def get_file_length(filepath):
"""
Returns the length of the sequence in the file specified by `filepath`
"""
signal_info, encoding_info = torchaudio.info(filepath)
return signal_info.length
def __getitem__(self, index):
"""
:return:
"""
# Check the size of the file
current_session = self.sessions.iloc[index]
# TODO is this required ?
nfo = torchaudio.info(
f"{self.data_path}/{current_session['file_id']}{self.data_file_extension}"
)
original_start = int(current_session['start'])
if self.overlap > 0:
lowest_shift = self.overlap / 2
highest_shift = self.overlap / 2
if original_start < (current_session['file_start'] *
self.sample_rate + self.sample_number / 2):
lowest_shift = int(original_start -
current_session['file_start'] *
self.sample_rate)
if original_start + self.sample_number > (
current_session['file_start'] +
current_session['file_duration']
) * self.sample_rate - self.sample_number / 2:
highest_shift = int((current_session['file_start'] +
current_session['file_duration']) *
self.sample_rate -
(original_start + self.sample_number))
start_frame = original_start + int(
random.uniform(-lowest_shift, highest_shift))
else:
start_frame = original_start
conversion_rate = nfo.sample_rate // self.sample_rate
if start_frame + conversion_rate * self.sample_number >= nfo.num_frames:
start_frame = numpy.min(nfo.num_frames -
conversion_rate * self.sample_number - 1)
speech, speech_fs = torchaudio.load(
f"{self.data_path}/{current_session['file_id']}{self.data_file_extension}",
frame_offset=conversion_rate * start_frame,
num_frames=conversion_rate * self.sample_number)
if nfo.sample_rate != self.sample_rate:
speech = torchaudio.transforms.Resample(
nfo.sample_rate, self.sample_rate).forward(speech)
speech += 10e-6 * torch.randn(speech.shape)
if len(self.transform) > 0:
speech = data_augmentation(speech,
self.sample_rate,
self.transform,
self.transform_number,
noise_df=self.noise_df,
rir_df=self.rir_df)
speaker_idx = current_session["speaker_idx"]
if self.output_format == "pytorch":
return speech, torch.tensor(speaker_idx)
else:
return speech, speaker_idx
def __init__(self, root_dir=''):
super(Aff2CompDataset, self).__init__()
self.video_dir = root_dir
self.extracted_dir = os.path.join(self.video_dir, 'extracted')
self.clip_len = 8
self.input_size = (112, 112)
self.dilation = 6
self.label_frame = self.clip_len * self.dilation
# audio params
self.window_size = 20e-3
self.window_stride = 10e-3
self.sample_rate = 44100
num_fft = 2**math.ceil(math.log2(self.window_size * self.sample_rate))
window_fn = torch.hann_window
self.sample_len_secs = 10
self.sample_len_frames = self.sample_len_secs * self.sample_rate
self.audio_shift_sec = 5
self.audio_shift_samples = self.audio_shift_sec * self.sample_rate
# transforms
self.audio_transform = torchaudio.transforms.MelSpectrogram(
sample_rate=self.sample_rate,
n_mels=64,
n_fft=num_fft,
win_length=int(self.window_size * self.sample_rate),
hop_length=int(self.window_stride * self.sample_rate),
window_fn=window_fn)
self.audio_spec_transform = ComposeWithInvert(
[AmpToDB(), Normalize(mean=[-14.8], std=[19.895])])
self.clip_transform = ComposeWithInvert([
NumpyToTensor(),
Normalize(mean=[0.43216, 0.394666, 0.37645, 0.5],
std=[0.22803, 0.22145, 0.216989, 0.225])
])
all_videos = find_all_video_files(self.video_dir)
self.cached_metadata_path = os.path.join(self.video_dir, 'dataset.pkl')
if not os.path.isfile(self.cached_metadata_path):
print('creating cached_metadata... ')
self.image_path = [] # paths relative to self.extracted_dir
self.video_id = []
self.frame_id = []
self.label_au = []
self.label_ex = []
self.label_va = []
self.train_ids = []
self.val_ids = []
self.test_ids = []
self.features = []
self.feature_names = []
self.time_stamps = []
self.mask_available = False
self.video_db_nr = []
video_db_nr = 0
for video in tqdm(all_videos):
meta = Video(video).meta
meta['filename'] = get_filename(video)
meta['path'] = get_path(video)
meta['extension'] = get_extension(video)
num_frames_video = meta['num_frames']
audio_file = os.path.splitext(video)[0] + '.wav'
si, ei = torchaudio.info(audio_file)
assert si.rate == 44100
video_ts_file = os.path.join(
meta['path'], meta['filename'] + '_video_ts.txt')
if os.path.isfile(video_ts_file):
pass
else:
mkvfile = os.path.join(meta['path'], 'temp.mkv')
videofile = os.path.join(
meta['path'], meta['filename'] + meta['extension'])
command = 'mkvmerge -o ' + mkvfile + ' ' + videofile
subprocess.call(command, shell=True)
command = 'mkvextract ' + mkvfile + ' timestamps_v2 0:' + video_ts_file
