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