def __init__(self, path, INPUT_SR, TARGET_SR, WINDOW_LENGTH):
self.INPUT_SR = INPUT_SR
self.TARGET_SR = TARGET_SR
self.WINDOW_LENGTH = WINDOW_LENGTH
self.CONFIG = load_config('config_fr.json')
self.CONFIG['audio']['sample_rate'] = self.INPUT_SR
self.AP_INPUT = AudioProcessor(**self.CONFIG['audio'])
self.CONFIG['audio']['sample_rate'] = self.TARGET_SR
self.AP_TARGET = AudioProcessor(**self.CONFIG['audio'])
self.files = glob.glob(path + '/**/*.wav', recursive=True)
#If you change your dataset, delete cache.json
if os.path.isfile('./cache.json'):
with open('./cache.json', "r") as json_file:
self.pre_repertoir = json.load(json_file)
else:
print("> Computing wave files length...")
self.pre_repertoir = [
librosa.get_duration(filename=file)
for file in tqdm(self.files)
]
with open('./cache.json', mode="w") as json_file:
json.dump(self.pre_repertoir, json_file)
self.repertoir = [
int(item / WINDOW_LENGTH) for item in self.pre_repertoir
]
self.length = self.get_len()
def load_tts(self, tts_checkpoint, tts_config, use_cuda):
# pylint: disable=global-statement
global symbols, phonemes
print(" > Loading TTS model ...")
print(" | > model config: ", tts_config)
print(" | > checkpoint file: ", tts_checkpoint)
self.tts_config = load_config(tts_config)
self.use_phonemes = self.tts_config.use_phonemes
self.ap = AudioProcessor(**self.tts_config.audio)
if 'characters' in self.tts_config.keys():
symbols, phonemes = make_symbols(**self.tts_config.characters)
if self.use_phonemes:
self.input_size = len(phonemes)
else:
self.input_size = len(symbols)
# TODO: fix this for multi-speaker model - load speakers
if self.config.tts_speakers is not None:
self.tts_speakers = load_speaker_mapping(self.config.tts_speakers)
num_speakers = len(self.tts_speakers)
else:
num_speakers = 0
self.tts_model = setup_model(self.input_size,
num_speakers=num_speakers,
c=self.tts_config)
# load model state
cp = torch.load(tts_checkpoint, map_location=torch.device('cpu'))
# load the model
self.tts_model.load_state_dict(cp['model'])
if use_cuda:
self.tts_model.cuda()
self.tts_model.eval()
self.tts_model.decoder.max_decoder_steps = 3000
if 'r' in cp:
self.tts_model.decoder.set_r(cp['r'])
print(f" > model reduction factor: {cp['r']}")
def test_scaler(self):
scaler_stats_path = os.path.join(get_tests_input_path(), 'scale_stats.npy')
conf.audio['stats_path'] = scaler_stats_path
conf.audio['preemphasis'] = 0.0
conf.audio['do_trim_silence'] = True
conf.audio['signal_norm'] = True
ap = AudioProcessor(**conf.audio)
mel_mean, mel_std, linear_mean, linear_std, _ = ap.load_stats(scaler_stats_path)
ap.setup_scaler(mel_mean, mel_std, linear_mean, linear_std)
self.ap.signal_norm = False
self.ap.preemphasis = 0.0
# test scaler forward and backward transforms
wav = self.ap.load_wav(WAV_FILE)
mel_reference = self.ap.melspectrogram(wav)
mel_norm = ap.melspectrogram(wav)
mel_denorm = ap._denormalize(mel_norm)
assert abs(mel_reference - mel_denorm).max() < 1e-4
def __init__(self, *args, **kwargs):
super(TestTTSDataset, self).__init__(*args, **kwargs)
self.max_loader_iter = 4
self.ap = AudioProcessor(**c.audio)
class TestTTSDataset(unittest.TestCase):
def __init__(self, *args, **kwargs):
super(TestTTSDataset, self).__init__(*args, **kwargs)
self.max_loader_iter = 4
self.ap = AudioProcessor(**c.audio)
def _create_dataloader(self, batch_size, r, bgs):
items = ljspeech(c.data_path, 'metadata.csv')
dataset = TTSDataset.MyDataset(
r,
c.text_cleaner,
compute_linear_spec=True,
ap=self.ap,
meta_data=items,
tp=c.characters if 'characters' in c.keys() else None,
batch_group_size=bgs,
min_seq_len=c.min_seq_len,
max_seq_len=float("inf"),
use_phonemes=False)
dataloader = DataLoader(dataset,
batch_size=batch_size,
shuffle=False,
collate_fn=dataset.collate_fn,
drop_last=True,
num_workers=c.num_loader_workers)
return dataloader, dataset
def test_loader(self):
if ok_ljspeech:
dataloader, dataset = self._create_dataloader(2, c.r, 0)
for i, data in enumerate(dataloader):
if i == self.max_loader_iter:
break
text_input = data[0]
text_lengths = data[1]
speaker_name = data[2]
linear_input = data[3]
mel_input = data[4]
mel_lengths = data[5]
stop_target = data[6]
item_idx = data[7]
neg_values = text_input[text_input < 0]
check_count = len(neg_values)
assert check_count == 0, \
" !! Negative values in text_input: {}".format(check_count)
# TODO: more assertion here
assert isinstance(speaker_name[0], str)
assert linear_input.shape[0] == c.batch_size
assert linear_input.shape[2] == self.ap.fft_size // 2 + 1
assert mel_input.shape[0] == c.batch_size
assert mel_input.shape[2] == c.audio['num_mels']
# check normalization ranges
if self.ap.symmetric_norm:
assert mel_input.max() <= self.ap.max_norm
assert mel_input.min() >= -self.ap.max_norm #pylint: disable=invalid-unary-operand-type
assert mel_input.min() < 0
else:
assert mel_input.max() <= self.ap.max_norm
assert mel_input.min() >= 0
def test_batch_group_shuffle(self):
if ok_ljspeech:
dataloader, dataset = self._create_dataloader(2, c.r, 16)
last_length = 0
frames = dataset.items
for i, data in enumerate(dataloader):
if i == self.max_loader_iter:
break
text_input = data[0]
text_lengths = data[1]
speaker_name = data[2]
linear_input = data[3]
mel_input = data[4]
mel_lengths = data[5]
stop_target = data[6]
item_idx = data[7]
avg_length = mel_lengths.numpy().mean()
assert avg_length >= last_length
dataloader.dataset.sort_items()
is_items_reordered = False
for idx, item in enumerate(dataloader.dataset.items):
if item != frames[idx]:
is_items_reordered = True
break
assert is_items_reordered
def test_padding_and_spec(self):
if ok_ljspeech:
dataloader, dataset = self._create_dataloader(1, 1, 0)
for i, data in enumerate(dataloader):
if i == self.max_loader_iter:
break
text_input = data[0]
text_lengths = data[1]
speaker_name = data[2]
linear_input = data[3]
mel_input = data[4]
mel_lengths = data[5]
stop_target = data[6]
item_idx = data[7]
# check mel_spec consistency
wav = np.asarray(self.ap.load_wav(item_idx[0]),
dtype=np.float32)
mel = self.ap.melspectrogram(wav).astype('float32')
mel = torch.FloatTensor(mel).contiguous()
mel_dl = mel_input[0]