subprocess.call(command, shell=True)
os.remove(mkvfile)
with open(video_ts_file, 'r') as f:
time_stamps = np.genfromtxt(f)[:num_frames_video]
#os.remove(video_ts_file)
self.mask_available = True
extracted_dir = os.path.join(self.extracted_dir,
meta['filename'])
splits = []
if 'AU' in meta:
au_split = meta['AU']
splits.append(au_split)
if 'EX' in meta:
ex_split = meta['EX']
splits.append(ex_split)
if 'VA' in meta:
va_split = meta['VA']
splits.append(va_split)
splits = list(set(splits)) # UPDATED 03.06.2020 (was missing)
for split in splits:
self.time_stamps.append(time_stamps)
for image_filename in sorted(os.listdir(extracted_dir)):
if os.path.isdir(
os.path.join(extracted_dir, image_filename)):
continue
# path relative to self.extracted_dir
self.image_path.append(
os.path.relpath(
os.path.join(extracted_dir, image_filename),
self.extracted_dir))
self.video_id.append(meta['filename'])
self.video_db_nr.append(
video_db_nr
) # UPDATED 03.06.2020 (avoids using frames from neighbour videos)
frame_id = int(os.path.splitext(image_filename)[0])
self.frame_id.append(frame_id)
# add your own label loading here if you want to use this for training
self.label_au.append(None)
self.label_ex.append(None)
self.label_va.append(None)
self.train_ids.append(1 if split == 'train' else 0)
self.val_ids.append(1 if split == 'val' else 0)
self.test_ids.append(1 if split == 'test' else 0)
video_db_nr += 1
self.frame_id = np.stack(self.frame_id)
self.label_au = np.stack(self.label_au)
self.label_ex = np.stack(self.label_ex)
self.label_va = np.stack(self.label_va)
self.train_ids = np.stack(self.train_ids)
self.val_ids = np.stack(self.val_ids)
self.test_ids = np.stack(self.test_ids)
self.time_stamps = np.hstack(self.time_stamps)
with open(self.cached_metadata_path, 'wb') as f:
pickle.dump(
{
'frame_id': self.frame_id,
'label_au': self.label_au,
'label_ex': self.label_ex,
'label_va': self.label_va,
'video_id': self.video_id,
'image_path': self.image_path,
'train_ids': self.train_ids,
'val_ids': self.val_ids,
'test_ids': self.test_ids,
'time_stamps': self.time_stamps,
'mask_available': self.mask_available,
'video_db_nr': self.video_db_nr
}, f)
else:
with open(self.cached_metadata_path, 'rb') as f:
meta = pickle.load(f)
self.frame_id = meta['frame_id']
self.label_au = meta['label_au']
self.label_ex = meta['label_ex']
self.label_va = meta['label_va']
self.video_id = meta['video_id']
self.image_path = meta['image_path']
self.train_ids = meta['train_ids']
self.val_ids = meta['val_ids']
self.time_stamps = meta['time_stamps']
self.mask_available = meta['mask_available']
self.test_ids = meta['test_ids']
self.video_db_nr = meta['video_db_nr']
self.validation_video_ids()
self.test_video_ids()
self.use_mask = self.mask_available
def py_info_func(filepath: str) -> torch.classes.torchaudio.SignalInfo:
return torchaudio.info(filepath)
def inspect_file(path):
print("-" * 10)
print("Source:", path)
print("-" * 10)
print(f" - File size: {os.path.getsize(path)} bytes")
print_metadata(torchaudio.info(path))
def py_info_func(
filepath: str) -> torchaudio.backend.sox_io_backend.AudioMetaData:
return torchaudio.info(filepath)
def d(ID):
info = torchaudio.info(
f'/work3/s164419/01005WakeWordData/lectures/{ID}.wav')
return info.num_frames / info.sample_rate
def prepare_librispeech(
corpus_dir: Pathlike,
dataset_parts: Optional[Tuple[str]] = dataset_parts_mini,
output_dir: Optional[Pathlike] = None
) -> Dict[str, Dict[str, Union[RecordingSet, SupervisionSet]]]:
"""
Returns the manifests which consist of the Recordings and Supervisions
:param corpus_dir: Pathlike, the path of the data dir.
:param dataset_parts: dataset part name, e.g. 'train-clean-100', 'train-clean-5', 'dev-clean'
:param output_dir: Pathlike, the path where to write the manifests.
:return: a Dict whose key is the dataset part, and the value is Dicts with the keys 'audio' and 'supervisions'.
"""
corpus_dir = Path(corpus_dir)
assert corpus_dir.is_dir(), f'No such directory: {corpus_dir}'
if output_dir is not None:
output_dir = Path(output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
manifests = defaultdict(dict)
for part in dataset_parts:
# Generate a mapping: utt_id -> (audio_path, audio_info, text)
metadata = {}
part_path = corpus_dir / part
for trans_path in part_path.rglob('*.txt'):
with open(trans_path) as f:
for line in f:
idx, text = line.split(maxsplit=1)
audio_path = part_path / Path(idx.replace(
'-', '/')).parent / f'{idx}.flac'
if audio_path.is_file():
