def _process_utterance(out_dir, index, wav_path, text):
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out,
hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
# Write the spectrograms to disk:
audio_filename = 'ljspeech-audio-%05d.npy' % index
mel_filename = 'ljspeech-mel-%05d.npy' % index
# np.save(os.path.join(out_dir, audio_filename),
# out.astype(out_dtype), allow_pickle=False)
# np.save(os.path.join(out_dir, mel_filename),
# mel_spectrogram.astype(np.float32), allow_pickle=False)
# Return a tuple describing this training example:
return (audio_filename, mel_filename, timesteps, text)
def _process_utterance(out_dir, wav_path):
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out,
hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
return mel_spectrogram.astype(np.float32)
def _process_utterance(out_dir, index, wav_path, text, silence_threshold,
fft_size):
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
if hparams.input_type != "raw":
# Mu-law quantize
out = P.mulaw_quantize(wav)
# Trim silences
start, end = audio.start_and_end_indices(out, silence_threshold)
out = out[start:end]
wav = wav[start:end]
constant_value = P.mulaw_quantize(0, 256)
out_dtype = np.int16
else:
out = wav
constant_value = 0.
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_value)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
wav_id = os.path.basename(wav_path).split('.')[
0] # wav_id = wav_path.split('/')[-1].split('.')[0]
# Write the spectrograms to disk:
audio_filename = '{}-audio.npy'.format(wav_id)
mel_filename = '{}-mel.npy'.format(wav_id)
np.save(os.path.join(out_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32),
allow_pickle=False)
# Return a tuple describing this training example:
return audio_filename, mel_filename, timesteps, text
def _extract_mel(wav_path):
# Load the audio to a numpy array. Resampled if needed.
wav = audio.load_wav(wav_path)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out,
hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjast time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjastment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
assert len(out) // N == audio.get_hop_size()
timesteps = len(out)
return out, mel_spectrogram, timesteps, out_dtype
def get_voice_file(idx, duration, quantize_type):
"""
Gets one of the last VCTK voices
"""
BASE_PATH = "/projects/grail/audiovisual/datasets/VCTK-Corpus/wav48/test"
assert (idx in list(range(0, 100)))
if idx < 25:
speaker_path = os.path.join(BASE_PATH, "p345")
elif idx < 50:
speaker_path = os.path.join(BASE_PATH, "p361")
elif idx < 75:
speaker_path = os.path.join(BASE_PATH, "p362")
elif idx < 100:
speaker_path = os.path.join(BASE_PATH, "p374")
file_list = list(Path(speaker_path).rglob('*.wav'))
curr_file = random.choice(file_list)
y, sr = librosa.core.load(curr_file, sr=22050)
y /= abs(y).max()
start_idx = len(y) // 2
y = y[int(start_idx - duration / 2):int(start_idx + duration / 2)]
# Mulaw, linear or linear max audio
if quantize_type == 0:
quantized = P.mulaw_quantize(y, hparams.quantize_channels - 1)
elif quantize_type == 1:
quantized = linear_quantize(y, hparams.quantize_channels - 1)
return quantized
def wavenet_data():
out = P.mulaw_quantize(wav, hparams.quantize_channels)
out8 = P.mulaw_quantize(wav, 256)
# WAVENENT TRANFSORMATIONS
# Mu-law quantize
# Trim silences
start, end = audio.start_and_end_indices(out, hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
import matplotlib.pyplot as plt
plt.subplot(3, 1, 1)
specshow(mel_spectrogram.T, sr=20000, hop_length=hparams.hop_size)
plt.subplot(3, 1, 2)
plt.plot(out)
plt.xlim(0, len(out))
plt.subplot(3, 1, 3)
plt.plot(wav)
plt.xlim(0, len(wav))
plt.show()
out /= out.max()
def test_mulaw_real():
fs, x = wavfile.read(example_audio_file())
x = (x / 32768.0).astype(np.float32)
mu = 256
y = P.mulaw_quantize(x, mu)
assert y.min() >= 0 and y.max() < mu
assert y.dtype == np.int
x = P.inv_mulaw_quantize(y, mu) * 32768
assert x.dtype == np.float32
x = x.astype(np.int16)
def _process_utterance(out_dir, index, wav_path, text):
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
# Mu-law quantize
quantized = P.mulaw_quantize(wav)
# Trim silences
start, end = audio.start_and_end_indices(quantized, hparams.silence_threshold)
quantized = quantized[start:end]
wav = wav[start:end]
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjast time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
quantized = np.pad(quantized, (l, r), mode="constant",
constant_values=P.mulaw_quantize(0))
N = mel_spectrogram.shape[0]
assert len(quantized) >= N * audio.get_hop_size()
# time resolution adjastment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
quantized = quantized[:N * audio.get_hop_size()]
assert len(quantized) % audio.get_hop_size() == 0
timesteps = len(quantized)
# Write the spectrograms to disk:
audio_filename = 'ljspeech-audio-%05d.npy' % index
mel_filename = 'ljspeech-mel-%05d.npy' % index
np.save(os.path.join(out_dir, audio_filename),
quantized.astype(np.int16), allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32), allow_pickle=False)
# Return a tuple describing this training example:
return (audio_filename, mel_filename, timesteps, text)
def eval_model(global_step, writer, model, y, c, g, input_lengths, eval_dir):
model.eval()
idx = np.random.randint(0, len(y))
length = input_lengths[idx].data.cpu().numpy()[0]
# (T,)
y_target = y[idx].view(-1).data.cpu().long().numpy()[:length]
if c is not None:
c = c[idx, :, :length].unsqueeze(0)
assert c.dim() == 3
print("Shape of local conditioning features: {}".format(c.size()))
if g is not None:
# TODO: test
g = g[idx]
print("Shape of global conditioning features: {}".format(g.size()))
# Dummy silence
initial_value = P.mulaw_quantize(0)
print("Intial value:", initial_value)
# (C,)
initial_input = np_utils.to_categorical(initial_value,
num_classes=256).astype(np.float32)
initial_input = Variable(torch.from_numpy(initial_input),
volatile=True).view(1, 1, 256)
initial_input = initial_input.cuda() if use_cuda else initial_input
y_hat = model.incremental_forward(initial_input,
c=c,
g=g,
T=length,
tqdm=tqdm,
softmax=True,
quantize=True)
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat)
y_target = P.inv_mulaw_quantize(y_target)
# Save audio
os.makedirs(eval_dir, exist_ok=True)
path = join(eval_dir, "step{:09d}_predicted.wav".format(global_step))
librosa.output.write_wav(path, y_hat, sr=hparams.sample_rate)
path = join(eval_dir, "step{:09d}_target.wav".format(global_step))
librosa.output.write_wav(path, y_target, sr=hparams.sample_rate)
# save figure
path = join(eval_dir, "step{:09d}_waveplots.png".format(global_step))