# NOTE: Below needs to check == 0 but due to an unknown reason
# there is a slight difference between two matrices.
# TODO: Check this assert cond more in detail.
assert abs(mel.T - mel_dl).max() < 1e-5, abs(mel.T -
mel_dl).max()
# check mel-spec correctness
mel_spec = mel_input[0].cpu().numpy()
wav = self.ap.inv_melspectrogram(mel_spec.T)
self.ap.save_wav(wav, OUTPATH + '/mel_inv_dataloader.wav')
shutil.copy(item_idx[0],
OUTPATH + '/mel_target_dataloader.wav')
# check linear-spec
linear_spec = linear_input[0].cpu().numpy()
wav = self.ap.inv_spectrogram(linear_spec.T)
self.ap.save_wav(wav, OUTPATH + '/linear_inv_dataloader.wav')
shutil.copy(item_idx[0],
OUTPATH + '/linear_target_dataloader.wav')
# check the last time step to be zero padded
assert linear_input[0, -1].sum() != 0
assert linear_input[0, -2].sum() != 0
assert mel_input[0, -1].sum() != 0
assert mel_input[0, -2].sum() != 0
assert stop_target[0, -1] == 1
assert stop_target[0, -2] == 0
assert stop_target.sum() == 1
assert len(mel_lengths.shape) == 1
assert mel_lengths[0] == linear_input[0].shape[0]
assert mel_lengths[0] == mel_input[0].shape[0]
# Test for batch size 2
dataloader, dataset = self._create_dataloader(2, 1, 0)
for i, data in enumerate(dataloader):
if i == self.max_loader_iter:
break
text_input = data[0]
text_lengths = data[1]
speaker_name = data[2]
linear_input = data[3]
mel_input = data[4]
mel_lengths = data[5]
stop_target = data[6]
item_idx = data[7]
if mel_lengths[0] > mel_lengths[1]:
idx = 0
else:
idx = 1
# check the first item in the batch
assert linear_input[idx, -1].sum() != 0
assert linear_input[idx, -2].sum() != 0, linear_input
assert mel_input[idx, -1].sum() != 0
assert mel_input[idx, -2].sum() != 0, mel_input
assert stop_target[idx, -1] == 1
assert stop_target[idx, -2] == 0
assert stop_target[idx].sum() == 1
assert len(mel_lengths.shape) == 1
assert mel_lengths[idx] == mel_input[idx].shape[0]
assert mel_lengths[idx] == linear_input[idx].shape[0]
# check the second itme in the batch
assert linear_input[1 - idx, -1].sum() == 0
assert mel_input[1 - idx, -1].sum() == 0
assert stop_target[1, mel_lengths[1] - 1] == 1
assert stop_target[1, mel_lengths[1]:].sum() == 0
assert len(mel_lengths.shape) == 1
from torch import nn, optim
from mozilla_voice_tts.tts.layers.losses import MSELossMasked
from mozilla_voice_tts.tts.models.tacotron2 import Tacotron2
from mozilla_voice_tts.utils.io import load_config
from mozilla_voice_tts.utils.audio import AudioProcessor
#pylint: disable=unused-variable
torch.manual_seed(1)
use_cuda = torch.cuda.is_available()
device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
c = load_config(os.path.join(get_tests_input_path(), 'test_config.json'))
ap = AudioProcessor(**c.audio)
WAV_FILE = os.path.join(get_tests_input_path(), "example_1.wav")
class TacotronTrainTest(unittest.TestCase):
def test_train_step(self): # pylint: disable=no-self-use
input_dummy = torch.randint(0, 24, (8, 128)).long().to(device)
input_lengths = torch.randint(100, 128, (8, )).long().to(device)
input_lengths = torch.sort(input_lengths, descending=True)[0]
mel_spec = torch.rand(8, 30, c.audio['num_mels']).to(device)
mel_postnet_spec = torch.rand(8, 30, c.audio['num_mels']).to(device)
mel_lengths = torch.randint(20, 30, (8, )).long().to(device)
mel_lengths[0] = 30
stop_targets = torch.zeros(8, 30, 1).float().to(device)
speaker_ids = torch.randint(0, 5, (8, )).long().to(device)
class Mel2MelDataset(Dataset):
def __init__(self, path, INPUT_SR, TARGET_SR, WINDOW_LENGTH):
self.INPUT_SR = INPUT_SR
self.TARGET_SR = TARGET_SR
self.WINDOW_LENGTH = WINDOW_LENGTH
self.CONFIG = load_config('config_fr.json')
self.CONFIG['audio']['sample_rate'] = self.INPUT_SR
self.AP_INPUT = AudioProcessor(**self.CONFIG['audio'])
self.CONFIG['audio']['sample_rate'] = self.TARGET_SR
self.AP_TARGET = AudioProcessor(**self.CONFIG['audio'])
self.files = glob.glob(path + '/**/*.wav', recursive=True)
#If you change your dataset, delete cache.json
if os.path.isfile('./cache.json'):
with open('./cache.json', "r") as json_file:
self.pre_repertoir = json.load(json_file)
else:
print("> Computing wave files length...")
