Fork of https://github.com/alokprasad/fastspeech_squeezewave to also fix denoising in squeezewave
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# We retain the copyright notice by NVIDIA from the original code. However, we
# we reserve our rights on the modifications based on the original code.
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# SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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import os
import random
import argparse
import json
import torch
import torch.utils.data
import sys
from scipy.io.wavfile import read
# We're using the audio processing from TacoTron2 to make sure it matches
from TacotronSTFT import TacotronSTFT
MAX_WAV_VALUE = 32768.0
def files_to_list(filename):
"""
Takes a text file of filenames and makes a list of filenames
"""
with open(filename, encoding='utf-8') as f:
files = f.readlines()
files = [f.rstrip() for f in files]
return files
def load_wav_to_torch(full_path):
"""
Loads wavdata into torch array
"""
sampling_rate, data = read(full_path)
return torch.from_numpy(data).float(), sampling_rate
class Mel2Samp(torch.utils.data.Dataset):
"""
This is the main class that calculates the spectrogram and returns the
spectrogram, audio pair.
"""
def __init__(self, n_audio_channel, training_files, segment_length,
filter_length, hop_length, win_length, sampling_rate, mel_fmin,
mel_fmax):
self.audio_files = files_to_list(training_files)
random.seed(1234)
random.shuffle(self.audio_files)
self.stft = TacotronSTFT(filter_length=filter_length,
hop_length=hop_length,
win_length=win_length,
sampling_rate=sampling_rate,
mel_fmin=mel_fmin, mel_fmax=mel_fmax,
n_group=n_audio_channel)
self.segment_length = segment_length
self.sampling_rate = sampling_rate
def get_mel(self, audio):
audio_norm = audio / MAX_WAV_VALUE
audio_norm = audio_norm.unsqueeze(0)
audio_norm = torch.autograd.Variable(audio_norm, requires_grad=False)
melspec = self.stft.mel_spectrogram(audio_norm)
melspec = torch.squeeze(melspec, 0)
return melspec
def __getitem__(self, index):
# Read audio
filename = self.audio_files[index]
audio, sampling_rate = load_wav_to_torch(filename)
if sampling_rate != self.sampling_rate:
raise ValueError("{} SR doesn't match target {} SR".format(
sampling_rate, self.sampling_rate))
# Take segment
if audio.size(0) >= self.segment_length:
max_audio_start = audio.size(0) - self.segment_length
audio_start = random.randint(0, max_audio_start)
audio = audio[audio_start:audio_start+self.segment_length]
else:
audio = torch.nn.functional.pad(
audio, (0, self.segment_length - audio.size(0)),
'constant').data
mel = self.get_mel(audio)
audio = audio / MAX_WAV_VALUE
return (mel, audio)
def __len__(self):
return len(self.audio_files)
# ===================================================================
# Takes directory of clean audio and makes directory of spectrograms
# Useful for making test sets
# ===================================================================
if __name__ == "__main__":
# Get defaults so it can work with no Sacred
parser = argparse.ArgumentParser()
parser.add_argument('-f', "--filelist_path", required=True)
parser.add_argument('-c', '--config', type=str,
help='JSON file for configuration')
parser.add_argument('-o', '--output_dir', type=str,
help='Output directory')
args = parser.parse_args()
with open(args.config) as f:
data = f.read()
config = json.loads(data)
data_config = config["data_config"]
squeezewave_config = config["squeezewave_config"]
mel2samp = Mel2Samp(squeezewave_config['n_audio_channel'], **data_config)
filepaths = files_to_list(args.filelist_path)
# Make directory if it doesn't exist
if not os.path.isdir(args.output_dir):
os.makedirs(args.output_dir)
os.chmod(args.output_dir, 0o775)
for filepath in filepaths:
audio, sr = load_wav_to_torch(filepath)
melspectrogram = mel2samp.get_mel(audio)
filename = os.path.basename(filepath)
new_filepath = args.output_dir + '/' + filename + '.pt'
print(new_filepath)
torch.save(melspectrogram, new_filepath)