save_waveplot(path, y_hat, y_target)
def _test_data(sr=4000, N=3000, returns_power=False, mulaw=True):
x, _ = librosa.load(example_audio_file(), sr=sr)
x, _ = librosa.effects.trim(x, top_db=15)
# To save computational cost
x = x[:N]
# For power conditioning wavenet
if returns_power:
# (1 x N')
p = librosa.feature.rmse(x, frame_length=256, hop_length=128)
upsample_factor = x.size // p.size
# (1 x N)
p = np.repeat(p, upsample_factor, axis=-1)
if p.size < x.size:
# pad against time axis
p = np.pad(p, [(0, 0), (0, x.size - p.size)],
mode="constant",
constant_values=0)
# shape adajst
p = p.reshape(1, 1, -1)
# (T,)
if mulaw:
x = P.mulaw_quantize(x)
x_org = P.inv_mulaw_quantize(x)
# (C, T)
x = to_categorical(x, num_classes=256).T
# (1, C, T)
x = x.reshape(1, 256, -1).astype(np.float32)
else:
x_org = x
x = x.reshape(1, 1, -1)
if returns_power:
return x, x_org, p
return x, x_org
def get_piano_file(idx, duration, quantize_type):
"""
Gets one of the test supra piano samples
"""
BASE_PATH = "/projects/grail/audiovisual/datasets/supra-rw-mp3/test"
file_list = list(Path(BASE_PATH).rglob("*.mp3"))
curr_file = random.choice(file_list)
y, sr = librosa.core.load(curr_file, sr=22050)
y /= abs(y).max()
num_samples = y.shape[0]
start_idx = random.randint(0, num_samples - duration)
y = y[start_idx:start_idx + duration]
# Mulaw, linear or linear max audio
if quantize_type == 0:
quantized = P.mulaw_quantize(y, hparams.quantize_channels - 1)
elif quantize_type == 1:
quantized = linear_quantize(y, hparams.quantize_channels - 1)
return quantized
def _process_utterance(out_dir, index, speaker_id, wav_path, text):
sr = hparams.sample_rate
# Load the audio to a numpy array. Resampled if needed
wav = audio.load_wav(wav_path)
lab_path = wav_path.replace("wav/", "lab/").replace(".wav", ".lab")
# Trim silence from hts labels if available
# TODO
if exists(lab_path) and False:
labels = hts.load(lab_path)
b = int(start_at(labels) * 1e-7 * sr)
e = int(end_at(labels) * 1e-7 * sr)
wav = wav[b:e]
wav, _ = librosa.effects.trim(wav, top_db=20)
else:
wav, _ = librosa.effects.trim(wav, top_db=20)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out, hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
# Write the spectrograms to disk:
audio_filename = 'cmu_arctic-audio-%05d.npy' % index
mel_filename = 'cmu_arctic-mel-%05d.npy' % index
np.save(os.path.join(out_dir, audio_filename),
out.astype(out_dtype), allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32), allow_pickle=False)
# Return a tuple describing this training example:
return (audio_filename, mel_filename, timesteps, text, speaker_id)
def main(args):
model = ModelWrapper()
model.eval()
if args["--downsample_interval"] is None:
raise(ValueError("Must specify downsample fraction with --downsample_interval"))
downsample_interval = int(args["--downsample_interval"])
receptive_field = model.receptive_field
# Change the output dir if you want
writing_dir = args["<output-dir>"]
os.makedirs(writing_dir, exist_ok=True)
print("writing dir: {}".format(writing_dir))
# Load up a samples
x_original = librosa.core.load(args["<input-file>"], sr=hparams.sample_rate, mono=True)[0]
# Hacky way to allow processing some or all of the file
global SAMPLE_SIZE
if SAMPLE_SIZE == -1:
SAMPLE_SIZE = x_original.shape[0]
x_original = x_original[:SAMPLE_SIZE]
# Normalize to reduce encoding artifacts
x_original /= abs(x_original).max()
sf.write(os.path.join(writing_dir, "x_original.wav"), x_original, hparams.sample_rate)
# Cut the sampling rate
x_modified = x_original[::downsample_interval]
x_modified_out = librosa.core.resample(x_modified,
int(hparams.sample_rate / downsample_interval), hparams.sample_rate)
sf.write(join(writing_dir, "x_modified.wav"), x_modified_out, hparams.sample_rate)
x_modified = P.mulaw_quantize(x_modified, hparams.quantize_channels - 1)
# Update constraint mask for super resolution. Masked spots don't update
mask = np.ones_like(x_original)
mask[::downsample_interval] = 0
mask = torch.Tensor(mask).unsqueeze(0).to(device)
# Initialize with noise for the samples we need to fill in, or x original for the samples
# we are allowed to use
noise = np.random.uniform(0, 256, size=x_original.shape)
mask_np = mask[0].detach().cpu().numpy()
x = P.mulaw_quantize(x_original, hparams.quantize_channels - 1) * (1 - mask_np) + noise * (mask_np)
x = torch.FloatTensor(x).unsqueeze(0).to(device)
x.requires_grad = True
sigmas = [175.9, 110., 68.7, 54.3, 42.9, 34.0, 26.8, 21.2, 16.8, 13.3, 10.5, 8.29, 6.55, 5.18, 4.1, 3.24, 2.56, 1.6, 1.0, 0.625, 0.39, 0.244, 0.15, 0.1]
for idx, sigma in enumerate(sigmas):
# Make sure each sample is updated on average N_STEPS times
n_steps_sgld = int((SAMPLE_SIZE/(SGLD_WINDOW*BATCH_SIZE)) * N_STEPS)
print("Number of SGLD steps {}".format(n_steps_sgld))
# Bump down a model
checkpoint_path = join(args["<checkpoint>"], CHECKPOINTS[sigma], "checkpoint_latest_ema.pth")
model.load_checkpoint(checkpoint_path)
parmodel = torch.nn.DataParallel(model)
parmodel.to(device)
eta = .05 * (sigma ** 2)
for i in range(n_steps_sgld):
# need to get a good sampling of the beginning/end (boundary effects)
# to understand this: think about how often we would update x[receptive_field] (first point)
# if we only sampled U(receptive_field,x0.shape-receptive_field-SGLD_WINDOW)
j = np.random.randint(-SGLD_WINDOW, x.shape[1], BATCH_SIZE)
j = np.maximum(j, 0)
j = np.minimum(j, x.shape[1]-(SGLD_WINDOW))
patches = []
for k in range(BATCH_SIZE):
patches.append(x[:, j[k]:j[k] + SGLD_WINDOW])
patches = torch.stack(patches, axis=0)
# Forward pass
log_prob, prediction = parmodel(patches, sigma=sigma)
log_prob = torch.sum(log_prob)
grad = torch.autograd.grad(log_prob, patches)[0]
x_update = eta * grad
# Langevin step
epsilon = np.sqrt(2 * eta) * torch.normal(0, 1, size=x_update.shape, device=device)
x_update += epsilon
with torch.no_grad():
for k in range(BATCH_SIZE):
x_update[k] *= mask[:, j[k] : j[k] + SGLD_WINDOW]
x[:, j[k] : j[k] + SGLD_WINDOW] += x_update[k]
if (not i % 20) or (i == (n_steps_sgld - 1)): # debugging
print("--------------")
print('sigma = {}'.format(sigma))
print('eta = {}'.format(eta))
print("i {}".format(i))
print("Max sample {}".format(
abs(x).max()))
print('Mean sample logpx: {}'.format(log_prob / (BATCH_SIZE*SGLD_WINDOW)))
print("Max gradient update: {}".format(eta * abs(grad).max()))
t0 = time.time()
out = P.inv_mulaw_quantize(x[0].detach().cpu().numpy(), hparams.quantize_channels - 1)
out = np.clip(out, -1, 1)
sf.write(os.path.join(writing_dir, "out_{}.wav".format(sigma)), out, hparams.sample_rate)
def wavegen(model,
length=None,
c=None,
g=None,
initial_value=None,
fast=False,
tqdm=tqdm):
"""Generate waveform samples by WaveNet.