# info[0]: info of the raw audio (e.g. channel number, sample rate, duration ... )
# info[1]: info about the encoding (e.g. FLAC/ALAW/ULAW ...)
info = torchaudio.info(str(audio_path))
metadata[idx] = LibriSpeechMetaData(
audio_path=audio_path,
audio_info=info[0],
text=text)
else:
logging.warning(f'No such file: {audio_path}')
# Audio
audio = RecordingSet.from_recordings(
Recording(id=idx,
sources=[
AudioSource(type='file',
channels=[0],
source=str(metadata[idx].audio_path))
],
sampling_rate=int(metadata[idx].audio_info.rate),
num_samples=metadata[idx].audio_info.length,
duration=(metadata[idx].audio_info.length /
metadata[idx].audio_info.rate))
for idx in metadata)
# Supervision
supervision = SupervisionSet.from_segments(
SupervisionSegment(id=idx,
recording_id=idx,
start=0.0,
duration=audio.recordings[idx].duration,
channel=0,
language='English',
speaker=re.sub(r'-.*', r'', idx),
text=metadata[idx].text.strip())
for idx in audio.recordings)
if output_dir is not None:
supervision.to_json(output_dir / f'supervisions_{part}.json')
audio.to_json(output_dir / f'recordings_{part}.json')
manifests[part] = {'recordings': audio, 'supervisions': supervision}
return manifests
def get_file_info(p):
if isinstance(p, Path): p = p.as_posix()
signal_info, _ = torchaudio.info(p)
return signal_info
def test_get_info(self):
input_path = os.path.join(self.test_dirpath, 'assets', 'sinewave.wav')
info_expected = (1, 64000, 16000, 32)
info_load = torchaudio.info(input_path)
self.assertEqual(info_load, info_expected)
if args.segments is None:
audio_list = load_wav_scp(args.wav_scp)
else:
audio_list = load_wav_segments(args.wav_scp, args.segments)
count = 0
with open(args.out_ark, 'wb') as ark_fout, \
open(args.out_scp, 'w', encoding='utf8') as scp_fout:
for item in audio_list:
if len(item) == 2:
key, wav_path = item
waveform, sample_rate = torchaudio.load_wav(wav_path)
else:
assert len(item) == 4
key, wav_path, start, end = item
sample_rate = torchaudio.info(wav_path).sample_rate
frame_offset = int(start * sample_rate)
num_frames = int((end - start) * sample_rate)
waveform, sample_rate = torchaudio.load_wav(
wav_path, frame_offset, num_frames)
mat = kaldi.fbank(waveform,
num_mel_bins=args.num_mel_bins,
frame_length=args.frame_length,
frame_shift=args.frame_shift,
dither=args.dither,
energy_floor=0.0,
sample_frequency=sample_rate)
mat = mat.detach().numpy()
kaldi_io.write_ark_scp(key, mat, ark_fout, scp_fout)
count += 1
def prepare_ami(
data_dir: Pathlike,
output_dir: Optional[Pathlike] = None,
) -> Dict[str, Dict[str, Union[RecordingSet, SupervisionSet]]]:
"""
Returns the manifests which consist of the Recordings and Supervisions
:param data_dir: Pathlike, the path of the data dir.
:param output_dir: Pathlike, the path where to write the manifests.
:return: a Dict whose key is ('train', 'dev', 'eval'), and the value is Dicts with keys 'audio' and 'supervisions'.
"""
data_dir = Path(data_dir)
assert data_dir.is_dir(), f'No such directory: {data_dir}'
if output_dir is not None:
output_dir = Path(output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
anotation_lists = parse_ami_annotations(data_dir / 'annotations.gzip')
# Create a mapping from a tuple of (session_id, channel) to the list of annotations.
# This way we can map the supervisions to the right channels in a multi-channel recording.
annotation_by_id_and_channel = {
(filename.split('.')[0], int(filename[-5])): annotations
for filename, annotations in anotation_lists.items()
}
wav_dir = data_dir / 'wav_db'
audio_paths = wav_dir.rglob('*.wav')
# Group together multiple channels from the same session.
# We will use that to create a Recording with multiple sources (channels).
from cytoolz import groupby
channel_wavs = groupby(lambda p: p.parts[-3], audio_paths)
manifests = defaultdict(dict)
for part in dataset_parts:
# Audio
recordings = []
for session_name, channel_paths in channel_wavs.items():
if session_name not in dataset_parts[part]:
continue
audio_info = torchaudio.info(str(channel_paths[0]))[0]
recordings.append(Recording(
id=session_name,
sources=[
AudioSource(
type='file',
channels=[idx],
source=str(audio_path)
)
for idx, audio_path in enumerate(sorted(channel_paths))
],
sampling_rate=int(audio_info.rate),
num_samples=audio_info.length,
duration=audio_info.length / audio_info.rate,
))
audio = RecordingSet.from_recordings(recordings)
# Supervisions
segments_by_pause = []
for recording in audio:
for source in recording.sources:
# In AMI "source.channels" will always be a one-element list
channel, = source.channels
anotation = annotation_by_id_and_channel.get((recording.id, channel))
if anotation is None:
logging.warning(f'No annotation found for recording "{recording.id}" channel {channel} '
f'(file {source.source})')
continue
for seg_idx, seg_info in enumerate(anotation):
for subseg_idx, subseg_info in enumerate(seg_info):
duration = subseg_info.end_time - subseg_info.begin_time
if duration > 0:
segments_by_pause.append(SupervisionSegment(
id=f'{recording.id}-{seg_idx}-{subseg_idx}',
recording_id=recording.id,
start=subseg_info.begin_time,
duration=duration,
channel=channel,
language='English',
speaker=subseg_info.speaker,
gender=subseg_info.gender,
text=subseg_info.text
))
supervision = SupervisionSet.from_segments(segments_by_pause)
if output_dir is not None:
audio.to_json(output_dir / f'recordings_{part}.json')
supervision.to_json(output_dir / f'supervisions_{part}.json')
manifests[part] = {
'recordings': audio,
'supervisions': supervision
}
return manifests
def duration(self):
if self._sample_rate is not None:
return len(self.data_signal) / self.sample_rate
else:
si, ei = torchaudio.info(str(self.path))
return si.length / si.rate
def sample_rate(self):
if not hasattr(self, '_sample_rate') or self._sample_rate is None:
# Gets metadata from an audio file without loading the signal.
si, ei = torchaudio.info(str(self.path))
self._sample_rate = si.rate
return self._sample_rate
def prepare_ljspeech(
corpus_dir: Pathlike,
output_dir: Optional[Pathlike] = None
) -> Dict[str, Union[RecordingSet, SupervisionSet]]:
"""
Returns the manifests which consist of the Recordings and Supervisions
:param corpus_dir: Pathlike, the path of the data dir.
