def main():
print("trim audio beggin....")
input = os.path.join("/Users/zhuribing/Documents", "p225_028.wav")
output = os.path.join("/Users/zhuribing/Documents", "trim_p225_028.wav")
wav = audio.load_wav(input, sr=16000)
wav = audio.trim_silence(wav, hparams)
audio.save_wav(wav, output, 16000)
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text, hparams):
"""
<<<<<<< HEAD
Preprocesses a single utterance wavs/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wavs into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:rescaling_max
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: # catch missing wavs exception
print('file {} present in csv metadata is not present in wavs folder. skipping!'.format(
wav_path))
return None
# rescale wavs
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = 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 = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = mulaw(wav, hparams.quantize_channels)
constant_values = mulaw(0., hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text,
speaker_num, lan_num, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print(
'file {} present in csv metadata is not present in wav folder. skipping!'
.format(wav_path))
return None
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#Get spectrogram from wav
ret = audio.wav2spectrograms(wav, hparams)
if ret is None:
return None
out, mel_spectrogram, linear_spectrogram, time_steps, mel_frames = ret
# Write the spectrogram and audio to disk
audio_filename = 'audio-{}.npy'.format(index)
mel_filename = 'mel-{}.npy'.format(index)
linear_filename = 'linear-{}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename),
out.astype(np.float32),
allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
# Return a tuple describing this training example
return (audio_filename, mel_filename, linear_filename, time_steps,
mel_frames, text, speaker_num, lan_num)
def get_second_part_wave(wav, start_time, end_time, hparams):
start_time = int(start_time * 1000)
end_time = int(end_time * 1000)
sentence = wav[start_time:end_time]
temp = sentence.export('temp.wav', format="wav")
sentence = audio.load_wav('temp.wav', sr=hparams.sample_rate)
return sentence
def _process_wave(self, wav_file, num_frames):
try:
wav = audio.load_wav(wav_file, sr=audio_hparams.sample_rate)
except FileNotFoundError:
print(
'file {} present in csv metadata is not present in wav folder. skipping!'
.format(wav_file))
if audio_hparams.trim_silence:
wav = audio.trim_silence(wav, audio_hparams)
expect_len = num_frames * audio_hparams.hop_size + audio_hparams.win_size
if len(wav) < expect_len:
wav = np.concatenate([wav] * np.math.ceil(expect_len / len(wav)))
if len(wav) > expect_len:
sp = random.randint(0, len(wav) - expect_len)
wav = wav[sp:sp + expect_len]
wav = audio.preemphasis(wav, audio_hparams.preemphasis,
audio_hparams.preemphasize)
if audio_hparams.rescale:
wav = wav / np.abs(wav).max() * audio_hparams.rescaling_max
mels = audio.melspectrogram(wav, audio_hparams).astype(np.float32).T
return mels
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text):
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path)
except:
print('file {} present in csv not in folder'.format(wav_path))
return None
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
if hparams.trim_silence:
wav = audio.trim_silence(wav)
out = mulaw_quantize(wav, hparams.quantize_channels)
start, end = audio.start_and_end_indices(out, hparams.silence_threshold)
wav = wav[start:end]
out = out[start:end]
constant_values = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
linear_spectrogram = audio.linearspectrogram(wav).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
assert linear_frames == mel_frames
l, r = audio.pad_lr(wav, hparams.fft_size, audio.get_hop_size())
out = np.pad(out, (l, r), mode='constant', constant_values=constant_values)
time_steps = len(out)
assert time_steps >= mel_frames * audio.get_hop_size()
out = out[:mel_frames * audio.get_hop_size()]
assert time_steps % audio.get_hop_size() == 0
audio_filename = 'speech-audio-{:05d}.npy'.format(index)
mel_filename = 'speech-mel-{:05d}.npy'.format(index)
linear_filename = 'speech-linear-{:05d}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
return (audio_filename, mel_filename, linear_filename, time_steps,
mel_frames, text)
def main():
process_type = '1'
input_path = './wav_files/sample.wav'
output_path = input_path.split(
'.wav')[0] + '_enhanced_mmse' + process_type + '.wav'
wav, sample_rate = audio.load_wav(input_path, sr=None)
wav = wav / np.abs(wav).max() * 0.999
clean_wav = speech_enhancement.mmse_stsa(wav,
sample_rate=sample_rate,
process_type=process_type)
# Plot result
plt.figure()
plt.subplot(2, 1, 1)
librosa.display.waveplot(wav, sr=sample_rate)
plt.title('Noisy Time Signal')
plt.subplot(2, 1, 2)
librosa.display.waveplot(clean_wav, sr=sample_rate)
plt.title('Estimated Clean Time Signal')
plt.figure()
plt.subplot(2, 1, 1)
librosa.display.specshow(librosa.power_to_db(
librosa.feature.melspectrogram(y=wav,
sr=sample_rate,
n_fft=1024,
hop_length=512),
ref=np.max),
sr=sample_rate,
x_axis='time',
y_axis='linear')
plt.title('Noisy Spectrogram')
plt.colorbar(format='%+2.0f dB', boundaries=np.linspace(-70, 0, 10))
plt.subplot(2, 1, 2)
librosa.display.specshow(librosa.power_to_db(
librosa.feature.melspectrogram(y=clean_wav,
sr=sample_rate,
n_fft=1024,
hop_length=512),
ref=np.max),
sr=sample_rate,
x_axis='time',
y_axis='linear')
plt.title('Estimated Clean Spectrogram')
plt.colorbar(format='%+2.0f dB', boundaries=np.linspace(-70, 0, 10))
plt.tight_layout()
plt.show()
clean_wav *= 32767 / max(0.01, np.max(np.abs(clean_wav)))
# proposed by @dsmiller
wavfile.write(output_path, sample_rate, clean_wav.astype(np.int16))
def _process_utterance_libri(mel_dir, label_dir, index, wav_path, label, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- label_dir: the directory to write the label into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- label: time steps spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (wav_path, mel_filename, time_steps, mel_frames, label_filename)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(wav_path))
return None
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#[-1, 1]
out = wav
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(out_dtype)
mel_spectrogram = mel_spectrogram[:, -len(label):]
mel_frames = mel_spectrogram.shape[1]
time_steps = len(out)
# Write the spectrogram and audio to disk
mel_filename = 'mel-{}.npy'.format(index)
label_filename = 'label-{}.npy'.format(index)
np.save(os.path.join(mel_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
np.save(os.path.join(label_dir, label_filename), label, allow_pickle=False)
# Return a tuple describing this training example
return (wav_path, mel_filename, time_steps, mel_frames, label_filename)
def vad():
print("trim audio beggin....")
dataset_root = Path("/Users/zhuribing/Project/AccelerateServerTest/audio")
files = list(dataset_root.joinpath("org").glob("*"))
print(len(files))
cnt = 0
for input in files:
output = str(input).replace("org", "trim", 1)
wav = audio.load_wav(input, sr=16000)
wav = audio.trim_silence(wav, hparams)
audio.save_wav(wav, output, 16000)
cnt = cnt + 1
if (cnt % 10 == 0):
print("complete:", cnt)
def _process_utterance(feat_dir, index, wav_path, text, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, hparams)
except FileNotFoundError: #catch missing wav exception
print(
'file {} present in csv metadata is not present in wav folder. skipping!'