Args:
model (nn.Module) : WaveNet decoder
length (int): Time steps to generate. If conditinlal features are given,
then this is determined by the feature size.
c (numpy.ndarray): Conditional features, of shape T x C
g (scaler): Speaker ID
initial_value (int) : initial_value for the WaveNet decoder.
fast (Bool): Whether to remove weight normalization or not.
tqdm (lambda): tqdm
Returns:
numpy.ndarray : Generated waveform samples
"""
from train import sanity_check
sanity_check(model, c, g)
c = _to_numpy(c)
g = _to_numpy(g)
model.eval()
if fast:
model.make_generation_fast_()
if c is None:
assert length is not None
else:
# (Tc, D)
if c.ndim != 2:
raise RuntimeError(
"Expected 2-dim shape (T, {}) for the conditional feature, but {} was actually given."
.format(hparams.cin_channels, c.shape))
assert c.ndim == 2
Tc = c.shape[0]
upsample_factor = audio.get_hop_size()
# Overwrite length according to feature size
length = Tc * upsample_factor
# (Tc, D) -> (Tc', D)
# Repeat features before feeding it to the network
if not hparams.upsample_conditional_features:
c = np.repeat(c, upsample_factor, axis=0)
# B x C x T
c = torch.FloatTensor(c.T).unsqueeze(0)
if initial_value is None:
if is_mulaw_quantize(hparams.input_type):
initial_value = P.mulaw_quantize(0, hparams.quantize_channels - 1)
else:
initial_value = 0.0
if is_mulaw_quantize(hparams.input_type):
assert initial_value >= 0 and initial_value < hparams.quantize_channels
initial_input = np_utils.to_categorical(
initial_value,
num_classes=hparams.quantize_channels).astype(np.float32)
initial_input = torch.from_numpy(initial_input).view(
1, 1, hparams.quantize_channels)
else:
initial_input = torch.zeros(1, 1, 1).fill_(initial_value)
g = None if g is None else torch.LongTensor([g])
# Transform data to GPU
initial_input = initial_input.to(device)
g = None if g is None else g.to(device)
c = None if c is None else c.to(device)
with torch.no_grad():
y_hat = model.incremental_forward(initial_input,
c=c,
g=g,
T=length,
tqdm=tqdm,
softmax=True,
quantize=True,
log_scale_min=hparams.log_scale_min)
if is_mulaw_quantize(hparams.input_type):
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
y_hat = P.inv_mulaw(
y_hat.view(-1).cpu().data.numpy(), hparams.quantize_channels)
else:
y_hat = y_hat.view(-1).cpu().data.numpy()
if hparams.postprocess is not None and hparams.postprocess not in [
"", "none"
]:
y_hat = getattr(audio, hparams.postprocess)(y_hat)
if hparams.global_gain_scale > 0:
y_hat /= hparams.global_gain_scale
return y_hat
def eval_model(global_step,
writer,
model,
y,
c,
g,
input_lengths,
eval_dir,
ema=None):
if ema is not None:
print("Using averaged model for evaluation")
model = clone_as_averaged_model(model, ema)
model.eval()
idx = np.random.randint(0, len(y))
length = input_lengths[idx].data.cpu().numpy()[0]
# (T,)
y_target = y[idx].view(-1).data.cpu().numpy()[:length]
if c is not None:
c = c[idx, :, :length].unsqueeze(0)
assert c.dim() == 3
print("Shape of local conditioning features: {}".format(c.size()))
if g is not None:
# TODO: test
g = g[idx]
print("Shape of global conditioning features: {}".format(g.size()))
# Dummy silence
if is_mulaw_quantize(hparams.input_type):
initial_value = P.mulaw_quantize(0, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
initial_value = P.mulaw(0.0, hparams.quantize_channels)
else:
initial_value = 0.0
print("Intial value:", initial_value)
# (C,)
if is_mulaw_quantize(hparams.input_type):
initial_input = np_utils.to_categorical(
initial_value,
num_classes=hparams.quantize_channels).astype(np.float32)
initial_input = Variable(torch.from_numpy(initial_input)).view(
1, 1, hparams.quantize_channels)
else:
initial_input = Variable(torch.zeros(1, 1, 1).fill_(initial_value))
initial_input = initial_input.cuda() if use_cuda else initial_input
# Run the model in fast eval mode
y_hat = model.incremental_forward(initial_input,
c=c,
g=g,
T=length,
softmax=True,
quantize=True,
tqdm=tqdm,
log_scale_min=hparams.log_scale_min)
if is_mulaw_quantize(hparams.input_type):
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat, hparams.quantize_channels)
y_target = P.inv_mulaw_quantize(y_target, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
y_hat = P.inv_mulaw(
y_hat.view(-1).cpu().data.numpy(), hparams.quantize_channels)
y_target = P.inv_mulaw(y_target, hparams.quantize_channels)
else:
y_hat = y_hat.view(-1).cpu().data.numpy()
# Save audio
os.makedirs(eval_dir, exist_ok=True)
path = join(eval_dir, "step{:09d}_predicted.wav".format(global_step))
librosa.output.write_wav(path, y_hat, sr=hparams.sample_rate)
path = join(eval_dir, "step{:09d}_target.wav".format(global_step))
librosa.output.write_wav(path, y_target, sr=hparams.sample_rate)
# save figure
path = join(eval_dir, "step{:09d}_waveplots.png".format(global_step))
save_waveplot(path, y_hat, y_target)
def collate_fn(batch):
"""Create batch
Args:
batch(tuple): List of tuples
- x[0] (ndarray,int) : list of (T,)
- x[1] (ndarray,int) : list of (T, D)
- x[2] (ndarray,int) : list of (1,), speaker id
Returns:
tuple: Tuple of batch
- x (FloatTensor) : Network inputs (B, C, T)
- y (LongTensor) : Network targets (B, T, 1)
"""
local_conditioning = len(batch[0]) >= 2 and hparams.cin_channels > 0
global_conditioning = len(batch[0]) >= 3 and hparams.gin_channels > 0
if hparams.max_time_sec is not None:
max_time_steps = int(hparams.max_time_sec * hparams.sample_rate)
elif hparams.max_time_steps is not None:
max_time_steps = hparams.max_time_steps
else:
max_time_steps = None
# Time resolution adjustment
if local_conditioning:
new_batch = []
for idx in range(len(batch)):
x, c, g = batch[idx]
if hparams.upsample_conditional_features:
assert_ready_for_upsampling(x, c)
if max_time_steps is not None:
max_steps = ensure_divisible(max_time_steps,
audio.get_hop_size(), True)
if len(x) > max_steps:
max_time_frames = max_steps // audio.get_hop_size()
s = np.random.randint(0, len(c) - max_time_frames)
ts = s * audio.get_hop_size()
x = x[ts:ts + audio.get_hop_size() * max_time_frames]
c = c[s:s + max_time_frames, :]
assert_ready_for_upsampling(x, c)
else:
x, c = audio.adjust_time_resolution(x, c)
if max_time_steps is not None and len(x) > max_time_steps:
s = np.random.randint(0, len(x) - max_time_steps)
x, c = x[s:s + max_time_steps], c[s:s + max_time_steps, :]
assert len(x) == len(c)
new_batch.append((x, c, g))
batch = new_batch
else:
new_batch = []
for idx in range(len(batch)):
x, c, g = batch[idx]
x = audio.trim(x)
if max_time_steps is not None and len(x) > max_time_steps:
s = np.random.randint(0, len(x) - max_time_steps)
if local_conditioning:
x, c = x[s:s + max_time_steps], c[s:s + max_time_steps, :]
else:
x = x[s:s + max_time_steps]
new_batch.append((x, c, g))
batch = new_batch
# Lengths
input_lengths = [len(x[0]) for x in batch]
max_input_len = max(input_lengths)
