ref_s = None
jump = sr*split_dur
total_len = len(audio)
#Need to init before the loop
mel_tensor = self.preprocess.wave_preprocess(audio[0:jump]).to(device)
ref_s = self.style_encoder(mel_tensor.unsqueeze(1))
count += 1
for i in range(jump, total_len, jump):
if i+jump >= total_len:
left_dur = (total_len-i)/sr
if left_dur >= 1: #Still count if left over dur is >= 1s
mel_tensor = self.preprocess.wave_preprocess(audio[i:total_len]).to(device)
ref_s += self.style_encoder(mel_tensor.unsqueeze(1))
count += 1
continue
mel_tensor = self.preprocess.wave_preprocess(audio[i:i+jump]).to(device)
ref_s += self.style_encoder(mel_tensor.unsqueeze(1))
count += 1
ref_s /= count
else:
mel_tensor = self.preprocess.wave_preprocess(audio).to(device)
ref_s = self.style_encoder(mel_tensor.unsqueeze(1))
return ref_s
def __inference(self, phonem, ref_s, speed=1, prev_d_mean=0, t=0.1):
device = self.get_device.device
speed = min(max(speed, 0.0001), 2) #speed range [0, 2]
phonem = ' '.join(word_tokenize(phonem))
tokens = self.cleaner(phonem)
tokens.insert(0, 0)
tokens.append(0)
tokens = torch.LongTensor(tokens).to(device).unsqueeze(0)
with torch.no_grad():
input_lengths = torch.LongTensor([tokens.shape[-1]]).to(device)
text_mask = self.preprocess.length_to_mask(input_lengths).to(device)
# encode
t_en = self.text_encoder(tokens, input_lengths, text_mask)
s = ref_s.to(device)
# cal alignment
d = self.predictor.text_encoder(t_en, s, input_lengths, text_mask)
x, _ = self.predictor.lstm(d)
duration = self.predictor.duration_proj(x)
duration = torch.sigmoid(duration).sum(axis=-1)
if prev_d_mean != 0:#Stabilize speaking speed between splits
dur_stats = torch.empty(duration.shape).normal_(mean=prev_d_mean, std=duration.std()).to(device)
else:
dur_stats = torch.empty(duration.shape).normal_(mean=duration.mean(), std=duration.std()).to(device)
duration = duration*(1-t) + dur_stats*t
duration[:,1:-2] = self.__replace_outliers_zscore(duration[:,1:-2]) #Normalize outlier
duration /= speed
pred_dur = torch.round(duration.squeeze()).clamp(min=1)
pred_aln_trg = torch.zeros(input_lengths, int(pred_dur.sum().data))
c_frame = 0
for i in range(pred_aln_trg.size(0)):
pred_aln_trg[i, c_frame:c_frame + int(pred_dur[i].data)] = 1
c_frame += int(pred_dur[i].data)
alignment = pred_aln_trg.unsqueeze(0).to(device)
# encode prosody
en = (d.transpose(-1, -2) @ alignment)
F0_pred, N_pred = self.predictor.F0Ntrain(en, s)
asr = (t_en @ pred_aln_trg.unsqueeze(0).to(device))
out = self.decoder(asr, F0_pred, N_pred, s)
return out.squeeze().cpu().numpy(), duration.mean()
def get_styles(self, speaker, denoise=0.3, avg_style=True, load_styles=False):
if not load_styles:
if avg_style: split_dur = 3
else: split_dur = 0
self.ref_s = self.__compute_style(speaker['path'], denoise=denoise, split_dur=split_dur)
else:
if self.ref_s is None:
raise Exception("Have to compute or load the styles first!")
style = {
'style': self.ref_s,
'path': speaker['path'],
'speed': speaker['speed'],
}
return style
def save_styles(self, save_dir):
if self.ref_s is not None:
torch.save(self.ref_s, save_dir)
print("Saved styles!")
else:
raise Exception("Have to compute the styles before saving it.")
def load_styles(self, save_dir):
try:
self.ref_s = torch.load(save_dir)
print("Loaded styles!")
except Exception as e:
print(e)
def generate(self, phonem, style, stabilize=True, n_merge=16):
if stabilize: smooth_value=0.2
else: smooth_value=0
list_wav = []
prev_d_mean = 0
print("Generating Audio...")
text_norm = self.preprocess.text_preprocess(phonem, n_merge=n_merge)
for sentence in text_norm:
wav, prev_d_mean = self.__inference(sentence, style['style'], speed=style['speed'], prev_d_mean=prev_d_mean, t=smooth_value)
wav = wav[4000:-4000] #Remove weird pulse and silent tokens
list_wav.append(wav)
final_wav = np.concatenate(list_wav)
final_wav = np.concatenate([np.zeros([4000]), final_wav, np.zeros([4000])], axis=0) # add padding
return final_wav |