:param output_dir: Pathlike, the path where to write the manifests.
:return: The RecordingSet and SupervisionSet with the keys 'audio' and 'supervisions'.
"""
corpus_dir = Path(corpus_dir)
assert corpus_dir.is_dir(), f'No such directory: {corpus_dir}'
if output_dir is not None:
output_dir = Path(output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
# Generate a mapping: utt_id -> (audio_path, audio_info, text)
metadata_csv_path = corpus_dir / 'metadata.csv'
assert metadata_csv_path.is_file(), f'No such file: {metadata_csv_path}'
metadata = {}
with open(metadata_csv_path) as f:
for line in f:
idx, text, _ = line.split('|')
audio_path = corpus_dir / 'wavs' / f'{idx}.wav'
if audio_path.is_file():
# info[0]: info of the raw audio (e.g. channel number, sample rate, duration ... )
# info[1]: info about the encoding (e.g. FLAC/ALAW/ULAW ...)
info = torchaudio.info(str(audio_path))
metadata[idx] = LJSpeechMetaData(audio_path=audio_path, audio_info=info[0], text=text)
else:
logging.warning(f'No such file: {audio_path}')
# Audio
audio = RecordingSet.from_recordings(
Recording(
id=idx,
sources=[
AudioSource(
type='file',
channels=[0],
source=str(metadata[idx].audio_path)
)
],
sampling_rate=int(metadata[idx].audio_info.rate),
num_samples=metadata[idx].audio_info.length,
duration=(metadata[idx].audio_info.length / metadata[idx].audio_info.rate)
)
for idx in metadata
)
# Supervision
supervision = SupervisionSet.from_segments(
SupervisionSegment(
id=idx,
recording_id=idx,
start=0.0,
duration=audio.recordings[idx].duration,
channel=0,
language='English',
gender='female',
text=metadata[idx].text
)
for idx in audio.recordings
)
if output_dir is not None:
supervision.to_json(output_dir / 'supervisions.json')
audio.to_json(output_dir / 'audio.json')
return {'audio': audio, 'supervisions': supervision}
def duration(self):
if (self.sig is not None): return self.nsamples / self.sr
else:
si, ei = torchaudio.info(str(self.path))
return si.length / si.rate
def prepare_mobvoihotwords(
corpus_dir: Pathlike,
output_dir: Optional[Pathlike] = None
) -> Dict[str, Dict[str, Union[RecordingSet, SupervisionSet]]]:
"""
Returns the manifests which consist of the Recordings and Supervisions
:param corpus_dir: Pathlike, the path of the data dir.
:param output_dir: Pathlike, the path where to write the manifests.
:return: a Dict whose key is the dataset part, and the value is Dicts with the keys 'audio' and 'supervisions'.
"""
corpus_dir = Path(corpus_dir)
assert corpus_dir.is_dir(), f'No such directory: {corpus_dir}'
if output_dir is not None:
output_dir = Path(output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
manifests = defaultdict(dict)
dataset_parts = ['train', 'dev', 'test']
for part in dataset_parts:
# Generate a mapping: utt_id -> (audio_path, audio_info, speaker, text)
metadata = {}
for prefix in ['p_', 'n_']:
prefixed_part = prefix + part
json_path = corpus_dir / 'mobvoi_hotword_dataset_resources' / f'{prefixed_part}.json'
with open(json_path, 'r', encoding='utf-8') as f:
json_data = json.load(f)
for entry in json_data:
idx = entry['utt_id']
speaker = idx if entry['speaker_id'] is None else entry['speaker_id']
audio_path = corpus_dir / 'mobvoi_hotword_dataset' / f'{idx}.wav'
text = 'FREETEXT'
if entry['keyword_id'] == 0:
text = 'HiXiaowen'
elif entry['keyword_id'] == 1:
text = 'NihaoWenwen'
else:
assert entry['keyword_id'] == -1
if audio_path.is_file():
# info[0]: info of the raw audio (e.g. channel number, sample rate, duration ... )
# info[1]: info about the encoding (e.g. FLAC/ALAW/ULAW ...)
info = torchaudio.info(str(audio_path))
metadata[idx] = MobvoiHotwordsMetaData(
audio_path=audio_path, audio_info=info[0], speaker=speaker, text=text
)
else:
logging.warning(f'No such file: {audio_path}')
# Audio
audio = RecordingSet.from_recordings(
Recording(
id=idx,
sources=[
AudioSource(
type='file',
channels=[0],
source=str(metadata[idx].audio_path)
)
],
sampling_rate=int(metadata[idx].audio_info.rate),
num_samples=metadata[idx].audio_info.length,
duration=(metadata[idx].audio_info.length / metadata[idx].audio_info.rate)
)
for idx in metadata
)
# Supervision
supervision = SupervisionSet.from_segments(
SupervisionSegment(
id=idx,
recording_id=idx,
start=0.0,
duration=audio.recordings[idx].duration,
channel=0,
language='Chinese',
speaker=metadata[idx].speaker,
text=metadata[idx].text.strip()
)
for idx in audio.recordings
)
if output_dir is not None:
supervision.to_json(output_dir / f'supervisions_{part}.json')
audio.to_json(output_dir / f'recordings_{part}.json')
manifests[part] = {
'recordings': audio,
'supervisions': supervision
}
return manifests
###################################################################### # Audio I/O # ========= # # torchaudio integrates ``libsox`` and provides a rich set of audio I/O. # ###################################################################### # Quering audio metadata # ---------------------- # # ``torchaudio.info`` function fetches metadata of audio. You can provide # a path-like object or file-like object. # metadata = torchaudio.info(SAMPLE_WAV_PATH) print_metadata(metadata, src=SAMPLE_WAV_PATH) ###################################################################### # Where # # - ``sample_rate`` is the sampling rate of the audio # - ``num_channels`` is the number of channels # - ``num_frames`` is the number of frames per channel # - ``bits_per_sample`` is bit depth # - ``encoding`` is the sample coding format # # The values ``encoding`` can take are one of the following # # - ``"PCM_S"``: Signed integer linear PCM # - ``"PCM_U"``: Unsigned integer linear PCM
def extractLength(couple):
speaker, locPath = couple
info = torchaudio.info(str(locPath))[0]
return info.length
def create_csv(
orig_tsv_file,
csv_file,
data_folder,
accented_letters=False,
language="en",
):
"""
Creates the csv file given a list of wav files.