# (B, T, C)
# pad for time-axis
if is_mulaw_quantize(hparams.input_type):
padding_value = P.mulaw_quantize(0, mu=hparams.quantize_channels)
x_batch = np.array([
_pad_2d(
np_utils.to_categorical(x[0],
num_classes=hparams.quantize_channels),
max_input_len, 0, padding_value) for x in batch
],
dtype=np.float32)
else:
x_batch = np.array(
[_pad_2d(x[0].reshape(-1, 1), max_input_len) for x in batch],
dtype=np.float32)
assert len(x_batch.shape) == 3
# (B, T)
if is_mulaw_quantize(hparams.input_type):
padding_value = P.mulaw_quantize(0, mu=hparams.quantize_channels)
y_batch = np.array([
_pad(x[0], max_input_len, constant_values=padding_value)
for x in batch
],
dtype=np.int)
else:
y_batch = np.array([_pad(x[0], max_input_len) for x in batch],
dtype=np.float32)
assert len(y_batch.shape) == 2
# (B, T, D)
if local_conditioning:
max_len = max([len(x[1]) for x in batch])
c_batch = np.array([_pad_2d(x[1], max_len) for x in batch],
dtype=np.float32)
assert len(c_batch.shape) == 3
# (B x C x T)
c_batch = torch.FloatTensor(c_batch).transpose(1, 2).contiguous()
else:
c_batch = None
if global_conditioning:
g_batch = torch.LongTensor([x[2] for x in batch])
else:
g_batch = None
# Covnert to channel first i.e., (B, C, T)
x_batch = torch.FloatTensor(x_batch).transpose(1, 2).contiguous()
# Add extra axis
if is_mulaw_quantize(hparams.input_type):
y_batch = torch.LongTensor(y_batch).unsqueeze(-1).contiguous()
else:
y_batch = torch.FloatTensor(y_batch).unsqueeze(-1).contiguous()
input_lengths = torch.LongTensor(input_lengths)
return x_batch, y_batch, c_batch, g_batch, input_lengths
def _process_utterance(out_dir, index, audio_filepath, text):
# Load the audio to a numpy array:
wav_whole = audio.load_wav(audio_filepath)
if hparams.rescaling:
wav_whole = wav_whole / np.abs(wav_whole).max() * hparams.rescaling_max
# This is a librivox source, so the audio files are going to be v. long
# compared to a typical 'utterance' : So split the wav into chunks
tup_results = []
n_samples = int(8.0 * hparams.sample_rate) # All 8 second utterances
n_chunks = wav_whole.shape[0] // n_samples
for chunk_idx in range(n_chunks):
chunk_start, chunk_end = chunk_idx * \
n_samples, (chunk_idx + 1) * n_samples
if chunk_idx == n_chunks - 1:
# This is the last chunk - allow it
# to extend to the end of the file
chunk_end = None
wav = wav_whole[chunk_start:chunk_end]
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out,
hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution
# between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r),
mode="constant",
constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
# Write the spectrograms to disk:
audio_filename = 'librivox-audio-%04d-%05d.npy' % (
index,
chunk_idx,
)
mel_filename = 'librivox-mel-%04d-%05d.npy' % (
index,
chunk_idx,
)
text_idx = '%s - %05d' % (
text,
chunk_idx,
)
np.save(os.path.join(out_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32),
allow_pickle=False)
# Add results tuple describing this training example:
tup_results.append((audio_filename, mel_filename, timesteps, text_idx))
# Return all the audio results tuples (unpack in caller)
return tup_results
def test_mulaw():
# Check corner cases
assert P.mulaw_quantize(-1.0, 2) == 0
assert P.mulaw_quantize(-0.5, 2) == 0
assert P.mulaw_quantize(-0.001, 2) == 0
assert P.mulaw_quantize(0.0, 2) == 1
assert P.mulaw_quantize(0.0001, 2) == 1
assert P.mulaw_quantize(0.5, 2) == 1
assert P.mulaw_quantize(0.99999, 2) == 1
assert P.mulaw_quantize(1.0, 2) == 2
np.random.seed(1234)
# forward/backward correctness
for mu in [128, 256, 512]:
for x in np.random.rand(100):
y = P.mulaw(x, mu)
assert y >= 0 and y <= 1
x_hat = P.inv_mulaw(y, mu)
assert np.allclose(x, x_hat)
# forward/backward correctness for quantize
for mu in [128, 256, 512]:
for x, y in [(-1.0, 0), (0.0, mu // 2), (0.99999, mu - 1)]:
y_hat = P.mulaw_quantize(x, mu)
err = np.abs(x - P.inv_mulaw_quantize(y_hat, mu))
print(y, y_hat, err)
assert np.allclose(y, y_hat)
# have small quantize error
assert err <= 0.1
# ndarray input
for mu in [128, 256, 512]:
x = np.random.rand(10)
y = P.mulaw(x, mu)
x_hat = P.inv_mulaw(y, mu)
assert np.allclose(x, x_hat)
P.inv_mulaw_quantize(P.mulaw_quantize(x))
# torch array input
from warnings import warn
import torch
torch.manual_seed(1234)
for mu in [128, 256, 512]:
x = torch.rand(10)
y = P.mulaw(x, mu)
x_hat = P.inv_mulaw(y, mu)
assert np.allclose(x, x_hat)
P.inv_mulaw_quantize(P.mulaw_quantize(x))
def _process_utterance(out_dir, index, speaker_id, wav_path, text):
sr = hparams.sample_rate
# Load the audio to a numpy array. Resampled if needed
wav = audio.load_wav(wav_path)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out, hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
#print("Wavepath is ", wav_path)
filename = wav_path.split('/wav/')[-1].split('.wav')[0]
fname = filename
filename = ccoeffs_feats_path + '/' + filename + '.mcep'
mel_spectrogram = np.loadtxt(filename)
#print("Shape of mel scptrogram is ", mel_spectrogram.shape)
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
#mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
#l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
#out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
#out = ensure_divisible(out, N)
#print("Length of out: ", len(out), "N ", N)
#print("Out and N: ", len(out), N)
#if len(out) < N * audio.get_hop_size():
#print("Out and N: ", filename, len(out), N, N * audio.get_hop_size())
# sys.exit()
#assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
#out = out[:N * 80]
#out = ensure_divisible(out, N)
g = open('logfile','a')
g.write("Processing " + fname + '\n')
g.close()
out,mel_spectrogram = ensure_frameperiod(out,mel_spectrogram)
#out = ensure_divisible(out, audio.get_hop_size())
#assert len(out) % audio.get_hop_size() == 0
#assert len(out) % N == 0
timesteps = len(out)
g = open('logfile','a')
g.write(fname + ' ' + str(len(out)) + ' ' + str(N) + ' ' + str(len(out) % N) + '\n')
g.write('\n')
g.close()
# Write the spectrograms to disk:
audio_filename = fname + '-audio-%05d.npy' % index
mel_filename = fname + '-mel-%05d.npy' % index
np.save(os.path.join(out_dir, audio_filename),
out.astype(out_dtype), allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32), allow_pickle=False)
# Return a tuple describing this training example:
return (audio_filename, mel_filename, timesteps, text, speaker_id)
def _process_utterance(wav_path, out_dir):
fname = wav_path.split(os.sep)[-1].split(".")[0]
audio_filename = '{}_resolved.npy'.format(fname)
mel_filename = '{}_mel.npy'.format(fname)
apth = os.path.join(out_dir, audio_filename)
mpth = os.path.join(out_dir, mel_filename)
if os.path.exists(apth) and os.path.exists(mpth):
print("File {} already processed".format(wav_path))
return
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out, hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
