|
import os
|
|
import re
|
|
import json
|
|
import torch
|
|
import librosa
|
|
import soundfile
|
|
import torchaudio
|
|
import numpy as np
|
|
import torch.nn as nn
|
|
from tqdm import tqdm
|
|
import torch
|
|
|
|
from . import utils
|
|
from . import commons
|
|
from .models import SynthesizerTrn
|
|
from .split_utils import split_sentence
|
|
from .mel_processing import spectrogram_torch, spectrogram_torch_conv
|
|
from .download_utils import load_or_download_config, load_or_download_model
|
|
|
|
class TTS(nn.Module):
|
|
def __init__(self,
|
|
language,
|
|
device='auto',
|
|
use_hf=True,
|
|
config_path=None,
|
|
ckpt_path=None):
|
|
super().__init__()
|
|
if device == 'auto':
|
|
device = 'cpu'
|
|
if torch.cuda.is_available(): device = 'cuda'
|
|
if torch.backends.mps.is_available(): device = 'mps'
|
|
if 'cuda' in device:
|
|
assert torch.cuda.is_available()
|
|
|
|
|
|
hps = load_or_download_config(language, use_hf=use_hf, config_path=config_path)
|
|
|
|
num_languages = hps.num_languages
|
|
num_tones = hps.num_tones
|
|
symbols = hps.symbols
|
|
|
|
model = SynthesizerTrn(
|
|
len(symbols),
|
|
hps.data.filter_length // 2 + 1,
|
|
hps.train.segment_size // hps.data.hop_length,
|
|
n_speakers=hps.data.n_speakers,
|
|
num_tones=num_tones,
|
|
num_languages=num_languages,
|
|
**hps.model,
|
|
).to(device)
|
|
|
|
model.eval()
|
|
self.model = model
|
|
self.symbol_to_id = {s: i for i, s in enumerate(symbols)}
|
|
self.hps = hps
|
|
self.device = device
|
|
|
|
|
|
checkpoint_dict = load_or_download_model(language, device, use_hf=use_hf, ckpt_path=ckpt_path)
|
|
self.model.load_state_dict(checkpoint_dict['model'], strict=True)
|
|
|
|
language = language.split('_')[0]
|
|
self.language = 'ZH_MIX_EN' if language == 'ZH' else language
|
|
|
|
@staticmethod
|
|
def audio_numpy_concat(segment_data_list, sr, speed=1.):
|
|
audio_segments = []
|
|
for segment_data in segment_data_list:
|
|
audio_segments += segment_data.reshape(-1).tolist()
|
|
audio_segments += [0] * int((sr * 0.05) / speed)
|
|
audio_segments = np.array(audio_segments).astype(np.float32)
|
|
return audio_segments
|
|
|
|
@staticmethod
|
|
def split_sentences_into_pieces(text, language, quiet=False):
|
|
texts = split_sentence(text, language_str=language)
|
|
if not quiet:
|
|
print(" > Text split to sentences.")
|
|
print('\n'.join(texts))
|
|
print(" > ===========================")
|
|
return texts
|
|
|
|
def tts_to_file(self, text, speaker_id, output_path=None, sdp_ratio=0.2, noise_scale=0.6, noise_scale_w=0.8, speed=1.0, pbar=None, format=None, position=None, quiet=False,):
|
|
language = self.language
|
|
texts = self.split_sentences_into_pieces(text, language, quiet)
|
|
audio_list = []
|
|
if pbar:
|
|
tx = pbar(texts)
|
|
else:
|
|
if position:
|
|
tx = tqdm(texts, position=position)
|
|
elif quiet:
|
|
tx = texts
|
|
else:
|
|
tx = tqdm(texts)
|
|
for t in tx:
|
|
if language in ['EN', 'ZH_MIX_EN']:
|
|
t = re.sub(r'([a-z])([A-Z])', r'\1 \2', t)
|
|
device = self.device
|
|
bert, ja_bert, phones, tones, lang_ids = utils.get_text_for_tts_infer(t, language, self.hps, device, self.symbol_to_id)
|
|
with torch.no_grad():
|
|
x_tst = phones.to(device).unsqueeze(0)
|
|
tones = tones.to(device).unsqueeze(0)
|
|
lang_ids = lang_ids.to(device).unsqueeze(0)
|
|
bert = bert.to(device).unsqueeze(0)
|
|
ja_bert = ja_bert.to(device).unsqueeze(0)
|
|
x_tst_lengths = torch.LongTensor([phones.size(0)]).to(device)
|
|
del phones
|
|
speakers = torch.LongTensor([speaker_id]).to(device)
|
|
audio = self.model.infer(
|
|
x_tst,
|
|
x_tst_lengths,
|
|
speakers,
|
|
tones,
|
|
lang_ids,
|
|
bert,
|
|
ja_bert,
|
|
sdp_ratio=sdp_ratio,
|
|
noise_scale=noise_scale,
|
|
noise_scale_w=noise_scale_w,
|
|
length_scale=1. / speed,
|
|
)[0][0, 0].data.cpu().float().numpy()
|
|
del x_tst, tones, lang_ids, bert, ja_bert, x_tst_lengths, speakers
|
|
|
|
audio_list.append(audio)
|
|
torch.cuda.empty_cache()
|
|
audio = self.audio_numpy_concat(audio_list, sr=self.hps.data.sampling_rate, speed=speed)
|
|
|
|
if output_path is None:
|
|
return audio
|
|
else:
|
|
if format:
|
|
soundfile.write(output_path, audio, self.hps.data.sampling_rate, format=format)
|
|
else:
|
|
soundfile.write(output_path, audio, self.hps.data.sampling_rate)
|
|
|