Arguments
---------
orig_tsv_file : str
Path to the Common Voice tsv file (standard file).
data_folder : str
Path of the CommonVoice dataset.
accented_letters : bool, optional
Defines if accented letters will be kept as individual letters or
transformed to the closest non-accented letters.
Returns
-------
None
"""
# Check if the given files exists
if not os.path.isfile(orig_tsv_file):
msg = "\t%s doesn't exist, verify your dataset!" % (orig_tsv_file)
logger.info(msg)
raise FileNotFoundError(msg)
# We load and skip the header
loaded_csv = open(orig_tsv_file, "r").readlines()[1:]
nb_samples = str(len(loaded_csv))
msg = "Preparing CSV files for %s samples ..." % (str(nb_samples))
logger.info(msg)
# Adding some Prints
msg = "Creating csv lists in %s ..." % (csv_file)
logger.info(msg)
csv_lines = [["ID", "duration", "wav", "spk_id", "wrd"]]
# Start processing lines
total_duration = 0.0
for line in tzip(loaded_csv):
line = line[0]
# Path is at indice 1 in Common Voice tsv files. And .mp3 files
# are located in datasets/lang/clips/
mp3_path = data_folder + "/clips/" + line.split("\t")[1]
file_name = mp3_path.split(".")[-2].split("/")[-1]
spk_id = line.split("\t")[0]
snt_id = file_name
# Reading the signal (to retrieve duration in seconds)
if os.path.isfile(mp3_path):
info = torchaudio.info(mp3_path)
else:
msg = "\tError loading: %s" % (str(len(file_name)))
logger.info(msg)
continue
duration = info.num_frames / info.sample_rate
total_duration += duration
# Getting transcript
words = line.split("\t")[2]
# !! Language specific cleaning !!
# Important: feel free to specify the text normalization
# corresponding to your alphabet.
if language in ["en", "fr", "it", "rw"]:
words = re.sub("[^'A-Za-z0-9À-ÖØ-öø-ÿЀ-ӿ]+", " ", words).upper()
elif language == "ar":
HAMZA = "\u0621"
ALEF_MADDA = "\u0622"
ALEF_HAMZA_ABOVE = "\u0623"
letters = ("ابتةثجحخدذرزسشصضطظعغفقكلمنهويءآأؤإئ" + HAMZA +
ALEF_MADDA + ALEF_HAMZA_ABOVE)
words = re.sub("[^" + letters + "]+", " ", words).upper()
elif language == "ga-IE":
# Irish lower() is complicated, but upper() is nondeterministic, so use lowercase
def pfxuc(a):
return len(a) >= 2 and a[0] in "tn" and a[1] in "AEIOUÁÉÍÓÚ"
def galc(w):
return w.lower(
) if not pfxuc(w) else w[0] + "-" + w[1:].lower()
words = re.sub("[^-A-Za-z'ÁÉÍÓÚáéíóú]+", " ", words)
words = " ".join(map(galc, words.split(" ")))
# Remove accents if specified
if not accented_letters:
nfkd_form = unicodedata.normalize("NFKD", words)
words = "".join(
[c for c in nfkd_form if not unicodedata.combining(c)])
words = words.replace("'", " ")
# Remove multiple spaces
words = re.sub(" +", " ", words)
# Remove spaces at the beginning and the end of the sentence
words = words.lstrip().rstrip()
# Getting chars
chars = words.replace(" ", "_")
chars = " ".join([char for char in chars][:])
# Remove too short sentences (or empty):
if len(words) < 3:
continue
# Composition of the csv_line
csv_line = [snt_id, str(duration), mp3_path, spk_id, str(words)]
# Adding this line to the csv_lines list
csv_lines.append(csv_line)
# Writing the csv lines
with open(csv_file, mode="w", encoding="utf-8") as csv_f:
csv_writer = csv.writer(csv_f,
delimiter=",",
quotechar='"',
quoting=csv.QUOTE_MINIMAL)
for line in csv_lines:
csv_writer.writerow(line)
# Final prints
msg = "%s successfully created!" % (csv_file)
logger.info(msg)
msg = "Number of samples: %s " % (str(len(loaded_csv)))
logger.info(msg)
msg = "Total duration: %s Hours" % (str(round(total_duration / 3600, 2)))
logger.info(msg)
def extractLength(couple):
speaker, locPath = couple
info = torchaudio.info(str(locPath))
return info.num_frames
def calc_waveform_length(path: str, sample_rate: int) -> int:
info, _ = ta.info(path)
return math.ceil(info.length * sample_rate / info.rate / info.channels)
def create_csv(wav_list, csv_file):
"""
Creates the csv file given a list of wav files.