# Write the spectrograms to disk:
np.save(apth,
out.astype(out_dtype), allow_pickle=False)
np.save(mpth,
mel_spectrogram.astype(np.float32), allow_pickle=False)
def _process_utterance_single(out_dir, text, wav_path, hparams=hparams):
# modified version of LJSpeech _process_utterance
audio.set_hparams(hparams)
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
sr = hparams.sample_rate
# Added from the multispeaker version
lab_path = wav_path.replace("wav48/", "lab/").replace(".wav", ".lab")
if not exists(lab_path):
lab_path = os.path.splitext(wav_path)[0]+'.lab'
# Trim silence from hts labels if available
if exists(lab_path):
labels = hts.load(lab_path)
wav = clean_by_phoneme(labels, wav, sr)
wav, _ = librosa.effects.trim(wav, top_db=25)
else:
if hparams.process_only_htk_aligned:
return None
wav, _ = librosa.effects.trim(wav, top_db=15)
# End added from the multispeaker version
if hparams.rescaling:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
if hparams.max_audio_length != 0 and librosa.core.get_duration(y=wav, sr=sr) > hparams.max_audio_length:
return None
if hparams.min_audio_length != 0 and librosa.core.get_duration(y=wav, sr=sr) < hparams.min_audio_length:
return None
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels)
# Trim silences
start, end = audio.start_and_end_indices(out, hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = P.mulaw(wav, hparams.quantize_channels)
constant_values = P.mulaw(0.0, hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, hparams.fft_size, audio.get_hop_size())
# zero pad for quantized signal
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
timesteps = len(out)
# Write the spectrograms to disk:
# Get filename from wav_path
wav_name = os.path.basename(wav_path)
wav_name = os.path.splitext(wav_name)[0]
out_filename = 'audio-{}.npy'.format(wav_name)
mel_filename = 'mel-{}.npy'.format(wav_name)
np.save(os.path.join(out_dir, out_filename), out.astype(out_dtype), allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename), mel_spectrogram.astype(np.float32), allow_pickle=False)
# Return a tuple describing this training example:
return (out_filename, mel_filename, timesteps, text)
def eval_model(global_step, writer, device, model, y, c, g, input_lengths, eval_dir, ema=None):
if ema is not None:
print("Using averaged model for evaluation")
model = clone_as_averaged_model(device, model, ema)
model.make_generation_fast_()
model.eval()
#pick one of the available waves to try to emulate
idx = np.random.randint(0, len(y))
length = input_lengths[idx].data.cpu().item()
# (T,)
y_target = y[idx].view(-1).data.cpu().numpy()[:length]
if c is not None:
if hparams.upsample_conditional_features:
c = c[idx, :, :length // audio.get_hop_size()].unsqueeze(0)
else:
c = c[idx, :, :length].unsqueeze(0)
assert c.dim() == 3
print("Shape of local conditioning features: {}".format(c.size()))
if g is not None:
# TODO: test
g = g[idx]
print("Shape of global conditioning features: {}".format(g.size()))
# Dummy silence
if is_mulaw_quantize(hparams.input_type):
initial_value = P.mulaw_quantize(0, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
initial_value = P.mulaw(0.0, hparams.quantize_channels)
else:
#initial_value = 0.0
initial_value = float(y_target[0])
#TODO change initial value to first value of actual waveform instead of zero?? <MLK, 10/19>
print("Intial value:", initial_value)
# (C,)
if is_mulaw_quantize(hparams.input_type):
initial_input = np_utils.to_categorical(
initial_value, num_classes=hparams.quantize_channels).astype(np.float32)
initial_input = torch.from_numpy(initial_input).view(
1, 1, hparams.quantize_channels)
else:
initial_input = torch.zeros(1, 1, 1).fill_(initial_value)
initial_input = initial_input.to(device)
# Run the model in fast eval mode
with torch.no_grad():
y_hat = model.incremental_forward(
initial_input, c=c, g=g, T=length, softmax=True, quantize=True, tqdm=tqdm,
log_scale_min=hparams.log_scale_min)
if is_mulaw_quantize(hparams.input_type):
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat, hparams.quantize_channels)
y_target = P.inv_mulaw_quantize(y_target, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
y_hat = P.inv_mulaw(y_hat.view(-1).cpu().data.numpy(), hparams.quantize_channels)
y_target = P.inv_mulaw(y_target, hparams.quantize_channels)
else:
y_hat = y_hat.view(-1).cpu().data.numpy()
# Save audio
os.makedirs(eval_dir, exist_ok=True)
path = join(eval_dir, "step_noncausal_{:09d}_predicted.npy".format(global_step))
np.save(path, y_hat)
path = join(eval_dir, "step_noncausal_{:09d}_target.npy".format(global_step))
np.save(path, y_target)
# save figure
path = join(eval_dir, "step_noncausal_{:09d}_waveplots.png".format(global_step))
save_waveplot(path, y_hat, y_target)
def _process_utterance(
out_dir,
index,
speaker_id,
wav_path,
text,
silence_threshold,
fft_size,
):
sr = hparams.sample_rate
# Load the audio to a numpy array. Resampled if needed
wav = audio.load_wav(wav_path)
lab_path = wav_path.replace("wav/", "lab/").replace(".wav", ".lab")
# Trim silence from hts labels if available
# TODO
if exists(lab_path) and False:
labels = hts.load(lab_path)
b = int(start_at(labels) * 1e-7 * sr)
e = int(end_at(labels) * 1e-7 * sr)
wav = wav[b:e]
wav, _ = librosa.effects.trim(wav, top_db=20)
else:
wav, _ = librosa.effects.trim(wav, top_db=20)
# Mu-law quantize
quantized = P.mulaw_quantize(wav)
# Trim silences
start, end = audio.start_and_end_indices(quantized, silence_threshold)
quantized = quantized[start:end]
wav = wav[start:end]
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32).T
# lws pads zeros internally before performing stft
# this is needed to adjast time resolution between audio and mel-spectrogram
l, r = audio.lws_pad_lr(wav, fft_size, audio.get_hop_size())
# zero pad for quantized signal
quantized = np.pad(quantized, (l, r),
mode="constant",
constant_values=P.mulaw_quantize(0))
N = mel_spectrogram.shape[0]
assert len(quantized) >= N * audio.get_hop_size()
# time resolution adjastment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
quantized = quantized[:N * audio.get_hop_size()]
assert len(quantized) % audio.get_hop_size() == 0
timesteps = len(quantized)
wav_id = wav_path.split('/')[-1].split('.')[0]
# Write the spectrograms to disk:
audio_filename = '{}-audio.npy'.format(wav_id)
mel_filename = '{}-mel.npy'.format(wav_id)
np.save(os.path.join(out_dir, audio_filename),
quantized.astype(np.int16),
allow_pickle=False)
np.save(os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32),
allow_pickle=False)
# Return a tuple describing this training example:
return (audio_filename, mel_filename, timesteps, text, speaker_id)
def wavegen(model,
length=None,
c=None,
g=None,
initial_value=None,
fast=False,
tqdm=tqdm):
"""Generate waveform samples by WaveNet.