self.pre_repertoir = [
librosa.get_duration(filename=file)
for file in tqdm(self.files)
]
with open('./cache.json', mode="w") as json_file:
json.dump(self.pre_repertoir, json_file)
self.repertoir = [
int(item / WINDOW_LENGTH) for item in self.pre_repertoir
]
self.length = self.get_len()
def __len__(self):
return self.length
def __getitem__(self, id):
ref = self.get_reference(id)
input_wav, _ = librosa.load(self.files[ref[0]],
offset=self.WINDOW_LENGTH * ref[1],
duration=self.WINDOW_LENGTH)
target_wav = librosa.resample(input_wav, self.INPUT_SR, self.TARGET_SR)
input = torch.tensor(self.AP_INPUT.melspectrogram(input_wav))
target = torch.tensor(self.AP_TARGET.melspectrogram(target_wav))
scale_factor = (target.shape[0] / input.shape[0],
target.shape[1] / input.shape[1])
input = torch.nn.functional.interpolate(
input.unsqueeze(0).unsqueeze(0),
scale_factor=scale_factor,
mode='bilinear').reshape(target.shape)
return {
'image':
self.normalize(input).unsqueeze(0).type(torch.FloatTensor),
'mask': self.normalize(target).unsqueeze(0).type(torch.FloatTensor)
}
def get_reference(self, id):
i = 0
sum = 0
while True:
if (sum > id):
return (i - 1, id - sum + self.repertoir[i - 1])
else:
sum += self.repertoir[i]
i += 1
def get_len(self):
sum = 0
for num in self.repertoir:
sum += num
return sum
def normalize(self, tensor):
return tensor / 8 + 0.5
def denormalize(self, tensor):
return tensor - 0.5 * 8
import torch
from tests import get_tests_input_path, get_tests_output_path, get_tests_path
from mozilla_voice_tts.utils.audio import AudioProcessor
from mozilla_voice_tts.utils.io import load_config
from mozilla_voice_tts.vocoder.layers.losses import MultiScaleSTFTLoss, STFTLoss, TorchSTFT
TESTS_PATH = get_tests_path()
OUT_PATH = os.path.join(get_tests_output_path(), "audio_tests")
os.makedirs(OUT_PATH, exist_ok=True)
WAV_FILE = os.path.join(get_tests_input_path(), "example_1.wav")
C = load_config(os.path.join(get_tests_input_path(), 'test_config.json'))
ap = AudioProcessor(**C.audio)
def test_torch_stft():
torch_stft = TorchSTFT(ap.fft_size, ap.hop_length, ap.win_length)
# librosa stft
wav = ap.load_wav(WAV_FILE)
M_librosa = abs(ap._stft(wav)) # pylint: disable=protected-access
# torch stft
wav = torch.from_numpy(wav[None, :]).float()
M_torch = torch_stft(wav)
# check the difference b/w librosa and torch outputs
assert (M_librosa - M_torch[0].data.numpy()).max() < 1e-5
def test_stft_loss():
def main(args): # pylint: disable=redefined-outer-name
# pylint: disable=global-variable-undefined
global meta_data_train, meta_data_eval, symbols, phonemes
# Audio processor
ap = AudioProcessor(**c.audio)
if 'characters' in c.keys():
symbols, phonemes = make_symbols(**c.characters)
# DISTRUBUTED
if num_gpus > 1:
init_distributed(args.rank, num_gpus, args.group_id,
c.distributed["backend"], c.distributed["url"])
num_chars = len(phonemes) if c.use_phonemes else len(symbols)
# load data instances
meta_data_train, meta_data_eval = load_meta_data(c.datasets)
# set the portion of the data used for training
if 'train_portion' in c.keys():
meta_data_train = meta_data_train[:int(len(meta_data_train) * c.train_portion)]
if 'eval_portion' in c.keys():
meta_data_eval = meta_data_eval[:int(len(meta_data_eval) * c.eval_portion)]
# parse speakers
if c.use_speaker_embedding:
speakers = get_speakers(meta_data_train)
if args.restore_path:
if c.use_external_speaker_embedding_file: # if restore checkpoint and use External Embedding file
prev_out_path = os.path.dirname(args.restore_path)
speaker_mapping = load_speaker_mapping(prev_out_path)
if not speaker_mapping:
print("WARNING: speakers.json was not found in restore_path, trying to use CONFIG.external_speaker_embedding_file")
speaker_mapping = load_speaker_mapping(c.external_speaker_embedding_file)
if not speaker_mapping:
raise RuntimeError("You must copy the file speakers.json to restore_path, or set a valid file in CONFIG.external_speaker_embedding_file")
speaker_embedding_dim = len(speaker_mapping[list(speaker_mapping.keys())[0]]['embedding'])
elif not c.use_external_speaker_embedding_file: # if restore checkpoint and don't use External Embedding file
prev_out_path = os.path.dirname(args.restore_path)
speaker_mapping = load_speaker_mapping(prev_out_path)
speaker_embedding_dim = None
assert all([speaker in speaker_mapping
for speaker in speakers]), "As of now you, you cannot " \
"introduce new speakers to " \
"a previously trained model."