Arguments
---------
wav_list : list of str
The list of wav files.
csv_file : str
The path of the output json file
"""
# Adding some Prints
msg = f"Creating csv lists in {csv_file} ..."
logger.info(msg)
csv_lines = []
# Start processing lines
total_duration = 0.0
# Starting index
idx = 0
for wav_file in tzip(wav_list):
wav_file = wav_file[0]
path_parts = wav_file.split(os.path.sep)
file_name, wav_format = os.path.splitext(path_parts[-1])
# Peeking at the signal (to retrieve duration in seconds)
if os.path.isfile(wav_file):
info = torchaudio.info(wav_file)
else:
msg = "\tError loading: %s" % (str(len(file_name)))
logger.info(msg)
continue
audio_duration = info.num_frames / info.sample_rate
total_duration += audio_duration
# Actual name of the language
language = path_parts[-4]
# Create a row with whole utterences
csv_line = [
idx, # ID
wav_file, # File name
wav_format, # File format
str(info.num_frames / info.sample_rate), # Duration (sec)
language, # Language
]
# Adding this line to the csv_lines list
csv_lines.append(csv_line)
# Increment index
idx += 1
# CSV column titles
csv_header = ["ID", "wav", "wav_format", "duration", "language"]
# Add titles to the list at indexx 0
csv_lines.insert(0, csv_header)
# Writing the csv lines
with open(csv_file, mode="w", encoding="utf-8") as csv_f:
csv_writer = csv.writer(csv_f,
delimiter=",",
quotechar='"',
quoting=csv.QUOTE_MINIMAL)
for line in csv_lines:
csv_writer.writerow(line)
# Final prints
msg = f"{csv_file} sucessfully created!"
logger.info(msg)
msg = f"Number of samples: {len(wav_list)}."
logger.info(msg)
msg = f"Total duration: {round(total_duration / 3600, 2)} hours."
logger.info(msg)
def test_save(self):
# load signal
x, sr = load(self.test_filepath)
# check save
new_filepath = os.path.join(self.test_dirpath, "test.wav")
save(new_filepath, x, sr)
self.assertTrue(os.path.isfile(new_filepath))
os.unlink(new_filepath)
# check automatic normalization
x /= 1 << 31
save(new_filepath, x, sr)
self.assertTrue(os.path.isfile(new_filepath))
os.unlink(new_filepath)
# test save 1d tensor
x = x[:, 0] # get mono signal
x.squeeze_() # remove channel dim
save(new_filepath, x, sr)
self.assertTrue(os.path.isfile(new_filepath))
os.unlink(new_filepath)
# don't allow invalid sizes as inputs
with self.assertRaises(ValueError):
x.unsqueeze_(0) # N x L not L x N
save(new_filepath, x, sr)
with self.assertRaises(ValueError):
x.squeeze_()
x.unsqueeze_(1)
x.unsqueeze_(0) # 1 x L x 1
save(new_filepath, x, sr)
# automatically convert sr from floating point to int
x.squeeze_(0)
save(new_filepath, x, float(sr))
self.assertTrue(os.path.isfile(new_filepath))
os.unlink(new_filepath)
# don't save to folders that don't exist
with self.assertRaises(OSError):
new_filepath = os.path.join(self.test_dirpath, "no-path",
"test.wav")
save(new_filepath, x, sr)
# save created file
sinewave_filepath = os.path.join(self.test_dirpath, "assets",
"sinewave.wav")
sr = 16000
freq = 440
volume = 0.3
y = (torch.cos(2 * math.pi * torch.arange(0, 4 * sr).float() * freq /
sr))
y.unsqueeze_(1)
# y is between -1 and 1, so must scale
y = (y * volume * 2**31).long()
save(sinewave_filepath, y, sr)
self.assertTrue(os.path.isfile(sinewave_filepath))
# test precision
new_filepath = os.path.join(self.test_dirpath, "test.wav")
si, ei = torchaudio.info(sinewave_filepath)
save(new_filepath, y, sr, precision=16)
si16, ei16 = torchaudio.info(new_filepath)
self.assertEqual(si.precision, 32)
self.assertEqual(si16.precision, 16)
os.unlink(new_filepath)
def __getitem__(self, index) -> AudioAndLabels:
track, audio_paths, tsv_path = self.file_list[index]
audio = None
if index < self.max_files_in_memory:
audio = self.audios[index]
# The first time the audio needs to be loaded in memory
if audio is None:
audio = load_audio(audio_paths, normalize=False)
self.audios[index] = audio
labels: Labels = self.labels[index]
# The first the labels needs to be loaded in memory
if labels is None:
labels = self.load_labels(audio_paths, tsv_path)
self.labels[index] = labels
audio_length = torchaudio.info(audio_paths[0]).num_frames
start_frame = None
end_frame = None
if self.sequence_length is not None:
possible_start_interval = audio_length - self.sequence_length
if self.reproducable_load_sequences:
step_begin = (
int(hashlib.sha256("".join(audio_paths).encode("utf-8")).hexdigest(), 16) % possible_start_interval
)
else:
step_begin = self.random.randint(possible_start_interval)
step_begin //= HOP_LENGTH
n_steps = self.sequence_length // HOP_LENGTH
step_end = step_begin + n_steps
begin = step_begin * HOP_LENGTH
end = begin + self.sequence_length
num_frames = end - begin
if audio is None:
audio = load_audio(audio_paths, frame_offset=begin, num_frames=num_frames, normalize=False).to(
self.device
)
else:
audio = audio[begin:end].to(self.device)
label = labels.label[step_begin:step_end, :].to(self.device)