Args:
model (nn.Module) : WaveNet decoder
length (int): Time steps to generate. If conditinlal features are given,
then determined by the feature size.
c (numpy.ndarray): Conditional features, of shape T x C
g (scaler): Speaker ID
initial_value (int) : initial_value for the WaveNet decoder.
fast (Bool): Whether to remove weight normalization or not.
tqdm (lambda): tqdm
Returns:
numpy.ndarray : Generated waveform samples
"""
c = _to_numpy(c)
g = _to_numpy(g)
if use_cuda:
model = model.cuda()
model.eval()
if fast:
model.make_generation_fast_()
if c is None:
assert length is not None
else:
# (N, D)
assert c.ndim == 2
# (T, D)
if not hparams.upsample_conditional_features:
upsample_factor = audio.get_hop_size()
c = np.repeat(c, upsample_factor, axis=0)
length = c.shape[0]
# B x C x T
c = c.T.reshape(1, -1, length)
c = Variable(torch.FloatTensor(c))
if initial_value is None:
initial_value = P.mulaw_quantize(0) # dummy silence
assert initial_value >= 0 and initial_value < 256
initial_input = np_utils.to_categorical(initial_value,
num_classes=256).astype(np.float32)
initial_input = Variable(torch.from_numpy(initial_input)).view(1, 1, 256)
g = None if g is None else Variable(torch.LongTensor([g]))
if use_cuda:
initial_input = initial_input.cuda()
g = None if g is None else g.cuda()
c = None if c is None else c.cuda()
y_hat = model.incremental_forward(initial_input,
c=c,
g=g,
T=length,
tqdm=tqdm,
softmax=True,
quantize=True)
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat)
return y_hat
def eval_model(global_step, writer, device, model, y, c, g, input_lengths, eval_dir, ema=None):
if ema is not None:
print("Using averaged model for evaluation")
model = clone_as_averaged_model(device, model, ema)
model.make_generation_fast_()
model.eval()
idx = np.random.randint(0, len(y))
length = input_lengths[idx].data.cpu().item()
# (T,)
y_target = y[idx].view(-1).data.cpu().numpy()[:length]
if c is not None:
if hparams.upsample_conditional_features:
c = c[idx, :, :length // audio.get_hop_size() + hparams.cin_pad * 2].unsqueeze(0)
else:
c = c[idx, :, :length].unsqueeze(0)
assert c.dim() == 3
print("Shape of local conditioning features: {}".format(c.size()))
if g is not None:
# TODO: test
g = g[idx]
print("Shape of global conditioning features: {}".format(g.size()))
# Dummy silence
if is_mulaw_quantize(hparams.input_type):
initial_value = P.mulaw_quantize(0, hparams.quantize_channels - 1)
elif is_mulaw(hparams.input_type):
initial_value = P.mulaw(0.0, hparams.quantize_channels)
else:
initial_value = 0.0
# (C,)
if is_mulaw_quantize(hparams.input_type):
initial_input = to_categorical(
initial_value, num_classes=hparams.quantize_channels).astype(np.float32)
initial_input = torch.from_numpy(initial_input).view(
1, 1, hparams.quantize_channels)
else:
initial_input = torch.zeros(1, 1, 1).fill_(initial_value)
initial_input = initial_input.to(device)
# Run the model in fast eval mode
with torch.no_grad():
y_hat = model.incremental_forward(
initial_input, c=c, g=g, T=length, softmax=True, quantize=True, tqdm=tqdm,
log_scale_min=hparams.log_scale_min)
if is_mulaw_quantize(hparams.input_type):
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat, hparams.quantize_channels - 1)
y_target = P.inv_mulaw_quantize(y_target, hparams.quantize_channels - 1)
elif is_mulaw(hparams.input_type):
y_hat = P.inv_mulaw(y_hat.view(-1).cpu().data.numpy(), hparams.quantize_channels)
y_target = P.inv_mulaw(y_target, hparams.quantize_channels)
else:
y_hat = y_hat.view(-1).cpu().data.numpy()
# Save audio
os.makedirs(eval_dir, exist_ok=True)
path = join(eval_dir, "step{:09d}_predicted.wav".format(global_step))
librosa.output.write_wav(path, y_hat, sr=hparams.sample_rate)
path = join(eval_dir, "step{:09d}_target.wav".format(global_step))
librosa.output.write_wav(path, y_target, sr=hparams.sample_rate)
# save figure
path = join(eval_dir, "step{:09d}_waveplots.png".format(global_step))
save_waveplot(path, y_hat, y_target)
# add audio and figures to tensorboard
writer.add_audio('target_audio', y_target, global_step, hparams.sample_rate)
writer.add_audio('generated_audio', y_hat, global_step, hparams.sample_rate)
def wavegen(model, length=None, c=None, g=None, initial_value=None,
fast=False, tqdm=tqdm):
"""Generate waveform samples by WaveNet.
Multiple waveforms can be generated in single batch
Args:
model (nn.Module) : WaveNet decoder
length (int): Time steps to generate. If conditinlal features are given,
then this is determined by the feature size.
c (numpy.ndarray or list): Conditional features, of shape T x C
g (scalar or list): Speaker ID
initial_value (int) : initial_value for the WaveNet decoder.
fast (Bool): Whether to remove weight normalization or not.