elif c.use_external_speaker_embedding_file and c.external_speaker_embedding_file: # if start new train using External Embedding file
speaker_mapping = load_speaker_mapping(c.external_speaker_embedding_file)
speaker_embedding_dim = len(speaker_mapping[list(speaker_mapping.keys())[0]]['embedding'])
elif c.use_external_speaker_embedding_file and not c.external_speaker_embedding_file: # if start new train using External Embedding file and don't pass external embedding file
raise "use_external_speaker_embedding_file is True, so you need pass a external speaker embedding file, run GE2E-Speaker_Encoder-ExtractSpeakerEmbeddings-by-sample.ipynb or AngularPrototypical-Speaker_Encoder-ExtractSpeakerEmbeddings-by-sample.ipynb notebook in notebooks/ folder"
else: # if start new train and don't use External Embedding file
speaker_mapping = {name: i for i, name in enumerate(speakers)}
speaker_embedding_dim = None
save_speaker_mapping(OUT_PATH, speaker_mapping)
num_speakers = len(speaker_mapping)
print("Training with {} speakers: {}".format(num_speakers,
", ".join(speakers)))
else:
num_speakers = 0
speaker_embedding_dim = None
speaker_mapping = None
model = setup_model(num_chars, num_speakers, c, speaker_embedding_dim)
params = set_weight_decay(model, c.wd)
optimizer = RAdam(params, lr=c.lr, weight_decay=0)
if c.stopnet and c.separate_stopnet:
optimizer_st = RAdam(model.decoder.stopnet.parameters(),
lr=c.lr,
weight_decay=0)
else:
optimizer_st = None
if c.apex_amp_level == "O1":
# pylint: disable=import-outside-toplevel
from apex import amp
model.cuda()
model, optimizer = amp.initialize(model, optimizer, opt_level=c.apex_amp_level)
else:
amp = None
# setup criterion
criterion = TacotronLoss(c, stopnet_pos_weight=10.0, ga_sigma=0.4)
if args.restore_path:
checkpoint = torch.load(args.restore_path, map_location='cpu')
try:
# TODO: fix optimizer init, model.cuda() needs to be called before
# optimizer restore
# optimizer.load_state_dict(checkpoint['optimizer'])
if c.reinit_layers:
raise RuntimeError
model.load_state_dict(checkpoint['model'])
except KeyError:
print(" > Partial model initialization.")
model_dict = model.state_dict()
model_dict = set_init_dict(model_dict, checkpoint['model'], c)
# torch.save(model_dict, os.path.join(OUT_PATH, 'state_dict.pt'))
# print("State Dict saved for debug in: ", os.path.join(OUT_PATH, 'state_dict.pt'))
model.load_state_dict(model_dict)
del model_dict
if amp and 'amp' in checkpoint:
amp.load_state_dict(checkpoint['amp'])
for group in optimizer.param_groups:
group['lr'] = c.lr
print(" > Model restored from step %d" % checkpoint['step'],
flush=True)
args.restore_step = checkpoint['step']
else:
args.restore_step = 0
if use_cuda:
model.cuda()
criterion.cuda()
# DISTRUBUTED
if num_gpus > 1:
model = apply_gradient_allreduce(model)
if c.noam_schedule:
scheduler = NoamLR(optimizer,
warmup_steps=c.warmup_steps,
last_epoch=args.restore_step - 1)
else:
scheduler = None
num_params = count_parameters(model)
print("\n > Model has {} parameters".format(num_params), flush=True)
if 'best_loss' not in locals():
best_loss = float('inf')
global_step = args.restore_step
for epoch in range(0, c.epochs):
c_logger.print_epoch_start(epoch, c.epochs)
# set gradual training
if c.gradual_training is not None:
r, c.batch_size = gradual_training_scheduler(global_step, c)
c.r = r
model.decoder.set_r(r)
if c.bidirectional_decoder:
model.decoder_backward.set_r(r)
print("\n > Number of output frames:", model.decoder.r)
train_avg_loss_dict, global_step = train(model, criterion, optimizer,
optimizer_st, scheduler, ap,
global_step, epoch, amp, speaker_mapping)
eval_avg_loss_dict = evaluate(model, criterion, ap, global_step, epoch, speaker_mapping)
c_logger.print_epoch_end(epoch, eval_avg_loss_dict)
target_loss = train_avg_loss_dict['avg_postnet_loss']
if c.run_eval:
target_loss = eval_avg_loss_dict['avg_postnet_loss']
best_loss = save_best_model(target_loss, best_loss, model, optimizer, global_step, epoch, c.r,
OUT_PATH, amp_state_dict=amp.state_dict() if amp else None)
class Synthesizer(object):
def __init__(self, config):
self.wavernn = None
self.vocoder_model = None
self.config = config
print(config)
self.seg = self.get_segmenter("en")
self.use_cuda = self.config.use_cuda
if self.use_cuda:
assert torch.cuda.is_available(
), "CUDA is not availabe on this machine."
self.load_tts(self.config.tts_checkpoint, self.config.tts_config,
self.config.use_cuda)
if self.config.vocoder_checkpoint:
self.load_vocoder(self.config.vocoder_checkpoint,
self.config.vocoder_config, self.config.use_cuda)
if self.config.wavernn_lib_path:
self.load_wavernn(self.config.wavernn_lib_path,
self.config.wavernn_checkpoint,
self.config.wavernn_config, self.config.use_cuda)
@staticmethod
def get_segmenter(lang):
return pysbd.Segmenter(language=lang, clean=True)
def load_tts(self, tts_checkpoint, tts_config, use_cuda):
# pylint: disable=global-statement
global symbols, phonemes
print(" > Loading TTS model ...")
print(" | > model config: ", tts_config)
print(" | > checkpoint file: ", tts_checkpoint)
self.tts_config = load_config(tts_config)
self.use_phonemes = self.tts_config.use_phonemes
self.ap = AudioProcessor(**self.tts_config.audio)
if 'characters' in self.tts_config.keys():
symbols, phonemes = make_symbols(**self.tts_config.characters)
if self.use_phonemes:
self.input_size = len(phonemes)
else:
self.input_size = len(symbols)
# TODO: fix this for multi-speaker model - load speakers
if self.config.tts_speakers is not None:
self.tts_speakers = load_speaker_mapping(self.config.tts_speakers)
num_speakers = len(self.tts_speakers)
else:
num_speakers = 0
self.tts_model = setup_model(self.input_size,
num_speakers=num_speakers,
c=self.tts_config)
# load model state
cp = torch.load(tts_checkpoint, map_location=torch.device('cpu'))
# load the model
self.tts_model.load_state_dict(cp['model'])
if use_cuda:
self.tts_model.cuda()
self.tts_model.eval()
self.tts_model.decoder.max_decoder_steps = 3000
if 'r' in cp:
self.tts_model.decoder.set_r(cp['r'])
print(f" > model reduction factor: {cp['r']}")
def load_vocoder(self, model_file, model_config, use_cuda):
self.vocoder_config = load_config(model_config)
self.vocoder_model = setup_generator(self.vocoder_config)
self.vocoder_model.load_state_dict(
torch.load(model_file, map_location="cpu")["model"])
self.vocoder_model.remove_weight_norm()
self.vocoder_model.inference_padding = 0
self.vocoder_config = load_config(model_config)
if use_cuda:
self.vocoder_model.cuda()
self.vocoder_model.eval()
def load_wavernn(self, lib_path, model_file, model_config, use_cuda):
# TODO: set a function in wavernn code base for model setup and call it here.
sys.path.append(
lib_path) # set this if WaveRNN is not installed globally
#pylint: disable=import-outside-toplevel
from WaveRNN.models.wavernn import Model
print(" > Loading WaveRNN model ...")