velocity = labels.velocity[step_begin:step_end, :].to(self.device)
start_frame = begin
end_frame = end
else:
if audio is None:
audio = load_audio(audio_paths, normalize=False).to(self.device)
else:
audio = audio.to(self.device)
label = labels.label.to(self.device)
velocity = labels.velocity.to(self.device).float()
start_frame = 0
end_frame = audio_length
onset = (label == 4).float()
frame = (label > 1).float()
offset = ((label == 1) + (label == 2)).float()
velocity = velocity.float().div_(128.0)
return AudioAndLabels(
track=track,
start_time=start_frame/SAMPLE_RATE,
end_time=end_frame/SAMPLE_RATE,
audio=audio,
annotation=MusicAnnotation(onset=onset, offset=offset, frame=frame, velocity=velocity),
)
def audio_pipeline(
mix_wav,
): # this is dummy --> it means one epoch will be same as without dynamic mixing
"""
This audio pipeline defines the compute graph for dynamic mixing
"""
speakers = np.random.choice(spk_list,
hparams["num_spks"],
replace=False,
p=spk_weights)
if hparams["use_wham_noise"]:
noise_file = np.random.choice(noise_files, 1, replace=False)
noise, fs_read = torchaudio.load(noise_file[0])
noise = noise.squeeze()
# select two speakers randomly
sources = []
spk_files = [
np.random.choice(spk_hashtable[spk], 1, False)[0]
for spk in speakers
]
minlen = min(
*[torchaudio.info(x).num_frames for x in spk_files],
hparams["training_signal_len"],
)
meter = pyloudnorm.Meter(hparams["sample_rate"])
MAX_AMP = 0.9
MIN_LOUDNESS = -33
MAX_LOUDNESS = -25
def normalize(signal, is_noise=False):
"""
This function normalizes the audio signals for loudness
"""
with warnings.catch_warnings():
warnings.simplefilter("ignore")
c_loudness = meter.integrated_loudness(signal)
if is_noise:
target_loudness = random.uniform(MIN_LOUDNESS - 5,
MAX_LOUDNESS - 5)
else:
target_loudness = random.uniform(MIN_LOUDNESS,
MAX_LOUDNESS)
signal = pyloudnorm.normalize.loudness(signal, c_loudness,
target_loudness)
# check for clipping
if np.max(np.abs(signal)) >= 1:
signal = signal * MAX_AMP / np.max(np.abs(signal))
return torch.from_numpy(signal)
for i, spk_file in enumerate(spk_files):
# select random offset
length = torchaudio.info(spk_file).num_frames
start = 0
stop = length
if length > minlen: # take a random window
start = np.random.randint(0, length - minlen)
stop = start + minlen
tmp, fs_read = torchaudio.load(
spk_file,
frame_offset=start,
num_frames=stop - start,
)
tmp = tmp[0].numpy()
tmp = normalize(tmp)
sources.append(tmp)
sources = torch.stack(sources)
mixture = torch.sum(sources, 0)
if hparams["use_wham_noise"]:
len_noise = len(noise)
len_mix = len(mixture)
min_len = min(len_noise, len_mix)
noise = normalize(noise.numpy(), is_noise=True)
mixture = mixture[:min_len] + noise[:min_len]
# check for clipping
max_amp_insig = mixture.abs().max().item()
if max_amp_insig > MAX_AMP:
weight = MAX_AMP / max_amp_insig
else:
weight = 1
sources = weight * sources
mixture = weight * mixture
yield mixture
for i in range(hparams["num_spks"]):
yield sources[i]
# If the number of speakers is 2, yield None for the 3rd speaker
if hparams["num_spks"] == 2:
yield None
if hparams["use_wham_noise"]:
noise = noise * weight
yield noise
else:
yield None
def __getitem__(self, index):
"""
:param index:
:return:
"""
# Read start and stop and convert to time in seconds
if self.idmap.start[index] is None:
start = 0
else:
start = int(self.idmap.start[index] * 0.01 * self.sample_rate)
if self.idmap.stop[index] is None:
#speech, speech_fs = get_sample(f"{self.data_path}/{self.idmap.rightids[index]}.{self.file_extension}", resample=self.sample_rate)
nfo = torchaudio.info(
f"{self.data_path}/{self.idmap.rightids[index]}.{self.file_extension}"
)
speech, speech_fs = torchaudio.load(
f"{self.data_path}/{self.idmap.rightids[index]}.{self.file_extension}"
)
if nfo.sample_rate != self.sample_rate:
speech = torchaudio.transforms.Resample(
nfo.sample_rate, self.sample_rate).forward(speech)
duration = int(speech.shape[1] - start)
start = 0
stop = speech.shape[1]
else:
# TODO Check if that code is still relevant with torchaudio.load() in case of sample_rate mismatch
nfo = torchaudio.info(
f"{self.data_path}/{self.idmap.rightids[index]}.{self.file_extension}"
)
assert nfo.sample_rate == self.sample_rate
conversion_rate = nfo.sample_rate // self.sample_rate
start = start
stop = (int(self.idmap.stop[index] * 0.01 * self.sample_rate) -
start)
# add this in case the segment is too short
if stop - start <= self.min_duration * self.sample_rate:
middle = start + (stop - start) // 2
start = max(
0,
int(middle - (self.min_duration * self.sample_rate / 2)))
duration = int(self.min_duration * self.sample_rate)
speech, speech_fs = torchaudio.load(
f"{self.data_path}/{self.idmap.rightids[index]}.{self.file_extension}",
frame_offset=start * conversion_rate,
num_frames=duration * conversion_rate)
speech = torchaudio.transforms.Resample(