.format(wav_path))
return None
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
# feature extraction
feature = audio.feature_extract(wav, hparams)
n_frames = len(feature)
if n_frames > hparams.max_frame_num or len(text) > hparams.max_text_length:
return None
feat_file = '{}.npy'.format(index)
np.save(os.path.join(feat_dir, feat_file), feature, allow_pickle=False)
# Return a tuple describing this training example
return (feat_file, n_frames, text)
def run_eval(args, checkpoint_path, output_dir, hparams,synth):
if args.model == 'Tacotron-2':
assert os.path.normpath(eval_dir) == os.path.normpath(args.mels_dir)
# Create output path if it doesn't exist
if args.reference_audio_path is not None:
print('reference_audio:', args.reference_audio_path)
ref_wavs = os.listdir(args.reference_audio_path)
else:
if hparams.use_style_encoder == True:
print("*******************************")
print("TODO: add style weights when there is no reference audio. Now we use random weights, " +
"which may generate unintelligible audio sometimes.")
print("*******************************")
else:
# raise ValueError("You must set the reference audio if you don't want to use GSTs.")
print("233")
# Set inputs batch wise
counter=0
fault_ppgs=np.zeros((1,2,345),dtype=np.float32)
for ref_wav in ref_wavs:
speaker = ref_wav.split('_')[0]
sentence = ref_wav.split('_')[1]
save_path = output_dir+'/'+speaker+'/'+sentence
if not os.path.exists(save_path):
os.makedirs(save_path)
counter=counter+1
ref_wav_name = os.path.join(args.reference_audio_path,ref_wav)
save_name = os.path.join(save_path+'/'+ref_wav.split('.')[0]+'.npy')
ref_wav = load_wav(ref_wav_name, hparams.sample_rate)
reference_mel = melspectrogram(ref_wav, hparams).astype(np.float32).T
style_embedding = synth.synthesize_embedding(fault_ppgs,[reference_mel])[0]
np.save(save_name,style_embedding)
print(str(counter)+'/'+str(len(ref_wavs)))
def run_eval(args, checkpoint_path, output_dir, hparams, sentences):
eval_dir = os.path.join(output_dir, 'eval')
log_dir = os.path.join(output_dir, 'logs-eval')
if args.model == 'Tacotron-2':
assert os.path.normpath(eval_dir) == os.path.normpath(args.mels_dir)
#Create output path if it doesn't exist
os.makedirs(eval_dir, exist_ok=True)
os.makedirs(log_dir, exist_ok=True)
os.makedirs(os.path.join(log_dir, 'wavs'), exist_ok=True)
os.makedirs(os.path.join(log_dir, 'plots'), exist_ok=True)
log(hparams_debug_string())
synth = Synthesizer()
if args.reference_audio is not None:
ref_wav = audio.load_wav(args.reference_audio,sr=hparams.sample_rate)
reference_mel = audio.melspectrogram(ref_wav,hparams).astype(np.float32).T
else:
#raise ValueError("Evaluation without reference audio. Please provide path to reference audio.")
reference_mel = None
synth.load(checkpoint_path, hparams, reference_mel=reference_mel)
#Set inputs batch wise
sentences = [sentences[i: i+hparams.tacotron_synthesis_batch_size] for i in range(0, len(sentences), hparams.tacotron_synthesis_batch_size)]
log('Starting Synthesis')
with open(os.path.join(eval_dir, 'map.txt'), 'w') as file:
for i, texts in enumerate(tqdm(sentences)):
start = time.time()
basenames = ['batch_{:03d}_sentence_{:03d}'.format(i, j) for j in range(len(texts))]
mel_filenames, speaker_ids = synth.synthesize(texts, basenames, eval_dir, log_dir, None, reference_mel=reference_mel)
for elems in zip(texts, mel_filenames, speaker_ids):
file.write('|'.join([str(x) for x in elems]) + '\n')
log('synthesized mel spectrograms at {}'.format(eval_dir))
return eval_dir
def run_eval(args, checkpoint_path, output_dir, hparams, ppgs, speakers, Lf0s):
eval_dir = os.path.join(output_dir, 'eval')
log_dir = os.path.join(output_dir, 'logs-eval')
if args.model == 'Tacotron-2':
assert os.path.normpath(eval_dir) == os.path.normpath(args.mels_dir)
#Create output path if it doesn't exist
os.makedirs(eval_dir, exist_ok=True)
os.makedirs(log_dir, exist_ok=True)
os.makedirs(os.path.join(log_dir, 'wavs'), exist_ok=True)
os.makedirs(os.path.join(log_dir, 'plots'), exist_ok=True)
log(hparams_debug_string())
synth = Synthesizer()
synth.load(checkpoint_path, hparams, reference_mels=args.reference_audio)
if args.reference_audio is not None:
print('reference_audio:', args.reference_audio)
ref_wav = load_wav(args.reference_audio.strip(), hparams.sample_rate)
reference_mel = melspectrogram(ref_wav, hparams).astype(np.float32).T
else:
if hparams.use_style_encoder == True:
print("*******************************")
print(
"TODO: add style weights when there is no reference audio. Now we use random weights, "
+ "which may generate unintelligible audio sometimes.")
print("*******************************")
else:
#raise ValueError("You must set the reference audio if you don't want to use GSTs.")