print(" | > model config: ", model_config)
print(" | > model file: ", model_file)
self.wavernn_config = load_config(model_config)
# This is the default architecture we use for our models.
# You might need to update it
self.wavernn = Model(
rnn_dims=512,
fc_dims=512,
mode=self.wavernn_config.mode,
mulaw=self.wavernn_config.mulaw,
pad=self.wavernn_config.pad,
use_aux_net=self.wavernn_config.use_aux_net,
use_upsample_net=self.wavernn_config.use_upsample_net,
upsample_factors=self.wavernn_config.upsample_factors,
feat_dims=80,
compute_dims=128,
res_out_dims=128,
res_blocks=10,
hop_length=self.ap.hop_length,
sample_rate=self.ap.sample_rate,
).cuda()
check = torch.load(model_file, map_location="cpu")
self.wavernn.load_state_dict(check['model'])
if use_cuda:
self.wavernn.cuda()
self.wavernn.eval()
def save_wav(self, wav, path):
# wav *= 32767 / max(1e-8, np.max(np.abs(wav)))
wav = np.array(wav)
self.ap.save_wav(wav, path)
def split_into_sentences(self, text):
return self.seg.segment(text)
def tts(self, text, speaker_id=None):
start_time = time.time()
wavs = []
sens = self.split_into_sentences(text)
print(sens)
speaker_id = id_to_torch(speaker_id)
if speaker_id is not None and self.use_cuda:
speaker_id = speaker_id.cuda()
for sen in sens:
# preprocess the given text
inputs = text_to_seqvec(sen, self.tts_config)
inputs = numpy_to_torch(inputs, torch.long, cuda=self.use_cuda)
inputs = inputs.unsqueeze(0)
# synthesize voice
_, postnet_output, _, _ = run_model_torch(self.tts_model, inputs,
self.tts_config, False,
speaker_id, None)
if self.vocoder_model:
# use native vocoder model
vocoder_input = postnet_output[0].transpose(0, 1).unsqueeze(0)
wav = self.vocoder_model.inference(vocoder_input)
if self.use_cuda:
wav = wav.cpu().numpy()
else:
wav = wav.numpy()
wav = wav.flatten()
elif self.wavernn:
# use 3rd paty wavernn
vocoder_input = None
if self.tts_config.model == "Tacotron":
vocoder_input = torch.FloatTensor(
self.ap.out_linear_to_mel(
linear_spec=postnet_output.T).T).T.unsqueeze(0)
else:
vocoder_input = postnet_output[0].transpose(0,
1).unsqueeze(0)
if self.use_cuda:
vocoder_input.cuda()
wav = self.wavernn.generate(
vocoder_input,
batched=self.config.is_wavernn_batched,
target=11000,
overlap=550)
else:
# use GL
if self.use_cuda:
postnet_output = postnet_output[0].cpu()
else:
postnet_output = postnet_output[0]
postnet_output = postnet_output.numpy()
wav = inv_spectrogram(postnet_output, self.ap, self.tts_config)
# trim silence
wav = trim_silence(wav, self.ap)
wavs += list(wav)
wavs += [0] * 10000
out = io.BytesIO()
self.save_wav(wavs, out)
# compute stats
process_time = time.time() - start_time
audio_time = len(wavs) / self.tts_config.audio['sample_rate']
print(f" > Processing time: {process_time}")
print(f" > Real-time factor: {process_time / audio_time}")
return out
def gan_dataset_case(batch_size, seq_len, hop_len, conv_pad, return_segments,
use_noise_augment, use_cache, num_workers):
''' run dataloader with given parameters and check conditions '''
ap = AudioProcessor(**C.audio)
_, train_items = load_wav_data(test_data_path, 10)
dataset = GANDataset(ap,
train_items,
seq_len=seq_len,
hop_len=hop_len,
pad_short=2000,
conv_pad=conv_pad,
return_segments=return_segments,
use_noise_augment=use_noise_augment,
use_cache=use_cache)
loader = DataLoader(dataset=dataset,
batch_size=batch_size,
shuffle=True,
num_workers=num_workers,
pin_memory=True,
drop_last=True)
max_iter = 10
count_iter = 0
# return random segments or return the whole audio
if return_segments:
for item1, _ in loader:
feat1, wav1 = item1
# feat2, wav2 = item2
expected_feat_shape = (batch_size, ap.num_mels,
seq_len // hop_len + conv_pad * 2)
# check shapes
assert np.all(feat1.shape == expected_feat_shape
), f" [!] {feat1.shape} vs {expected_feat_shape}"
assert (feat1.shape[2] - conv_pad * 2) * hop_len == wav1.shape[2]
# check feature vs audio match
if not use_noise_augment:
for idx in range(batch_size):
audio = wav1[idx].squeeze()
feat = feat1[idx]
mel = ap.melspectrogram(audio)
# the first 2 and the last 2 frames are skipped due to the padding
# differences in stft
assert (feat - mel[:, :feat1.shape[-1]])[:, 2:-2].sum(
) <= 0, f' [!] {(feat - mel[:, :feat1.shape[-1]])[:, 2:-2].sum()}'
count_iter += 1
# if count_iter == max_iter:
# break
else:
for item in loader:
feat, wav = item
expected_feat_shape = (batch_size, ap.num_mels,