tqdm (lambda): tqdm
Returns:
numpy.ndarray or list : Generated waveform samples
"""
from train import sanity_check
sanity_check(model, c, g)
model.eval()
if fast:
model.make_generation_fast_()
# Prepare Local Condition
batch_size = 1
output_should_be_list = False
if c is None:
assert length is not None
else:
if type(c)==list :
output_should_be_list = True
c = [_to_numpy(x) for x in c]
for x in c :
if x.ndim != 2:
raise RuntimeError(
"Expected 2-dim shape (T, {}) for the conditional feature, but {} was actually given.".format(hparams.cin_channels, x.shape))
assert x.ndim == 2
batch_size = len(c)
batch = np.zeros([batch_size, max([x.shape[0] for x in c]), c[0].shape[1]])
for i in range(batch_size) :
batch[i,:c[i].shape[0],:] = c[i][:,:]
upsample_factor = audio.get_hop_size()
# length_list : used to cut silence when batch_size > 1
length_list = [x.shape[0]*upsample_factor for x in c]
length = max(length_list)
if not hparams.upsample_conditional_features:
batch = np.repeat(batch, upsample_factor, axis=1)
c = torch.FloatTensor(np.transpose(batch, [0, 2, 1]))
else :
c = _to_numpy(c)
# (Tc, D)
if c.ndim != 2:
raise RuntimeError(
"Expected 2-dim shape (T, {}) for the conditional feature, but {} was actually given.".format(hparams.cin_channels, c.shape))
assert c.ndim == 2
Tc = c.shape[0]
upsample_factor = audio.get_hop_size()
# Overwrite length according to feature size
length = Tc * upsample_factor
# (Tc, D) -> (Tc', D)
# Repeat features before feeding it to the network
if not hparams.upsample_conditional_features:
c = np.repeat(c, upsample_factor, axis=0)
# B x C x T
c = torch.FloatTensor(c.T).unsqueeze(0)
# Prepare initial_input
if initial_value is None:
if is_mulaw_quantize(hparams.input_type):
initial_value = P.mulaw_quantize(0, hparams.quantize_channels)
else:
initial_value = 0.0
if is_mulaw_quantize(hparams.input_type):
assert initial_value >= 0 and initial_value < hparams.quantize_channels
initial_input = np_utils.to_categorical(
initial_value, num_classes=hparams.quantize_channels).astype(np.float32)
initial_input = torch.from_numpy(initial_input).view(
1, 1, hparams.quantize_channels)
else:
initial_input = torch.zeros(1, 1, 1).fill_(initial_value)
initial_input = initial_input.repeat(batch_size, 1, 1)
# Prepare Global Condition
if type(g)==list :
g = [_to_numpy(x) for x in g]
g = torch.LongTensor(g)
elif g is not None :
g = _to_numpy(g)
g = torch.LongTensor([g])
# Transform data to GPU
initial_input = initial_input.to(device)
g = None if g is None else g.to(device)
c = None if c is None else c.to(device)
with torch.no_grad():
y_hat = model.incremental_forward(
initial_input, c=c, g=g, T=length, tqdm=tqdm, softmax=True, quantize=True,
log_scale_min=hparams.log_scale_min)
if is_mulaw_quantize(hparams.input_type):
y_hat = y_hat.max(1)[1].view(batch_size, -1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
y_hat = P.inv_mulaw(y_hat.view(batch_size, -1).cpu().data.numpy(), hparams.quantize_channels)
else:
y_hat = y_hat.view(batch_size, -1).cpu().data.numpy()
if output_should_be_list :
return [y_hat[i, :length_list[i]] for i in range(batch_size)]
else :
return y_hat[0, :]
def _process_utterance(out_dir, index, wav_path, text, trim_silence=False):
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
# Trim begin/end silences
# NOTE: the threshold was chosen for clean signals
# TODO: Remove, get this out of here.
if trim_silence:
wav, _ = librosa.effects.trim(wav,
top_db=60,
frame_length=2048,
hop_length=512)
if hparams.highpass_cutoff > 0.0:
wav = audio.low_cut_filter(wav, hparams.sample_rate,
hparams.highpass_cutoff)
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# Trim silences in mul-aw quantized domain
silence_threshold = 0
if silence_threshold > 0:
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels - 1)
start, end = audio.start_and_end_indices(out, silence_threshold)
wav = wav[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels - 1)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
constant_values = P.mulaw(0.0, hparams.quantize_channels - 1)
out_dtype = np.float32
else:
# [-1, 1]
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.logmelspectrogram(wav).astype(np.float32).T
if hparams.global_gain_scale > 0:
wav *= hparams.global_gain_scale
# Time domain preprocessing
if hparams.preprocess is not None and hparams.preprocess not in [
"", "none"
]:
f = getattr(audio, hparams.preprocess)
wav = f(wav)
# Clip
if np.abs(wav).max() > 1.0:
print("""Warning: abs max value exceeds 1.0: {}""".format(
np.abs(wav).max()))
# ignore this sample
return ("dummy", "dummy", -1, "dummy")
wav = np.clip(wav, -1.0, 1.0)
# Set waveform target (out)
if is_mulaw_quantize(hparams.input_type):
out = P.mulaw_quantize(wav, hparams.quantize_channels - 1)
elif is_mulaw(hparams.input_type):
out = P.mulaw(wav, hparams.quantize_channels - 1)
else:
out = wav
# zero pad
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.pad_lr(out, hparams.fft_size, audio.get_hop_size())
if l > 0 or r > 0:
out = np.pad(out, (l, r),
mode="constant",
constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
assert_ready_for_upsampling(out, mel_spectrogram, cin_pad=0, debug=True)
# Write the spectrograms to disk:
name = splitext(basename(wav_path))[0]
audio_filename = "%s-wave.npy" % (name)
mel_filename = "%s-feats.npy" % (name)
np.save(os.path.join(out_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(
os.path.join(out_dir, mel_filename),
mel_spectrogram.astype(np.float32),
allow_pickle=False,
)
# Return a tuple describing this training example:
return (audio_filename, mel_filename, N, text)
def _process_song(out_dir, index, wav_path, text):
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path)
# Trim begin/end silences
# NOTE: the threshold was chosen for clean signals
wav, _ = librosa.effects.trim(wav,
top_db=60,
frame_length=2048,
hop_length=512)
if hparams.highpass_cutoff > 0.0:
wav = audio.low_cut_filter(wav, hparams.sample_rate,
hparams.highpass_cutoff)
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# Trim silences in mul-aw quantized domain
silence_threshold = 0
if silence_threshold > 0:
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels - 1)
start, end = audio.start_and_end_indices(out, silence_threshold)
wav = wav[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels - 1)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
constant_values = P.mulaw(0.0, hparams.quantize_channels - 1)
out_dtype = np.float32
else:
# [-1, 1]
constant_values = 0.0
out_dtype = np.float32
#### CLAIRE Work here
wav_name = os.path.splitext(os.path.basename(wav_path))[0]
os.makedirs('./pwavs', exist_ok=True)
pwav_path = './pwavs/{0}.wav'.format(wav_name)
scipy.io.wavfile.write(pwav_path, 16000, wav)
# make the chord directory if it does not exist
chord_dir = "chord_dir"
os.makedirs(chord_dir, exist_ok=True)
# create xml file with notes and timestamps
#subprocess.check_call(['./extract_chord_notes.sh', wav_path, chord_dir], shell=True)
#os.system('./extract_chord_notes.sh {0} {1}'.format(pwav_path, chord_dir))
os.system('./extract_chromagram.sh {0} {1} > /dev/null 2>&1'.format(
pwav_path, chord_dir))
note_filename = '{0}/{1}.csv'.format(chord_dir, wav_name)
#### Instead of computing the Mel Spectrogram, here return a time series of one hot encoded chords.
# vector with 1 in row for each note played
# 1000 samples per second
note_samples = int(len(wav) / 2048)
# 12 notes per octave
chords_time_series = np.zeros((24, note_samples))
#print(np.shape(chords_time_series))
with open(note_filename, newline='\n') as csvfile:
#chordreader = csv.reader(csvfile, delimeter=',')
chordreader = csvfile.readlines()
#print(chordreader)
for idx, row in enumerate(chordreader):
row = row.split(",")
chromogram_samples = np.array(row).astype(np.float)[1:]
chords_time_series[:, idx] = chromogram_samples
chords_time_series = chords_time_series.T
# if hparams.global_gain_scale > 0:
# wav *= hparams.global_gain_scale
# Time domain preprocessing
if hparams.preprocess is not None and hparams.preprocess not in [
"", "none"
]:
f = getattr(audio, hparams.preprocess)
wav = f(wav)
# wav = np.clip(wav, -1.0, 1.0)
# Set waveform target (out)
if is_mulaw_quantize(hparams.input_type):
out = P.mulaw_quantize(wav, hparams.quantize_channels - 1)
elif is_mulaw(hparams.input_type):
out = P.mulaw(wav, hparams.quantize_channels - 1)
else:
out = wav
# zero pad
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.pad_lr(out, hparams.fft_size, audio.get_hop_size())
if l > 0 or r > 0:
out = np.pad(out, (l, r),
mode="constant",
constant_values=constant_values)
N = chords_time_series.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
# Write the spectrograms to disk:
name = splitext(basename(wav_path))[0]
audio_filename = '%s-wave.npy' % (name)
chords_filename = '%s-feats.npy' % (name)
np.save(os.path.join(out_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(out_dir, chords_filename),
chords_time_series.astype(out_dtype),
allow_pickle=False)
# Return a tuple describing this training example:
return (audio_filename, chords_filename, N, text)
def wavegen(model,
length=None,
c=None,
g=None,
initial_value=None,
fast=False,
tqdm=tqdm):
"""Generate waveform samples by WaveNet.