print("233")
#Set inputs batch wise
ppgs = [
ppgs[i:i + hparams.tacotron_synthesis_batch_size]
for i in range(0, len(ppgs), hparams.tacotron_synthesis_batch_size)
]
Lf0s = [
Lf0s[i:i + hparams.tacotron_synthesis_batch_size]
for i in range(0, len(Lf0s), hparams.tacotron_synthesis_batch_size)
]
if args.reference_audio is not None:
reference_mels = [reference_mel] * len(ppgs)
log('Starting Synthesis')
with open(os.path.join(eval_dir, 'map.txt'), 'w') as file:
for i, texts in enumerate(tqdm(ppgs)):
start = time.time()
basenames = [
'batch_{}_sentence_{}'.format(i, j) for j in range(len(texts))
]
if args.reference_audio is not None:
mel_filenames = synth.synthesize(texts, [speakers[i]],
basenames, eval_dir, log_dir,
None, [reference_mels[i]],
Lf0s[i])
else:
mel_filenames = synth.synthesize(texts, [speakers[i]],
basenames, eval_dir, log_dir,
None, None, Lf0s[i])
for elems in zip(texts, mel_filenames, [speakers[i]]):
file.write('|'.join([str(x) for x in elems]) + '\n')
log('synthesized mel spectrograms at {}'.format(eval_dir))
return eval_dir
def build_from_path_ispl(hparams, input_dirs, mel_dir, label_dir, tqdm=lambda x: x):
"""
Preprocesses the speech dataset from a gven input path to given output directories
Args:
- hparams: hyper parameters
- input_dirs: input directory that contains the files to prerocess
- mel_dir: output directory of the preprocessed speech mel-spectrogram dataset
- label_dir: the directory to write the label into
- tqdm: Optional, provides a nice progress bar
Returns:
- A list of tuple describing the train examples. this should be written to train.txt
"""
# We use ProcessPoolExecutor to parallelize across processes, this is just for
# optimization purposes and it can be omited
futures = []
index = 1
for input_dir in input_dirs:
files = find_files(os.path.join(input_dir))
for wav_path in files:
file_name = wav_path.split("\\")[-1]
if int(file_name.split('.')[0]) <= 10:
label_path = wav_path.split("\\")[0] + '/label.txt'
with open(label_path, encoding='utf-8') as f:
lines = f.readlines()
for line in lines:
if file_name in line:
labels = line.replace('[', '').replace(']', '').split(':')[1].replace(',\n', '').split(',')
start = []
end = []
for idx in range(0, len(labels), 2):
start.append(int(labels[idx]))
end.append(int(labels[idx+1]))
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: # catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(wav_path))
return None
# rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
# [-1, 1]
out = wav
out_dtype = np.float32
if int(file_name.split('.')[0]) <= 10:
label = np.zeros_like(out)
for idx in range(len(start)):
start[idx] = int(start[idx] / 1000 * hparams.sample_rate)
end[idx] = int(end[idx] / 1000 * hparams.sample_rate)
label[start[idx]:end[idx]] = 1.
else:
label = wav_path.split('.')[0] + '.label'
with open(label, encoding='utf-8') as f:
lines = f.readlines()
label = np.asarray([int(line.strip('\n')) for line in lines])
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(out_dtype)
mel_spectrogram = mel_spectrogram[:, -len(label):]
mel_frames = mel_spectrogram.shape[1]
# Ensure time resolution adjustement between audio and mel-spectrogram
pad = audio.librosa_pad_lr(wav, hparams.n_fft, audio.get_hop_size(hparams))
if int(file_name.split('.')[0]) <= 10:
# Reflect pad audio signal (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, (0, pad), mode='reflect')
label = np.pad(label, (0, pad), mode='reflect')
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
# time resolution adjustement
# ensure length of raw audio is multiple of hop size so that we can use
# transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
label = label[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
label = label[::audio.get_hop_size(hparams)]
time_steps = len(out)
else:
time_steps = len(out)
# Write the spectrogram and audio to disk
mel_filename = 'mel-{}.npy'.format(index)
label_filename = 'label-{}.npy'.format(index)
np.save(os.path.join(mel_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
np.save(os.path.join(label_dir, label_filename), label, allow_pickle=False)
futures.append((wav_path, mel_filename, time_steps, mel_frames, label_filename))
index += 1
return [future for future in tqdm(futures)]
def _process_utterance(mel_dir, linear_dir, wav_dir, spkid, uttid, wav_path,
text, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print(
'file {} present in csv metadata is not present in wav folder. skipping!'
.format(wav_path))
return None
#Trim lead/trail silences
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#Pre-emphasize
preem_wav = audio.preemphasis(wav, hparams.preemphasis,
hparams.preemphasize)
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
preem_wav = preem_wav / np.abs(preem_wav).max() * hparams.rescaling_max
#Assert all audio is in [-1, 1]
if (wav > 1.).any() or (wav < -1.).any():
raise RuntimeError('wav has invalid value: {}'.format(wav_path))
if (preem_wav > 1.).any() or (preem_wav < -1.).any():
raise RuntimeError('wav has invalid value: {}'.format(wav_path))
#Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
#[0, quantize_channels)
out = mulaw_quantize(wav, hparams.quantize_channels)
#Trim silences
start, end = audio.start_and_end_indices(out,
hparams.silence_threshold)
wav = wav[start:end]
preem_wav = preem_wav[start:end]
out = out[start:end]
constant_values = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
#[-1, 1]
out = mulaw(wav, hparams.quantize_channels)
constant_values = mulaw(0., hparams.quantize_channels)
out_dtype = np.float32
else:
#[-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(preem_wav,
hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
return None
#Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(preem_wav,
hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
#sanity check
assert linear_frames == mel_frames
if hparams.use_lws:
#Ensure time resolution adjustement between audio and mel-spectrogram
fft_size = hparams.n_fft if hparams.win_size is None else hparams.win_size
l, r = audio.pad_lr(wav, fft_size, audio.get_hop_size(hparams))
#Zero pad audio signal
out = np.pad(out, (l, r),
mode='constant',
constant_values=constant_values)
else:
#Ensure time resolution adjustement between audio and mel-spectrogram
l_pad, r_pad = audio.librosa_pad_lr(wav, hparams.n_fft,
audio.get_hop_size(hparams),
hparams.wavenet_pad_sides)
#Reflect pad audio signal on the right (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, (l_pad, r_pad),
mode='constant',
constant_values=constant_values)
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
# Write the spectrogram and audio to disk
sub_wav_dir = os.path.join(wav_dir, spkid)
sub_mel_dir = os.path.join(mel_dir, spkid)
sub_linear_dir = os.path.join(linear_dir, spkid)
os.makedirs(sub_wav_dir, exist_ok=True)
os.makedirs(sub_mel_dir, exist_ok=True)
os.makedirs(sub_linear_dir, exist_ok=True)
audio_filename = 'audio-{}.npy'.format(uttid)
mel_filename = 'mel-{}.npy'.format(uttid)
linear_filename = 'linear-{}.npy'.format(uttid)
np.save(os.path.join(sub_wav_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(sub_mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(sub_linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
# Return a tuple describing this training example
return (spkid, audio_filename, mel_filename, linear_filename, time_steps,
mel_frames, text)
def _process_utterance(out_dir, index, wav_path, pinyin, hparams):
'''Preprocesses a single utterance audio/text pair.