(wav.shape[-1] // hop_len) + (conv_pad * 2))
assert np.all(feat.shape == expected_feat_shape
), f" [!] {feat.shape} vs {expected_feat_shape}"
assert (feat.shape[2] - conv_pad * 2) * hop_len == wav.shape[2]
count_iter += 1
if count_iter == max_iter:
break
def __init__(self, *args, **kwargs):
super(TestAudio, self).__init__(*args, **kwargs)
self.ap = AudioProcessor(**conf.audio)
class TestAudio(unittest.TestCase):
def __init__(self, *args, **kwargs):
super(TestAudio, self).__init__(*args, **kwargs)
self.ap = AudioProcessor(**conf.audio)
def test_audio_synthesis(self):
""" 1. load wav
2. set normalization parameters
3. extract mel-spec
4. invert to wav and save the output
"""
print(" > Sanity check for the process wav -> mel -> wav")
def _test(max_norm, signal_norm, symmetric_norm, clip_norm):
self.ap.max_norm = max_norm
self.ap.signal_norm = signal_norm
self.ap.symmetric_norm = symmetric_norm
self.ap.clip_norm = clip_norm
wav = self.ap.load_wav(WAV_FILE)
mel = self.ap.melspectrogram(wav)
wav_ = self.ap.inv_melspectrogram(mel)
file_name = "/audio_test-melspec_max_norm_{}-signal_norm_{}-symmetric_{}-clip_norm_{}.wav"\
.format(max_norm, signal_norm, symmetric_norm, clip_norm)
print(" | > Creating wav file at : ", file_name)
self.ap.save_wav(wav_, OUT_PATH + file_name)
# maxnorm = 1.0
_test(1., False, False, False)
_test(1., True, False, False)
_test(1., True, True, False)
_test(1., True, False, True)
_test(1., True, True, True)
# maxnorm = 4.0
_test(4., False, False, False)
_test(4., True, False, False)
_test(4., True, True, False)
_test(4., True, False, True)
_test(4., True, True, True)
def test_normalize(self):
"""Check normalization and denormalization for range values and consistency """
print(" > Testing normalization and denormalization.")
wav = self.ap.load_wav(WAV_FILE)
wav = self.ap.sound_norm(wav) # normalize audio to get abetter normalization range below.
self.ap.signal_norm = False
x = self.ap.melspectrogram(wav)
x_old = x
self.ap.signal_norm = True
self.ap.symmetric_norm = False
self.ap.clip_norm = False
self.ap.max_norm = 4.0
x_norm = self.ap._normalize(x)
print(f" > MaxNorm: {self.ap.max_norm}, ClipNorm:{self.ap.clip_norm}, SymmetricNorm:{self.ap.symmetric_norm}, SignalNorm:{self.ap.signal_norm} Range-> {x_norm.max()} -- {x_norm.min()}")
assert (x_old - x).sum() == 0
# check value range
assert x_norm.max() <= self.ap.max_norm + 1, x_norm.max()
assert x_norm.min() >= 0 - 1, x_norm.min()
# check denorm.
x_ = self.ap._denormalize(x_norm)
assert (x - x_).sum() < 1e-3, (x - x_).mean()
self.ap.signal_norm = True
self.ap.symmetric_norm = False
self.ap.clip_norm = True
self.ap.max_norm = 4.0
x_norm = self.ap._normalize(x)
print(f" > MaxNorm: {self.ap.max_norm}, ClipNorm:{self.ap.clip_norm}, SymmetricNorm:{self.ap.symmetric_norm}, SignalNorm:{self.ap.signal_norm} Range-> {x_norm.max()} -- {x_norm.min()}")
assert (x_old - x).sum() == 0
# check value range
assert x_norm.max() <= self.ap.max_norm, x_norm.max()
assert x_norm.min() >= 0, x_norm.min()
# check denorm.
x_ = self.ap._denormalize(x_norm)
assert (x - x_).sum() < 1e-3, (x - x_).mean()
self.ap.signal_norm = True
self.ap.symmetric_norm = True
self.ap.clip_norm = False
self.ap.max_norm = 4.0
x_norm = self.ap._normalize(x)
print(f" > MaxNorm: {self.ap.max_norm}, ClipNorm:{self.ap.clip_norm}, SymmetricNorm:{self.ap.symmetric_norm}, SignalNorm:{self.ap.signal_norm} Range-> {x_norm.max()} -- {x_norm.min()}")
assert (x_old - x).sum() == 0
# check value range
assert x_norm.max() <= self.ap.max_norm + 1, x_norm.max()
assert x_norm.min() >= -self.ap.max_norm - 2, x_norm.min() #pylint: disable=invalid-unary-operand-type
assert x_norm.min() <= 0, x_norm.min()
# check denorm.
x_ = self.ap._denormalize(x_norm)
assert (x - x_).sum() < 1e-3, (x - x_).mean()
self.ap.signal_norm = True
self.ap.symmetric_norm = True
self.ap.clip_norm = True
self.ap.max_norm = 4.0
x_norm = self.ap._normalize(x)
print(f" > MaxNorm: {self.ap.max_norm}, ClipNorm:{self.ap.clip_norm}, SymmetricNorm:{self.ap.symmetric_norm}, SignalNorm:{self.ap.signal_norm} Range-> {x_norm.max()} -- {x_norm.min()}")
assert (x_old - x).sum() == 0
# check value range
assert x_norm.max() <= self.ap.max_norm, x_norm.max()
assert x_norm.min() >= -self.ap.max_norm, x_norm.min() #pylint: disable=invalid-unary-operand-type
assert x_norm.min() <= 0, x_norm.min()