nfo.sample_rate, self.sample_rate).forward(speech)
speech += 10e-6 * torch.randn(speech.shape)
if self.sliding_window:
speech = speech.squeeze().unfold(0, self.window_len,
self.window_shift)
middle_points = numpy.arange(
start + self.window_len / 2,
start + duration - self.window_len / 2, self.window_shift)
starts = middle_points - self.window_shift / 2
stops = middle_points + self.window_shift / 2
starts[0] = start
start = starts
stops[-1] = start + duration
else:
stop = start + duration
if len(self.transformation.keys()) > 0:
speech = data_augmentation(speech,
speech_fs,
self.transformation,
self.transform_number,
noise_df=self.noise_df,
rir_df=self.rir_df)
if self.backward:
speech = torch.flip(speech, [0, 1]).squeeze()
else:
speech = speech.squeeze()
return speech, self.idmap.leftids[index], self.idmap.rightids[
index], start, stop
def get_dataset_fast_api_version(file_location):
files = []
siginfo, _ = torchaudio.info(file_location)
length = siginfo.length // siginfo.channels
files.append((file_location, length))
return Audioset(files, with_path=True, sample_rate=sample_rate)
import torchaudio
import matplotlib.pyplot as plt
from pathlib import Path
# p = Path('Supercharger_Blockiergebuehr_Tesla_Fordert_Geld_V.mp3')
#
# print(p.exists())
filename = "../noise_cancellation/Supercharger_Blockiergebühr_Tesla_Fordert_Geld_V_25sec.mp3"
print(torchaudio.info(filename))
waveform, sample_rate = torchaudio.load(filename)
print("Shape of waveform: {}".format(waveform.size()))
print("Sample rate of waveform: {}".format(sample_rate))
plt.figure()
plt.plot(waveform.t().numpy())
plt.show()
specgram = torchaudio.transforms.Spectrogram()(waveform)
print("Shape of spectrogram: {}".format(specgram.size()))
plt.figure()
plt.imshow(specgram.log2()[0, :, :].numpy(), cmap='gray')
plt.show()
specgram = torchaudio.transforms.MelSpectrogram()(waveform)
print("Shape of spectrogram: {}".format(specgram.size()))
def __init__(self,
root_dir,
subset="train",
length=16384,
preload=False,
half=True,
use_soundfile=False):
"""
Args:
root_dir (str): Path to the root directory of the SignalTrain dataset.
subset (str, optional): Pull data either from "train", "val", or "test" subsets. (Default: "train")
length (int, optional): Number of samples in the returned examples. (Default: 40)
preload (bool, optional): Read in all data into RAM during init. (Default: False)
half (bool, optional): Store the float32 audio as float16. (Default: True)
use_soundfile (bool, optional): Use the soundfile library to load instead of torchaudio. (Default: False)
"""
self.root_dir = root_dir
self.subset = subset
self.length = length
self.preload = preload
self.half = half
self.use_soundfile = use_soundfile
# get all the target files files in the directory first
self.target_files = glob.glob(
os.path.join(self.root_dir, self.subset.capitalize(),
"target_*.wav"))
self.input_files = glob.glob(
os.path.join(self.root_dir, self.subset.capitalize(),
"input_*.wav"))
self.examples = []
self.hours = 0 # total number of hours of data in the subset
# ensure that the sets are ordered correctlty
self.target_files.sort()
self.input_files.sort()
# get the parameters
self.params = [(float(f.split("__")[1].replace(".wav", "")),
float(f.split("__")[2].replace(".wav", "")))
for f in self.target_files]
# loop over files to count total length
for idx, (tfile, ifile, params) in enumerate(
zip(self.target_files, self.input_files, self.params)):
ifile_id = int(os.path.basename(ifile).split("_")[1])
tfile_id = int(os.path.basename(tfile).split("_")[1])
if ifile_id != tfile_id:
raise RuntimeError(
f"Found non-matching file ids: {ifile_id} != {tfile_id}! Check dataset."
)
md = torchaudio.info(tfile)
self.hours += (md.num_frames / md.sample_rate) / 3600
num_frames = md.num_frames
if self.preload:
sys.stdout.write(
f"* Pre-loading... {idx+1:3d}/{len(self.target_files):3d} ...\r"
)
sys.stdout.flush()
input, sr = self.load(ifile)
target, sr = self.load(tfile)
num_frames = int(np.min([input.shape[-1], target.shape[-1]]))
if input.shape[-1] != target.shape[-1]:
print(os.path.basename(ifile), input.shape[-1],
os.path.basename(tfile), target.shape[-1])
raise RuntimeError("Found potentially corrupt file!")
if self.half:
input = input.half()
target = target.half()
else:
input = None
target = None
# create one entry for each patch
for n in range((num_frames // self.length) - 1):
offset = int(n * self.length)
end = offset + self.length
self.examples.append({
"idx":
idx,
"target_file":
tfile,
"input_file":
ifile,
"input_audio":
input[:, offset:end] if input is not None else None,
"target_audio":
target[:, offset:end] if input is not None else None,
"params":
params,
"offset":
offset,
"frames":
num_frames
})
# we then want to get the input files
print(
f"Located {len(self.examples)} examples totaling {self.hours:0.1f} hr in the {self.subset} subset."
)