This writes the mel and linear scale spectrograms to disk and returns a tuple to write
to the train.txt file.
Args:
out_dir: The directory to write the spectrograms into
index: The numeric index to use in the spectrogram filenames.
wav_path: Path to the audio file containing the speech input
pinyin: The pinyin of Chinese spoken in the input audio file
Returns:
A (spectrogram_filename, mel_filename, n_frames, text) tuple to write to train.txt
'''
mel_dir = out_dir + "/mels"
linear_dir = out_dir + "/linear"
wav_dir = out_dir + "/audio"
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
print("debug wav_path:", wav_path)
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
#[0, quantize_channels)
out = 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 = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
#[-1, 1]
out = mulaw(wav, hparams.quantize_channels)
constant_values = mulaw(0., hparams.quantize_channels)
out_dtype = np.float32
else:
#[-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute a mel-scale spectrogram from the wav:
#mel_spectrogram = audio.melspectrogram(wav).astype(np.float32)
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
print("debug --- drop wav_path:", wav_path, "mel_frames:", mel_frames)
return None
# Compute the linear-scale spectrogram from the wav:
#spectrogram = audio.spectrogram(wav).astype(np.float32)
#n_frames = spectrogram.shape[1]
linear_spectrogram = audio.linearspectrogram(wav,
hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
#sanity check
assert linear_frames == mel_frames
if hparams.use_lws:
#Ensure time resolution adjustement between audio and mel-spectrogram
fft_size = hparams.n_fft if hparams.win_size is None else hparams.win_size
l, r = audio.pad_lr(wav, fft_size, audio.get_hop_size(hparams))
#Zero pad audio signal
out = np.pad(out, (l, r),
mode='constant',
constant_values=constant_values)
else:
#Ensure time resolution adjustement between audio and mel-spectrogram
pad = audio.librosa_pad_lr(wav, hparams.n_fft,
audio.get_hop_size(hparams))
#Reflect pad audio signal (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, pad, mode='reflect')
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
# Write the spectrograms to disk:
#spectrogram_filename = 'thchs30-spec-%05d.npy' % index
#mel_filename = 'thchs30-mel-%05d.npy' % index
#np.save(os.path.join(out_dir, spectrogram_filename), spectrogram.T, allow_pickle=False)
#np.save(os.path.join(out_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
audio_filename = 'audio-{}.npy'.format(index)
mel_filename = 'mel-{}.npy'.format(index)
linear_filename = 'linear-{}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
print("debug save wav file:", os.path.join(wav_dir, audio_filename))
# Return a tuple describing this training example:
return (audio_filename, mel_filename, linear_filename, time_steps,
mel_frames, pinyin)
def _process_utterance(mel_dir, index, wav_path, start, end, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- start, end: start, end points of speech
- hparams: hyper parameters
Returns:
- A tuple: (wav_path, mel_filename, time_steps, mel_frames, start, end)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(wav_path))
return None
start += 1 * hparams.sample_rate
end += 1 * hparams.sample_rate
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#[-1, 1]
out = wav
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(out_dtype)
mel_frames = mel_spectrogram.shape[1]
# Ensure time resolution adjustement between audio and mel-spectrogram
pad = audio.librosa_pad_lr(wav, hparams.n_fft, audio.get_hop_size(hparams))
# Reflect pad audio signal (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, (0, pad), mode='reflect')
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
# time resolution adjustement
# ensure length of raw audio is multiple of hop size so that we can use
# transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
start = round(start/int(time_steps / mel_frames))
end = round(end/int(time_steps / mel_frames))
# Write the spectrogram and audio to disk
mel_filename = 'mel-{}.npy'.format(index)
np.save(os.path.join(mel_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
# Return a tuple describing this training example
return (wav_path, mel_filename, time_steps, mel_frames, start, end)
def _process_utterance(lf0_dir, mgc_dir, bap_dir, cmp_dir, linear_dir,
basename, wav_path, text, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- basename:
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
if hparams.trim_silence:
tar_wavfile = wav_path[:-4] + "_trim.wav"
print("raw wav path:%s" % wav_path)
wav_raw, fs = sf.read(wav_path)
wav_trim = audio.trim_silence(wav_raw, hparams)
sf.write(tar_wavfile, wav_trim, fs)
wav_path = tar_wavfile
nFFTHalf, alpha, bap_dim = audio.get_config(hparams.sample_rate)
mcsize = hparams.num_mgc - 1
filename = basename #os.path.basename(wav_path).split(".")[0]
print('extract feats for %s' % wav_path)
# extract f0,sp,ap
os.system("analysis %s %s/%s.f0 %s/%s.sp %s/%s.bapd" %
(wav_path, lf0_dir, filename, mgc_dir, filename, bap_dir,
filename)) # get float64???