# check denorm.
x_ = self.ap._denormalize(x_norm)
assert (x - x_).sum() < 1e-3, (x - x_).mean()
self.ap.signal_norm = True
self.ap.symmetric_norm = False
self.ap.max_norm = 1.0
x_norm = self.ap._normalize(x)
print(f" > MaxNorm: {self.ap.max_norm}, ClipNorm:{self.ap.clip_norm}, SymmetricNorm:{self.ap.symmetric_norm}, SignalNorm:{self.ap.signal_norm} Range-> {x_norm.max()} -- {x_norm.min()}")
assert (x_old - x).sum() == 0
assert x_norm.max() <= self.ap.max_norm, x_norm.max()
assert x_norm.min() >= 0, x_norm.min()
x_ = self.ap._denormalize(x_norm)
assert (x - x_).sum() < 1e-3
self.ap.signal_norm = True
self.ap.symmetric_norm = True
self.ap.max_norm = 1.0
x_norm = self.ap._normalize(x)
print(f" > MaxNorm: {self.ap.max_norm}, ClipNorm:{self.ap.clip_norm}, SymmetricNorm:{self.ap.symmetric_norm}, SignalNorm:{self.ap.signal_norm} Range-> {x_norm.max()} -- {x_norm.min()}")
assert (x_old - x).sum() == 0
assert x_norm.max() <= self.ap.max_norm, x_norm.max()
assert x_norm.min() >= -self.ap.max_norm, x_norm.min() #pylint: disable=invalid-unary-operand-type
assert x_norm.min() < 0, x_norm.min()
x_ = self.ap._denormalize(x_norm)
assert (x - x_).sum() < 1e-3
def test_scaler(self):
scaler_stats_path = os.path.join(get_tests_input_path(), 'scale_stats.npy')
conf.audio['stats_path'] = scaler_stats_path
conf.audio['preemphasis'] = 0.0
conf.audio['do_trim_silence'] = True
conf.audio['signal_norm'] = True
ap = AudioProcessor(**conf.audio)
mel_mean, mel_std, linear_mean, linear_std, _ = ap.load_stats(scaler_stats_path)
ap.setup_scaler(mel_mean, mel_std, linear_mean, linear_std)
self.ap.signal_norm = False
self.ap.preemphasis = 0.0
# test scaler forward and backward transforms
wav = self.ap.load_wav(WAV_FILE)
mel_reference = self.ap.melspectrogram(wav)
mel_norm = ap.melspectrogram(wav)
mel_denorm = ap._denormalize(mel_norm)
assert abs(mel_reference - mel_denorm).max() < 1e-4
type=str,
help="if CONFIG.use_external_speaker_embedding_file is true, name of speaker embedding reference file present in speakers.json, else target speaker_fileid if the model is multi-speaker.",
default=None)
parser.add_argument(
'--gst_style',
help="Wav path file for GST stylereference.",
default=None)
args = parser.parse_args()
# load the config
C = load_config(args.config_path)
C.forward_attn_mask = True
# load the audio processor
ap = AudioProcessor(**C.audio)
# if the vocabulary was passed, replace the default
if 'characters' in C.keys():
symbols, phonemes = make_symbols(**C.characters)
speaker_embedding = None
speaker_embedding_dim = None
num_speakers = 0
# load speakers
if args.speakers_json != '':
speaker_mapping = json.load(open(args.speakers_json, 'r'))
num_speakers = len(speaker_mapping)
if C.use_external_speaker_embedding_file:
if args.speaker_fileid is not None:
def main(args): # pylint: disable=redefined-outer-name
# pylint: disable=global-variable-undefined
global train_data, eval_data
print(f" > Loading wavs from: {c.data_path}")
if c.feature_path is not None:
print(f" > Loading features from: {c.feature_path}")
eval_data, train_data = load_wav_feat_data(c.data_path, c.feature_path,
c.eval_split_size)
else:
eval_data, train_data = load_wav_data(c.data_path, c.eval_split_size)
# setup audio processor
ap = AudioProcessor(**c.audio)
# DISTRUBUTED
# if num_gpus > 1:
# init_distributed(args.rank, num_gpus, args.group_id,
# c.distributed["backend"], c.distributed["url"])
# setup models
model_gen = setup_generator(c)
model_disc = setup_discriminator(c)
# setup optimizers
optimizer_gen = RAdam(model_gen.parameters(), lr=c.lr_gen, weight_decay=0)
optimizer_disc = RAdam(model_disc.parameters(),
lr=c.lr_disc,
weight_decay=0)
# schedulers
scheduler_gen = None
scheduler_disc = None
if 'lr_scheduler_gen' in c:
scheduler_gen = getattr(torch.optim.lr_scheduler, c.lr_scheduler_gen)
scheduler_gen = scheduler_gen(optimizer_gen,
**c.lr_scheduler_gen_params)
if 'lr_scheduler_disc' in c:
scheduler_disc = getattr(torch.optim.lr_scheduler, c.lr_scheduler_disc)
scheduler_disc = scheduler_disc(optimizer_disc,
**c.lr_scheduler_disc_params)
# setup criterion
criterion_gen = GeneratorLoss(c)
criterion_disc = DiscriminatorLoss(c)
if args.restore_path:
checkpoint = torch.load(args.restore_path, map_location='cpu')
try:
print(" > Restoring Generator Model...")
model_gen.load_state_dict(checkpoint['model'])
print(" > Restoring Generator Optimizer...")
optimizer_gen.load_state_dict(checkpoint['optimizer'])
print(" > Restoring Discriminator Model...")
model_disc.load_state_dict(checkpoint['model_disc'])
print(" > Restoring Discriminator Optimizer...")
optimizer_disc.load_state_dict(checkpoint['optimizer_disc'])
if 'scheduler' in checkpoint:
print(" > Restoring Generator LR Scheduler...")
scheduler_gen.load_state_dict(checkpoint['scheduler'])
# NOTE: Not sure if necessary
scheduler_gen.optimizer = optimizer_gen
if 'scheduler_disc' in checkpoint:
print(" > Restoring Discriminator LR Scheduler...")
scheduler_disc.load_state_dict(checkpoint['scheduler_disc'])
scheduler_disc.optimizer = optimizer_disc
except RuntimeError:
# retore only matching layers.
print(" > Partial model initialization...")
