def test_in_out(self):
dummy_input = T.rand(4, 20, 80) # B x T x D
dummy_hidden = [T.rand(2, 4, 128), T.rand(2, 4, 128)]
model = SpeakerEncoder(input_dim=80,
proj_dim=256,
lstm_dim=768,
num_lstm_layers=3)
# computing d vectors
output = model.forward(dummy_input)
assert output.shape[0] == 4
assert output.shape[1] == 256
output = model.inference(dummy_input)
assert output.shape[0] == 4
assert output.shape[1] == 256
# compute d vectors by passing LSTM hidden
# output = model.forward(dummy_input, dummy_hidden)
# assert output.shape[0] == 4
# assert output.shape[1] == 20
# assert output.shape[2] == 256
# check normalization
output_norm = T.nn.functional.normalize(output, dim=1, p=2)
assert_diff = (output_norm - output).sum().item()
assert output.type() == "torch.FloatTensor"
assert (abs(assert_diff) <
1e-4), f" [!] output_norm has wrong values - {assert_diff}"
# compute d for a given batch
dummy_input = T.rand(1, 240, 80) # B x T x D
output = model.compute_embedding(dummy_input,
num_frames=160,
overlap=0.5)
assert output.shape[0] == 1
assert output.shape[1] == 256
assert len(output.shape) == 2
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)
def setup_model(c):
model = SpeakerEncoder(
c.model_params["input_dim"],
c.model_params["proj_dim"],
c.model_params["lstm_dim"],
c.model_params["num_lstm_layers"],
)
return model
def test_speaker_embedding():
# load config
config = load_config(encoder_config_path)
config.audio.resample = True
# create a dummy speaker encoder
model = SpeakerEncoder(**config.model_params)
save_checkpoint(model, None, None, get_tests_input_path(), 0)
# load audio processor and speaker encoder
ap = AudioProcessor(**config.audio)
manager = SpeakerManager(encoder_model_path=encoder_model_path,
encoder_config_path=encoder_config_path)
# load a sample audio and compute embedding
waveform = ap.load_wav(sample_wav_path)
mel = ap.melspectrogram(waveform)
x_vector = manager.compute_x_vector(mel.T)
assert x_vector.shape[1] == 256
# compute x_vector directly from an input file
x_vector = manager.compute_x_vector_from_clip(sample_wav_path)
x_vector2 = manager.compute_x_vector_from_clip(sample_wav_path)
x_vector = torch.FloatTensor(x_vector)
x_vector2 = torch.FloatTensor(x_vector2)
assert x_vector.shape[0] == 256
assert (x_vector - x_vector2).sum() == 0.0
# compute x_vector from a list of wav files.
x_vector3 = manager.compute_x_vector_from_clip(
[sample_wav_path, sample_wav_path2])
x_vector3 = torch.FloatTensor(x_vector3)
assert x_vector3.shape[0] == 256
assert (x_vector - x_vector3).sum() != 0.0
# remove dummy model
os.remove(encoder_model_path)
#print(f'wav_file: {wav_file}')
if os.path.exists(wav_file):
wav_files.append(wav_file)
print(f'Count of wavs imported: {len(wav_files)}')
else:
# Parse all wav files in data_path
wav_path = data_path
wav_files = glob.glob(data_path + '/**/*.wav', recursive=True)
output_files = [
wav_file.replace(wav_path, args.output_path).replace('.wav', '.npy')
for wav_file in wav_files
]
for output_file in output_files:
os.makedirs(os.path.dirname(output_file), exist_ok=True)
model = SpeakerEncoder(**c.model)
model.load_state_dict(torch.load(args.model_path)['model'])
model.eval()
if args.use_cuda:
model.cuda()
for idx, wav_file in enumerate(tqdm(wav_files)):
mel_spec = ap.melspectrogram(ap.load_wav(wav_file)).T
mel_spec = torch.FloatTensor(mel_spec[None, :, :])
if args.use_cuda:
mel_spec = mel_spec.cuda()
embedd = model.compute_embedding(mel_spec)
np.save(output_files[idx], embedd.detach().cpu().numpy())
def setup_model(c):
model = SpeakerEncoder(c.model['input_dim'], c.model['proj_dim'],
c.model['lstm_dim'], c.model['num_lstm_layers'])
return model