# interpolate f0
f0 = np.fromfile("%s/%s.f0" % (lf0_dir, filename), dtype=np.float64)
continuous_f0 = interp1d(f0, kind="slinear")
continuous_f0.tofile("%s/%s.f0c" % (lf0_dir, filename))
# convert f0 to lf0
os.system("x2x +da %s/%s.f0c > %s/%s.f0a" %
(lf0_dir, filename, lf0_dir, filename))
os.system(
"x2x +af %s/%s.f0a | sopr -magic 0.0 -LN -MAGIC -1.0E+10 > %s/%s.lf0" %
(lf0_dir, filename, lf0_dir, filename))
# convert　sp to mgc
os.system("x2x +df %s/%s.sp | sopr -R -m 32768.0 | "
"mcep -a %f -m %d -l %d -e 1.0E-8 -j 0 -f 0.0 -q 3 "
"> %s/%s.mgc" %
(mgc_dir, filename, alpha, mcsize, nFFTHalf, mgc_dir, filename))
# convert ap to bap
os.system("x2x +df %s/%s.bapd > %s/%s.bap" %
(bap_dir, filename, bap_dir, filename))
# merge mgc,lf0 and bap to cmp
os.system("merge +f -s 0 -l 1 -L %d %s/%s.mgc < %s/%s.lf0 > %s/%s.ml" % (
(mcsize + 1), mgc_dir, filename, lf0_dir, filename, cmp_dir, filename))
os.system("merge +f -s 0 -l %d -L %d %s/%s.ml < %s/%s.bap > %s/%s.cmp" %
(bap_dim, (mcsize + 2), cmp_dir, filename, bap_dir, filename,
cmp_dir, filename))
#if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
# return None
#Compute the linear scale spectrogram from the wav
wav = audio.load_wav(wav_path, hparams.sample_rate)
linear_spectrogram = audio.linearspectrogram(wav,
hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
#sanity check
#assert linear_frames == mel_frames
lf0 = np.fromfile("%s/%s.lf0" % (lf0_dir, filename), dtype=np.float32)
mgc = np.fromfile("%s/%s.mgc" % (mgc_dir, filename), dtype=np.float32)
bap = np.fromfile("%s/%s.bap" % (bap_dir, filename), dtype=np.float32)
cmp = np.fromfile("%s/%s.cmp" % (cmp_dir, filename), dtype=np.float32)
cmp_dim = mcsize + 1 + 1 + bap_dim
cmp_frames = cmp.shape[0] / cmp_dim
#print(f0[:100])
#print(continuous_f0[:100])
print(lf0.shape)
print(continuous_f0.shape)
print(mgc.shape)
print(bap.shape)
print(cmp_frames)
print(continuous_f0.dtype)
print(mgc.dtype)
print(bap.dtype)
assert (mgc.shape[0] /
(mcsize + 1)) == (continuous_f0.shape[0] /
1) == (bap.shape[0] / bap_dim) == cmp_frames
assert cmp_dim == hparams.num_mels
#assert len(out) >= cmp_frames * audio.get_hop_size(hparams)
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
#out = out[:mel_frames * audio.get_hop_size(hparams)]
#assert len(out) % audio.get_hop_size(hparams) == 0
#time_steps = len(out)
# Write the spectrogram and audio to disk
#audio_filename = 'audio-{}.npy'.format(index)
cmp_mat = cmp.reshape(-1, cmp_dim)
cmp_filename = 'cmp-{}.npy'.format(basename)
linear_filename = 'linear-{}.npy'.format(basename)
#np.save(os.path.join(wav_dir, audio_filename), out.astype(out_dtype), allow_pickle=False)
np.save(os.path.join(cmp_dir, cmp_filename), cmp_mat, allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
# Return a tuple describing this training example
return (cmp_filename, linear_filename, cmp_frames, text)
def synthesize(self, texts, basenames, out_dir, log_dir, mel_filenames):
hparams = self._hparams
cleaner_names = [x.strip() for x in hparams.cleaners.split(',')]
# Repeat last sample until number of samples is dividable by the number of GPUs (last run scenario)
while len(texts) % hparams.tacotron_synthesis_batch_size != 0:
texts.append(texts[-1])
basenames.append(basenames[-1])
if mel_filenames is not None:
mel_filenames.append(mel_filenames[-1])
assert 0 == len(texts) % self._hparams.tacotron_num_gpus
seqs = [
np.asarray(text_to_sequence(text, cleaner_names)) for text in texts
]
input_lengths = [len(seq) for seq in seqs]
size_per_device = len(seqs) // self._hparams.tacotron_num_gpus
# Pad inputs according to each GPU max length
input_seqs = None
split_infos = []
for i in range(self._hparams.tacotron_num_gpus):
device_input = seqs[size_per_device * i:size_per_device * (i + 1)]
device_input, max_seq_len = self._prepare_inputs(device_input)
input_seqs = np.concatenate(
(input_seqs, device_input),
axis=1) if input_seqs is not None else device_input
split_infos.append([max_seq_len, 0, 0, 0])
feed_dict = {
self.inputs: input_seqs,
self.input_lengths: np.asarray(input_lengths, dtype=np.int32),
}
if self.gta:
np_targets = [
np.load(mel_filename) for mel_filename in mel_filenames
]
target_lengths = [len(np_target) for np_target in np_targets]
# pad targets according to each GPU max length
target_seqs = None
for i in range(self._hparams.tacotron_num_gpus):
device_target = np_targets[size_per_device *
i:size_per_device * (i + 1)]
device_target, max_target_len = self._prepare_targets(
device_target, self._hparams.outputs_per_step)
target_seqs = np.concatenate(
(target_seqs, device_target),
axis=1) if target_seqs is not None else device_target
split_infos[i][1] = \
max_target_len # Not really used but setting it in case for future development maybe?
feed_dict[self.targets] = target_seqs
assert len(np_targets) == len(texts)
if self.style_transfer and hparams.tacotron_style_reference_audio is not None and\
hparams.tacotron_style_alignment is None:
# only support one style reference audio
if hparams.tacotron_style_reference_audio[-4:] == '.wav':
wav = audio.load_wav(hparams.tacotron_style_reference_audio,
sr=hparams.sample_rate)
np_targets = audio.melspectrogram(wav, self._hparams).astype(
np.float32).T
else:
np_targets = np.load(hparams.tacotron_style_reference_audio)
target_lengths = len(np_targets)
# copy
np_targets = [np_targets for _ in range(len(texts))]
target_lengths = [target_lengths for _ in range(len(texts))]
# pad targets according to each GPU max length
target_seqs = None
for i in range(self._hparams.tacotron_num_gpus):
device_target = np_targets[size_per_device *
i:size_per_device * (i + 1)]
device_target, max_target_len = self._prepare_targets(
device_target, self._hparams.outputs_per_step)
target_seqs = np.concatenate(
(target_seqs, device_target),
axis=1) if target_seqs is not None else device_target
split_infos[i][1] = \
max_target_len # Not really used but setting it in case for future development maybe?
