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import glob
import re
import librosa
import torch
import yaml
from sklearn.preprocessing import StandardScaler
from torch import nn
from modules.parallel_wavegan.models import ParallelWaveGANGenerator
from modules.parallel_wavegan.utils import read_hdf5
from utils.hparams import hparams
from utils.pitch_utils import f0_to_coarse
from network.vocoders.base_vocoder import BaseVocoder, register_vocoder
import numpy as np
def load_pwg_model(config_path, checkpoint_path, stats_path):
# load config
with open(config_path, encoding='utf-8') as f:
config = yaml.load(f, Loader=yaml.Loader)
# setup
if torch.cuda.is_available():
device = torch.device("cuda")
else:
device = torch.device("cpu")
model = ParallelWaveGANGenerator(**config["generator_params"])
ckpt_dict = torch.load(checkpoint_path, map_location="cpu")
if 'state_dict' not in ckpt_dict: # official vocoder
model.load_state_dict(torch.load(checkpoint_path, map_location="cpu")["model"]["generator"])
scaler = StandardScaler()
if config["format"] == "hdf5":
scaler.mean_ = read_hdf5(stats_path, "mean")
scaler.scale_ = read_hdf5(stats_path, "scale")
elif config["format"] == "npy":
scaler.mean_ = np.load(stats_path)[0]
scaler.scale_ = np.load(stats_path)[1]
else:
raise ValueError("support only hdf5 or npy format.")
else: # custom PWG vocoder
fake_task = nn.Module()
fake_task.model_gen = model
fake_task.load_state_dict(torch.load(checkpoint_path, map_location="cpu")["state_dict"], strict=False)
scaler = None
model.remove_weight_norm()
model = model.eval().to(device)
print(f"| Loaded model parameters from {checkpoint_path}.")
print(f"| PWG device: {device}.")
return model, scaler, config, device
@register_vocoder
class PWG(BaseVocoder):
def __init__(self):
if hparams['vocoder_ckpt'] == '': # load LJSpeech PWG pretrained model
base_dir = 'wavegan_pretrained'
ckpts = glob.glob(f'{base_dir}/checkpoint-*steps.pkl')
ckpt = sorted(ckpts, key=
lambda x: int(re.findall(f'{base_dir}/checkpoint-(\d+)steps.pkl', x)[0]))[-1]
config_path = f'{base_dir}/config.yaml'
print('| load PWG: ', ckpt)
self.model, self.scaler, self.config, self.device = load_pwg_model(
config_path=config_path,
checkpoint_path=ckpt,
stats_path=f'{base_dir}/stats.h5',
)
else:
base_dir = hparams['vocoder_ckpt']
print(base_dir)
config_path = f'{base_dir}/config.yaml'
ckpt = sorted(glob.glob(f'{base_dir}/model_ckpt_steps_*.ckpt'), key=
lambda x: int(re.findall(f'{base_dir}/model_ckpt_steps_(\d+).ckpt', x)[0]))[-1]
print('| load PWG: ', ckpt)
self.scaler = None
self.model, _, self.config, self.device = load_pwg_model(
config_path=config_path,
checkpoint_path=ckpt,
stats_path=f'{base_dir}/stats.h5',
)
def spec2wav(self, mel, **kwargs):
# start generation
config = self.config
device = self.device
pad_size = (config["generator_params"]["aux_context_window"],
config["generator_params"]["aux_context_window"])
c = mel
if self.scaler is not None:
c = self.scaler.transform(c)
with torch.no_grad():
z = torch.randn(1, 1, c.shape[0] * config["hop_size"]).to(device)
c = np.pad(c, (pad_size, (0, 0)), "edge")
c = torch.FloatTensor(c).unsqueeze(0).transpose(2, 1).to(device)
p = kwargs.get('f0')
if p is not None:
p = f0_to_coarse(p)
p = np.pad(p, (pad_size,), "edge")
p = torch.LongTensor(p[None, :]).to(device)
y = self.model(z, c, p).view(-1)
wav_out = y.cpu().numpy()
return wav_out
@staticmethod
def wav2spec(wav_fn, return_linear=False):
from preprocessing.data_gen_utils import process_utterance
res = process_utterance(
wav_fn, fft_size=hparams['fft_size'],
hop_size=hparams['hop_size'],
win_length=hparams['win_size'],
num_mels=hparams['audio_num_mel_bins'],
fmin=hparams['fmin'],
fmax=hparams['fmax'],
sample_rate=hparams['audio_sample_rate'],
loud_norm=hparams['loud_norm'],
min_level_db=hparams['min_level_db'],
return_linear=return_linear, vocoder='pwg', eps=float(hparams.get('wav2spec_eps', 1e-10)))
if return_linear:
return res[0], res[1].T, res[2].T # [T, 80], [T, n_fft]
else:
return res[0], res[1].T
@staticmethod
def wav2mfcc(wav_fn):
fft_size = hparams['fft_size']
hop_size = hparams['hop_size']
win_length = hparams['win_size']
sample_rate = hparams['audio_sample_rate']
wav, _ = librosa.core.load(wav_fn, sr=sample_rate)
mfcc = librosa.feature.mfcc(y=wav, sr=sample_rate, n_mfcc=13,
n_fft=fft_size, hop_length=hop_size,
win_length=win_length, pad_mode="constant", power=1.0)
mfcc_delta = librosa.feature.delta(mfcc, order=1)
mfcc_delta_delta = librosa.feature.delta(mfcc, order=2)
mfcc = np.concatenate([mfcc, mfcc_delta, mfcc_delta_delta]).T
return mfcc
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