model_dict = model_gen.state_dict()
model_dict = set_init_dict(model_dict, checkpoint['model'], c)
model_gen.load_state_dict(model_dict)
model_dict = model_disc.state_dict()
model_dict = set_init_dict(model_dict, checkpoint['model_disc'], c)
model_disc.load_state_dict(model_dict)
del model_dict
# reset lr if not countinuining training.
for group in optimizer_gen.param_groups:
group['lr'] = c.lr_gen
for group in optimizer_disc.param_groups:
group['lr'] = c.lr_disc
print(" > Model restored from step %d" % checkpoint['step'],
flush=True)
args.restore_step = checkpoint['step']
else:
args.restore_step = 0
if use_cuda:
model_gen.cuda()
criterion_gen.cuda()
model_disc.cuda()
criterion_disc.cuda()
# DISTRUBUTED
# if num_gpus > 1:
# model = apply_gradient_allreduce(model)
num_params = count_parameters(model_gen)
print(" > Generator has {} parameters".format(num_params), flush=True)
num_params = count_parameters(model_disc)
print(" > Discriminator has {} parameters".format(num_params), flush=True)
if 'best_loss' not in locals():
best_loss = float('inf')
global_step = args.restore_step
for epoch in range(0, c.epochs):
c_logger.print_epoch_start(epoch, c.epochs)
_, global_step = train(model_gen, criterion_gen, optimizer_gen,
model_disc, criterion_disc, optimizer_disc,
scheduler_gen, scheduler_disc, ap, global_step,
epoch)
eval_avg_loss_dict = evaluate(model_gen, criterion_gen, model_disc,
criterion_disc, ap, global_step, epoch)
c_logger.print_epoch_end(epoch, eval_avg_loss_dict)
target_loss = eval_avg_loss_dict[c.target_loss]
best_loss = save_best_model(target_loss,
best_loss,
model_gen,
optimizer_gen,
scheduler_gen,
model_disc,
optimizer_disc,
scheduler_disc,
global_step,
epoch,
OUT_PATH,
model_losses=eval_avg_loss_dict)
def main():
"""Run preprocessing process."""
parser = argparse.ArgumentParser(
description="Compute mean and variance of spectrogtram features.")
parser.add_argument(
"--config_path",
type=str,
required=True,
help="TTS config file path to define audio processin parameters.")
parser.add_argument("--out_path",
default=None,
type=str,
help="directory to save the output file.")
args = parser.parse_args()
# load config
CONFIG = load_config(args.config_path)
CONFIG.audio['signal_norm'] = False # do not apply earlier normalization
CONFIG.audio['stats_path'] = None # discard pre-defined stats
# load audio processor
ap = AudioProcessor(**CONFIG.audio)
# load the meta data of target dataset
dataset_items = load_meta_data(CONFIG.datasets)[0] # take only train data
print(f" > There are {len(dataset_items)} files.")
mel_sum = 0
mel_square_sum = 0
linear_sum = 0
linear_square_sum = 0
N = 0
for item in tqdm(dataset_items):
# compute features
wav = ap.load_wav(item[1])
linear = ap.spectrogram(wav)
mel = ap.melspectrogram(wav)
# compute stats
N += mel.shape[1]
mel_sum += mel.sum(1)
linear_sum += linear.sum(1)
mel_square_sum += (mel**2).sum(axis=1)
linear_square_sum += (linear**2).sum(axis=1)
mel_mean = mel_sum / N
mel_scale = np.sqrt(mel_square_sum / N - mel_mean**2)
linear_mean = linear_sum / N
linear_scale = np.sqrt(linear_square_sum / N - linear_mean**2)
output_file_path = os.path.join(args.out_path, "scale_stats.npy")
stats = {}
stats['mel_mean'] = mel_mean
stats['mel_std'] = mel_scale
stats['linear_mean'] = linear_mean
stats['linear_std'] = linear_scale
print(f' > Avg mel spec mean: {mel_mean.mean()}')
print(f' > Avg mel spec scale: {mel_scale.mean()}')
print(f' > Avg linear spec mean: {linear_mean.mean()}')
print(f' > Avg lienar spec scale: {linear_scale.mean()}')
# set default config values for mean-var scaling
CONFIG.audio['stats_path'] = output_file_path
CONFIG.audio['signal_norm'] = True
# remove redundant values
del CONFIG.audio['max_norm']
del CONFIG.audio['min_level_db']
del CONFIG.audio['symmetric_norm']
del CONFIG.audio['clip_norm']
stats['audio_config'] = CONFIG.audio
np.save(output_file_path, stats, allow_pickle=True)
print(f' > scale_stats.npy is saved to {output_file_path}')
def main(args): # pylint: disable=redefined-outer-name
# pylint: disable=global-variable-undefined
global meta_data_train
global meta_data_eval
ap = AudioProcessor(**c.audio)
model = SpeakerEncoder(input_dim=c.model['input_dim'],
proj_dim=c.model['proj_dim'],
lstm_dim=c.model['lstm_dim'],
num_lstm_layers=c.model['num_lstm_layers'])
optimizer = RAdam(model.parameters(), lr=c.lr)
if c.loss == "ge2e":
criterion = GE2ELoss(loss_method='softmax')
elif c.loss == "angleproto":
criterion = AngleProtoLoss()
else:
raise Exception("The %s not is a loss supported" % c.loss)
if args.restore_path:
checkpoint = torch.load(args.restore_path)
try:
# TODO: fix optimizer init, model.cuda() needs to be called before
# optimizer restore
# optimizer.load_state_dict(checkpoint['optimizer'])
if c.reinit_layers:
raise RuntimeError
model.load_state_dict(checkpoint['model'])
except KeyError:
print(" > Partial model initialization.")
model_dict = model.state_dict()
model_dict = set_init_dict(model_dict, checkpoint, c)
model.load_state_dict(model_dict)
del model_dict
for group in optimizer.param_groups:
group['lr'] = c.lr
print(" > Model restored from step %d" % checkpoint['step'],
flush=True)
args.restore_step = checkpoint['step']
else:
args.restore_step = 0
if use_cuda:
model = model.cuda()
criterion.cuda()
if c.lr_decay:
scheduler = NoamLR(optimizer,
warmup_steps=c.warmup_steps,
last_epoch=args.restore_step - 1)
else:
scheduler = None
num_params = count_parameters(model)
print("\n > Model has {} parameters".format(num_params), flush=True)
# pylint: disable=redefined-outer-name
meta_data_train, meta_data_eval = load_meta_data(c.datasets)
global_step = args.restore_step
_, global_step = train(model, criterion, optimizer, scheduler, ap,
global_step)