feed_dict[self.targets] = target_seqs
feed_dict[self.target_lengths] = target_lengths
assert len(np_targets) == len(texts)
feed_dict[self.split_infos] = np.asarray(split_infos, dtype=np.int32)
if self.gta or not hparams.predict_linear:
mels, alignments, stop_tokens = self.session.run(
[
self.mel_outputs, self.alignments,
self.stop_token_prediction
],
feed_dict=feed_dict)
# Linearize outputs (1D arrays)
mels = [mel for gpu_mels in mels for mel in gpu_mels]
alignments = [
align for gpu_aligns in alignments for align in gpu_aligns
]
stop_tokens = [
token for gpu_token in stop_tokens for token in gpu_token
]
if not self.gta:
# Natural batch synthesis
# Get Mel lengths for the entire batch from stop_tokens predictions
target_lengths = self._get_output_lengths(stop_tokens)
# Take off the batch wise padding
mels = [
mel[:target_length, :]
for mel, target_length in zip(mels, target_lengths)
]
assert len(mels) == len(texts)
else:
linears, mels, alignments, stop_tokens = self.session.run(
[
self.linear_outputs, self.mel_outputs, self.alignments,
self.stop_token_prediction
],
feed_dict=feed_dict)
# Linearize outputs (1D arrays)
linears = [
linear for gpu_linear in linears for linear in gpu_linear
]
mels = [mel for gpu_mels in mels for mel in gpu_mels]
alignments = [
align for gpu_aligns in alignments for align in gpu_aligns
]
stop_tokens = [
token for gpu_token in stop_tokens for token in gpu_token
]
# Natural batch synthesis
# Get Mel/Linear lengths for the entire batch from stop_tokens predictions
# target_lengths = self._get_output_lengths(stop_tokens)
target_lengths = [9999]
# Take off the batch wise padding
mels = [
mel[:target_length, :]
for mel, target_length in zip(mels, target_lengths)
]
linears = [
linear[:target_length, :]
for linear, target_length in zip(linears, target_lengths)
]
assert len(mels) == len(linears) == len(texts)
if basenames is None:
# Generate wav and read it
wav = audio.inv_mel_spectrogram(mels.T, hparams)
audio.save_wav(wav, 'temp.wav',
sr=hparams.sample_rate) # Find a better way
chunk = 512
f = wave.open('temp.wav', 'rb')
p = pyaudio.PyAudio()
stream = p.open(format=p.get_format_from_width(f.getsampwidth()),
channels=f.getnchannels(),
rate=f.getframerate(),
output=True)
data = f.readframes(chunk)
while data:
stream.write(data)
data = f.readframes(chunk)
stream.stop_stream()
stream.close()
p.terminate()
return
saved_mels_paths = []
speaker_ids = []
for i, mel in enumerate(mels):
# Get speaker id for global conditioning (only used with GTA generally)
if hparams.gin_channels > 0:
raise RuntimeError(
'Please set the speaker_id rule in line 99 of tacotron/synthesizer.py to allow for global condition usage later.'
)
speaker_id = '<no_g>' # set the rule to determine speaker id. By using the file basename maybe? (basenames are inside "basenames" variable)
speaker_ids.append(
speaker_id
) # finish by appending the speaker id. (allows for different speakers per batch if your model is multispeaker)
else:
speaker_id = '<no_g>'
speaker_ids.append(speaker_id)
# Write the spectrogram to disk
# Note: outputs mel-spectrogram files and target ones have same names, just different folders
mel_filename = os.path.join(out_dir,
'mel-{}.npy'.format(basenames[i]))
np.save(mel_filename, mel, allow_pickle=False)
saved_mels_paths.append(mel_filename)
if log_dir is not None:
# save wav (mel -> wav)
wav = audio.inv_mel_spectrogram(mel.T, hparams)
audio.save_wav(wav,
os.path.join(
log_dir,
'wavs/wav-{}-mel.wav'.format(basenames[i])),
sr=hparams.sample_rate)
# save alignments
plot.plot_alignment(alignments[i],
os.path.join(
log_dir,
'plots/alignment-{}.png'.format(
basenames[i])),
title='{}'.format(texts[i]),
split_title=True,
max_len=target_lengths[i])
# save mel spectrogram plot
plot.plot_spectrogram(
mel,
os.path.join(log_dir,
'plots/mel-{}.png'.format(basenames[i])),
title='{}'.format(texts[i]),
split_title=True)
if hparams.predict_linear:
# save wav (linear -> wav)
wav = audio.inv_linear_spectrogram(linears[i].T, hparams)
audio.save_wav(wav,
os.path.join(
log_dir,
'wavs/wav-{}-linear.wav'.format(
basenames[i])),
sr=hparams.sample_rate)
# save linear spectrogram plot
plot.plot_spectrogram(linears[i],
os.path.join(
log_dir,
'plots/linear-{}.png'.format(
basenames[i])),
title='{}'.format(texts[i]),
split_title=True,
auto_aspect=True)
return saved_mels_paths, speaker_ids
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, ppgs,
lf0_path, speaker, refer, hparams):
"""
Preprocesses a single utterance wav/ppgs pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- ppgs: ppgs spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, refer_name ,time_steps, mel_frames, linear_frames, ppgs,speaker,lf0)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print(
'file {} present in csv metadata is not present in wav folder. skipping!'
.format(wav_path))
return None
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
return None
#Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(wav,
hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
#sanity check
assert linear_frames == mel_frames
if hparams.use_lws:
#Ensure time resolution adjustement between audio and mel-spectrogram
fft_size = hparams.n_fft if hparams.win_size is None else hparams.win_size
l, r = audio.pad_lr(wav, fft_size, audio.get_hop_size(hparams))
#Zero pad audio signal
out = np.pad(out, (l, r),
mode='constant',
constant_values=constant_values)
else:
#Ensure time resolution adjustement between audio and mel-spectrogram
pad = audio.librosa_pad_lr(wav, hparams.n_fft,
audio.get_hop_size(hparams))
#Reflect pad audio signal (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, pad, mode='reflect')
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
# Write the spectrogram and audio to disk
audio_filename = 'audio-{}.npy'.format(index)
mel_filename = 'mel-{}.npy'.format(index)
linear_filename = 'linear-{}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
# Return a tuple describing this training example
return (audio_filename, mel_filename, linear_filename, refer, time_steps,
mel_frames, ppgs, speaker, lf0_path)
def _process_utterance(wav_dir, mel_dir, index, wav_path, text, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print(
'file {} present in csv metadata is not present in wav folder. skipping!'