Args:
model (nn.Module) : WaveNet decoder
length (int): Time steps to generate. If conditinlal features are given,
then this is determined by the feature size.
c (numpy.ndarray): Conditional features, of shape T x C
g (scaler): Speaker ID
initial_value (int) : initial_value for the WaveNet decoder.
fast (Bool): Whether to remove weight normalization or not.
tqdm (lambda): tqdm
Returns:
numpy.ndarray : Generated waveform samples
"""
from train import sanity_check
sanity_check(model, c, g)
c = _to_numpy(c)
g = _to_numpy(g)
if use_cuda:
model = model.cuda()
model.eval()
if fast:
model.make_generation_fast_()
if c is None:
assert length is not None
else:
# (Tc, D)
assert c.ndim == 2
Tc = c.shape[0]
upsample_factor = audio.get_hop_size()
# Overwrite length according to feature size
length = Tc * upsample_factor
# (Tc, D) -> (Tc', D)
# Repeat features before feeding it to the network
if not hparams.upsample_conditional_features:
c = np.repeat(c, upsample_factor, axis=0)
# B x C x T
c = Variable(torch.FloatTensor(c.T).unsqueeze(0))
if initial_value is None:
if is_mulaw_quantize(hparams.input_type):
initial_value = P.mulaw_quantize(0, hparams.quantize_channels)
else:
initial_value = 0.0
if is_mulaw_quantize(hparams.input_type):
assert initial_value >= 0 and initial_value < hparams.quantize_channels
initial_input = np_utils.to_categorical(
initial_value,
num_classes=hparams.quantize_channels).astype(np.float32)
initial_input = Variable(torch.from_numpy(initial_input)).view(
1, 1, hparams.quantize_channels)
else:
initial_input = Variable(torch.zeros(1, 1, 1)).fill_(initial_value)
g = None if g is None else Variable(torch.LongTensor([g]))
if use_cuda:
initial_input = initial_input.cuda()
g = None if g is None else g.cuda()
c = None if c is None else c.cuda()
y_hat = model.incremental_forward(initial_input,
c=c,
g=g,
T=length,
tqdm=tqdm,
softmax=True,
quantize=True,
log_scale_min=hparams.log_scale_min)
if is_mulaw_quantize(hparams.input_type):
y_hat = y_hat.max(1)[1].view(-1).long().cpu().data.numpy()
y_hat = P.inv_mulaw_quantize(y_hat, hparams.quantize_channels)
elif is_mulaw(hparams.input_type):
y_hat = P.inv_mulaw(
y_hat.view(-1).cpu().data.numpy(), hparams.quantize_channels)
else:
y_hat = y_hat.view(-1).cpu().data.numpy()
return y_hat
def _process_utterance(out_dir,wav_path,sp2ind_dir,text):
sp_f = open(sp2ind_dir,'r')
sp2ind = json.load(sp_f)
sp = wav_path.split('/')[-1].split('.')[0].split('_')[0]
if sp in sp2ind:
sp_ind = sp2ind[sp]
else:
sp_ind = -1
wav = audio.load_wav(wav_path)
if not 'test' in wav_path:
wav,_ = librosa.effects.trim(wav,top_db=60,frame_length=2048,hop_length=512)
if hparams.highpass_cutoff > 0.0:
wav = audio.low_cut_filter(wav, hparams.sample_rate, hparams.highpass_cutoff)
if is_mulaw_quantize(hparams.input_type):
# Trim silences in mul-aw quantized domain
silence_threshold = 0
if silence_threshold > 0:
# [0, quantize_channels)
out = P.mulaw_quantize(wav, hparams.quantize_channels - 1)
start, end = audio.start_and_end_indices(out, silence_threshold)
wav = wav[start:end]
constant_values = P.mulaw_quantize(0, hparams.quantize_channels - 1)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
constant_values = P.mulaw(0.0, hparams.quantize_channels - 1)
out_dtype = np.float32
else:
# [-1, 1]
constant_values = 0.0
out_dtype = np.float32
# Compute a mel-scale spectrogram from the trimmed wav:
# (N, D)
mel_spectrogram = audio.logmelspectrogram(wav).astype(np.float32).T
mfcc = audio.mfcc(wav).astype(np.float32).T
if hparams.global_gain_scale > 0:
wav *= hparams.global_gain_scale
# Time domain preprocessing
if hparams.preprocess is not None and hparams.preprocess not in ["", "none"]:
f = getattr(audio, hparams.preprocess)
wav = f(wav)
# Clip
if np.abs(wav).max() > 1.0:
print("""Warning: abs max value exceeds 1.0: {}""".format(np.abs(wav).max()))
# ignore this sample
#return ("dummy", "dummy","dummy", -1,-1, "dummy")
wav = np.clip(wav, -1.0, 1.0)
# Set waveform target (out)
if is_mulaw_quantize(hparams.input_type):
out = P.mulaw_quantize(wav, hparams.quantize_channels - 1)
elif is_mulaw(hparams.input_type):
out = P.mulaw(wav, hparams.quantize_channels - 1)
else:
out = wav
# zero pad
# this is needed to adjust time resolution between audio and mel-spectrogram
l, r = audio.pad_lr(out, hparams.fft_size, audio.get_hop_size())
if l > 0 or r > 0:
out = np.pad(out, (l, r), mode="constant", constant_values=constant_values)
N = mel_spectrogram.shape[0]
assert len(out) >= N * audio.get_hop_size()
# time resolution adjustment
# ensure length of raw audio is multiple of hop_size so that we can use
# transposed convolution to upsample
out = out[:N * audio.get_hop_size()]
assert len(out) % audio.get_hop_size() == 0
# Write the spectrograms to disk:
#name = splitext(basename(wav_path))[0]
#audio_filename = '%s-wave.npy' % (name)
#mel_filename = '%s-feats.npy' % (name)
audio_filename = f'{out_dir}wave.npy'
mel_filename = f'{out_dir}mel.npy'
mfcc_filename = f'{out_dir}mfcc.npy'
assert mfcc.shape[0] == N
np.save(audio_filename,
out.astype(out_dtype), allow_pickle=False)
np.save(mel_filename,
mel_spectrogram.astype(np.float32), allow_pickle=False)
np.save(mfcc_filename,
mfcc.astype(np.float32), allow_pickle=False)
# Return a tuple describing this training example:
return (out_dir, N, sp_ind,text)