.format(wav_path))
return None
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#[-1, 1]
out = encode_mu_law(wav, mu=512)
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames or len(
text) > hparams.max_text_length:
return None
#Zero pad for quantized signal
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
r = mel_frames * audio.get_hop_size(hparams) - len(wav)
out = np.pad(out, (0, r), mode='constant', constant_values=0.)
assert len(out) == mel_frames * audio.get_hop_size(hparams)
time_steps = len(out)
# Write the spectrogram and audio to disk
filename = '{}.npy'.format(index)
np.save(os.path.join(wav_dir, filename),
out.astype(np.int16),
allow_pickle=False)
np.save(os.path.join(mel_dir, filename),
mel_spectrogram.T,
allow_pickle=False)
# Return a tuple describing this training example
return (filename, time_steps, mel_frames, text)
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: #catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(
wav_path))
return None
#rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
#Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
#[0, quantize_channels)
out = 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 = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
#[-1, 1]
out = mulaw(wav, hparams.quantize_channels)
constant_values = mulaw(0., hparams.quantize_channels)
out_dtype = np.float32
else:
#[-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
return None
#Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(wav, hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
#sanity check
assert linear_frames == mel_frames
#Ensure time resolution adjustement between audio and mel-spectrogram
fft_size = hparams.n_fft if hparams.win_size is None else hparams.win_size
l, r = audio.pad_lr(wav, fft_size, audio.get_hop_size(hparams))
#Zero pad for quantized signal
out = np.pad(out, (l, r), mode='constant', constant_values=constant_values)
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
# Write the spectrogram and audio to disk
audio_filename = 'speech-audio-{:05d}.npy'.format(index)
mel_filename = 'speech-mel-{:05d}.npy'.format(index)
linear_filename = 'speech-linear-{:05d}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename), out.astype(out_dtype), allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename), linear_spectrogram.T, allow_pickle=False)
# Return a tuple describing this training example
return (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, text)
def _process_utterance_clova(audio_dir, label_dir, index, wav_path, text_path, args):
"""
Preprocesses a single utterance wav/text_jamo pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text_jamo: text_jamo spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text_jamo)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=args.sample_rate)
except FileNotFoundError: #catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(wav_path))
return None
# rescale wav
if args.rescale:
wav = wav / np.abs(wav).max() * args.rescaling_max
# M-AILABS extra silence specific
if args.trim_silence:
wav = audio.trim_silence(wav, args)
# [-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, args).astype(out_dtype)
mel_frames = mel_spectrogram.shape[1]
# Ensure time resolution adjustement between audio and mel-spectrogram
pad = audio.librosa_pad_lr(wav, args.n_fft, audio.get_hop_size(args))
# Reflect pad audio signal (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, (0, pad), mode='reflect')
assert len(out) >= mel_frames * audio.get_hop_size(args)
# time resolution adjustment
# ensure length of raw audio is multiple of hop size so that we can use
# transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(args)]
assert len(out) % audio.get_hop_size(args) == 0
time_steps = len(out)
# text_jamo sequence
with open(text_path, 'r', encoding='utf-8', newline='') as f:
rdr = csv.reader(f)
for x in rdr:
if os.path.basename(wav_path) == x[0]:
line = x[1]
# ETRI transcription rule
line = sentence_filter(line).upper()
label_sequence = normalize(line)
print(label_sequence)
# Write the spectrogram and audio to disk
mel_filename = 'mel-{}.npy'.format(index)
label_filename = 'label-{}.txt'.format(index)
np.save(os.path.join(audio_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
with open(os.path.join(label_dir, label_filename), 'w', encoding='utf-8') as f_out:
f_out.write(label_sequence)
# Return a tuple describing this training example
return (wav_path, text_path, mel_filename, label_filename, time_steps, mel_frames)
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path)
except :
print('file {} present in csv not in folder'.format(
wav_path))
return None
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
#M-AILABS extra silence specific
if hparams.trim_silence:
wav = audio.trim_silence(wav)
#[0, quantize_channels)
out = 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 = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
#Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(wav).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
#sanity check
assert linear_frames == mel_frames
#Ensure time resolution adjustement between audio and mel-spectrogram
l, r = audio.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)
time_steps = len(out)
assert time_steps >= mel_frames * audio.get_hop_size()
#time resolution adjustement
#ensure length of raw audio is multiple of hop size so that we can use
#transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size()]
assert time_steps % audio.get_hop_size() == 0
# Write the spectrogram and audio to disk
audio_filename = 'speech-audio-{:05d}.npy'.format(index)
mel_filename = 'speech-mel-{:05d}.npy'.format(index)
linear_filename = 'speech-linear-{:05d}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename), out.astype(out_dtype), allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename), mel_spectrogram.T, allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename), linear_spectrogram.T, allow_pickle=False)
# Return a tuple describing this training example
return (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, text)
Args:
- mel_dir: the directory to write the mel spectograms into
- linear_dir: the directory to write the linear spectrograms into
- wav_dir: the directory to write the preprocessed wav into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
eliminated=0
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: # catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(
wav_path))
return None
try:
# rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
# M-AILABS extra silence specific
if hparams.trim_silence:
new_wav = audio.trim_silence(wav, hparams)
eliminated+=(len(wav)-len(new_wav))/hparams.sample_rate
except Exception as e:
def _process_utterance(out_dir, index, wav_path, text, hparams):
"""
Preprocesses a single utterance wav/text pair
this writes the mel scale spectogram to disk and return a tuple to write
to the train.txt file
Args:
- out_dir: the directory to write the msgpack into
- index: the numeric index to use in the spectogram filename
- wav_path: path to the audio file containing the speech input
- text: text spoken in the input audio file
- hparams: hyper parameters
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
"""
try:
# Load the audio as numpy array
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
except FileNotFoundError: # catch missing wav exception
print('file {} present in csv metadata is not present in wav folder. skipping!'.format(
wav_path))
return None
# Trim lead/trail silences
if hparams.trim_silence:
wav = audio.trim_silence(wav, hparams)
# Pre-emphasize
preem_wav = audio.preemphasis(wav, hparams.preemphasis, hparams.preemphasize)
# rescale wav
if hparams.rescale:
wav = wav / np.abs(wav).max() * hparams.rescaling_max
preem_wav = preem_wav / np.abs(preem_wav).max() * hparams.rescaling_max
# Assert all audio is in [-1, 1]
if (wav > 1.).any() or (wav < -1.).any():
raise RuntimeError('wav has invalid value: {}'.format(wav_path))
if (preem_wav > 1.).any() or (preem_wav < -1.).any():
raise RuntimeError('wav has invalid value: {}'.format(wav_path))
# [-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(preem_wav, hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
return None
# Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(preem_wav, hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
# sanity check
assert linear_frames == mel_frames
# Ensure time resolution adjustement between audio and mel-spectrogram
l_pad, r_pad = audio.librosa_pad_lr(wav, audio.get_hop_size(hparams), hparams.pad_sides)
# Reflect pad audio signal on the right (Just like it's done in Librosa to avoid frame inconsistency)
out = np.pad(out, (l_pad, r_pad), mode='constant', constant_values=constant_values)
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
# time resolution adjustement
# ensure length of raw audio is multiple of hop size so that we can use
# transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
npz_filename = '{}.npz'.format(index)
r = hparams.outputs_per_step
if hparams.symmetric_mels:
_pad_value = -hparams.max_abs_value
else:
_pad_value = 0.
# +2r for head and tail silence
mel_spec = np.pad(mel_spectrogram.T, [[r, r], [0, 0]], 'constant', constant_values=_pad_value)
linear_spec = np.pad(linear_spectrogram.T, [[r, r], [0, 0]], 'constant', constant_values=_pad_value)
target_length = len(linear_spec)
target_frames = (target_length // r + 1) * r
num_pad = target_frames - target_length
if num_pad != 0:
linear_spec = np.pad(linear_spec, ((0, num_pad), (0, 0)), "constant", constant_values=_pad_value)
mel_spec = np.pad(mel_spec, ((0, num_pad), (0, 0)), "constant", constant_values=_pad_value)
stop_token = np.concatenate(
[np.zeros(target_frames - 1, dtype=np.float32), np.ones(1, dtype=np.float32)],
axis=0)
data = {
'mel': mel_spec,
'linear': linear_spec,
'audio': out.astype(out_dtype),
'input_data': np.asarray(text_to_sequence(text)),
'time_steps': time_steps,
'mel_frames': target_frames,
'text': text,
'stop_token': stop_token,
}
dumps_msgpack(data, os.path.join(out_dir, npz_filename))
# Return a tuple describing this training example
return npz_filename, time_steps, mel_frames, text
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text,
hparams):
'''Preprocesses a single utterance audio/text pair.
This writes the mel and linear scale spectrograms to disk and returns a tuple to write
to the train.txt file.
Args:
out_dir: The directory to write the spectrograms into
index: The numeric index to use in the spectrogram filenames.
wav_path: Path to the audio file containing the speech input
text: The text spoken in the input audio file
Returns:
- A tuple: (audio_filename, mel_filename, linear_filename, time_steps, mel_frames, linear_frames, text)
'''
# Load the audio to a numpy array:
wav = audio.load_wav(wav_path, sr=hparams.sample_rate)
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = 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 = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = mulaw(wav, hparams.quantize_channels)
constant_values = mulaw(0., hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
# Compute the mel scale spectrogram from the wav
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(np.float32)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
return None
# Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(wav,
hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
# sanity check
assert linear_frames == mel_frames
# Ensure time resolution adjustement between audio and mel-spectrogram
fft_size = hparams.n_fft if hparams.win_size is None else hparams.win_size
l, r = audio.pad_lr(wav, fft_size, audio.get_hop_size(hparams))
# Zero pad for quantized signal
out = np.pad(out, (l, r), mode='constant', constant_values=constant_values)
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
# time resolution adjustement
# ensure length of raw audio is multiple of hop size so that we can use
# transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
# Write the spectrogram and audio to disk
audio_filename = 'speech-audio-{:05d}.npy'.format(index)
mel_filename = 'speech-mel-{:05d}.npy'.format(index)
linear_filename = 'speech-linear-{:05d}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
# Return a tuple describing this training example
return (audio_filename, mel_filename, linear_filename, time_steps,
mel_frames, text)
def _process_utterance(mel_dir, linear_dir, wav_dir, index, wav_path, text,
hparams):
wav = _trim_wav(audio.load_wav(wav_path, sr=hparams.sample_rate))
# Mu-law quantize
if is_mulaw_quantize(hparams.input_type):
# [0, quantize_channels)
out = 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 = mulaw_quantize(0, hparams.quantize_channels)
out_dtype = np.int16
elif is_mulaw(hparams.input_type):
# [-1, 1]
out = mulaw(wav, hparams.quantize_channels)
constant_values = mulaw(0., hparams.quantize_channels)
out_dtype = np.float32
else:
# [-1, 1]
out = wav
constant_values = 0.
out_dtype = np.float32
mel_spectrogram = audio.melspectrogram(wav, hparams).astype(np.float32)
name = os.path.splitext(os.path.basename(wav_path))[0]
speaker_id = _speaker_re.match(name).group(1)
mel_frames = mel_spectrogram.shape[1]
if mel_frames > hparams.max_mel_frames and hparams.clip_mels_length:
return None
# Compute the linear scale spectrogram from the wav
linear_spectrogram = audio.linearspectrogram(wav,
hparams).astype(np.float32)
linear_frames = linear_spectrogram.shape[1]
# sanity check
assert linear_frames == mel_frames
# Ensure time resolution adjustement between audio and mel-spectrogram
fft_size = hparams.n_fft if hparams.win_size is None else hparams.win_size
l, r = audio.pad_lr(wav, fft_size, audio.get_hop_size(hparams))
# Zero pad for quantized signal
out = np.pad(out, (l, r), mode='constant', constant_values=constant_values)
assert len(out) >= mel_frames * audio.get_hop_size(hparams)
# time resolution adjustement
# ensure length of raw audio is multiple of hop size so that we can use
# transposed convolution to upsample
out = out[:mel_frames * audio.get_hop_size(hparams)]
assert len(out) % audio.get_hop_size(hparams) == 0
time_steps = len(out)
# Write the spectrogram and audio to disk
audio_filename = 'speech-audio-{:05d}.npy'.format(index)
mel_filename = 'speech-mel-{:05d}.npy'.format(index)
linear_filename = 'speech-linear-{:05d}.npy'.format(index)
np.save(os.path.join(wav_dir, audio_filename),
out.astype(out_dtype),
allow_pickle=False)
np.save(os.path.join(mel_dir, mel_filename),
mel_spectrogram.T,
allow_pickle=False)
np.save(os.path.join(linear_dir, linear_filename),
linear_spectrogram.T,
allow_pickle=False)
return (audio_filename, mel_filename, linear_filename, time_steps,
mel_frames, speaker_id, text)
