class Lsun_Loader(object):
def __init__(self, device_idx=1):
self.batch_table = {
4: 128,
8: 128,
16: 128,
32: 64,
64: 32,
128: 16,
256: 8
}
self.device_idx = device_idx
#self.update(resolution_level=2)
def update(self, resolution_level, batch_size):
'''
:param resolution_level:
resolution_level:
2-2,3-8,4-16,5-32,6-64,7-128,8-256
:return:
'''
assert resolution_level >= 2 and resolution_level <= 8, "res error"
self.batchsize = int(self.batch_table[pow(2, resolution_level)])
self.pipeline = TFRecordPipeline(
batch_size=batch_size,
size=pow(2, resolution_level),
num_threads=4,
device_id=self.device_idx,
path=
"/home/guyuchao/ssd/dataset/lsun-master/lsun_torch_tfrecord2/-r%02d.tfrecords"
% resolution_level,
index_path=
"/home/guyuchao/ssd/dataset/lsun-master/lsun_torch_tfrecord2/-r%02d.idx"
% resolution_level)
self.pipeline.build()
self.dali_iter = DALIGenericIterator(
[self.pipeline], ["image/encoded"],
self.pipeline.epoch_size("Reader"),
auto_reset=True)
def get_batch(self):
#self.dali_iter.reset()
return self.dali_iter.next()[0]["image/encoded"]
class AudioToCharDALIDataset(Iterator):
"""
NVIDIA DALI pipeline that loads tensors via one or more manifest files where each line containing a sample descriptor in JSON,
including audio files, transcripts, and durations (in seconds).
Here's an example:
{"audio_filepath": "/path/to/audio.wav", "text_filepath": "/path/to/audio.txt", "duration": 23.147}
...
{"audio_filepath": "/path/to/audio.wav", "text": "the transcription", "offset": 301.75, "duration": 0.82, "utt":
"utterance_id", "ctm_utt": "en_4156", "side": "A"}
Args:
manifest_filepath: Path to manifest file with the format described above. Can be comma-separated paths.
labels: String containing all the possible characters to map to.
sample_rate (int): Sample rate to resample loaded audio to.
batch_size (int): Number of samples in a batch.
num_threads (int): Number of CPU processing threads to be created by the DALI pipeline.
max_duration (float): Determines the maximum allowed duration, in seconds, of the loaded audio files.
min_duration (float): Determines the minimum allowed duration, in seconds, of the loaded audio files.
blank_index (int): blank character index, default = -1
unk_index (int): unk_character index, default = -1
normalize (bool): whether to normalize transcript text (default): True
bos_id (int): Id of beginning of sequence symbol to append if not None
eos_id (int): Id of end of sequence symbol to append if not None
trim (bool): If True, it will extract the nonsilent region of the loaded audio signal.
shuffle (bool): If set to True, the dataset will shuffled after loading.
drop_last (bool): If set to True, the last batch will be dropped if incomplete. This will be the case when the shard size is not divisible by the batch size.
If set to False and the size of dataset is not divisible by the batch size, then the last batch will be smaller.
device (str): Determines the device type to be used for preprocessing. Allowed values are: 'cpu', 'gpu'.
device_id (int): Index of the GPU to be used (local_rank). Only applicable when device == 'gpu'. Defaults to 0.
global_rank (int): Worker rank, used for partitioning shards. Defaults to 0.
world_size (int): Total number of processes, used for partitioning shards. Defaults to 1.
preprocessor_cfg (DictConfig): Preprocessor configuration. Supports AudioToMelSpectrogramPreprocessor and AudioToMFCCPreprocessor.
"""
def __init__(
self,
manifest_filepath: str,
device: str,
batch_size: int,
labels: Union[str, List[str]],
sample_rate: int = 16000,
num_threads: int = 4,
max_duration: float = 0.0,
min_duration: float = 0.0,
blank_index: int = -1,
unk_index: int = -1,
normalize: bool = True,
bos_id: Optional[int] = None,
eos_id: Optional[int] = None,
trim: bool = False,
shuffle: bool = True,
drop_last: bool = False,
parser: Union[str, Callable] = 'en',
device_id: int = 0,
global_rank: int = 0,
world_size: int = 1,
preprocessor_cfg: DictConfig = None,
):
if not HAVE_DALI:
raise ModuleNotFoundError(
f"{self} requires NVIDIA DALI to be installed. "
f"See: https://docs.nvidia.com/deeplearning/dali/user-guide/docs/installation.html#id1"
)
if device not in ('cpu', 'gpu'):
raise ValueError(
f"{self} received an unexpected device argument {device}. Supported values are: 'cpu', 'gpu'"
)
self.batch_size = batch_size # Used by NeMo
self.device = device
self.device_id = device_id
if world_size > 1:
self.shard_id = global_rank
self.num_shards = world_size
else:
self.shard_id = None
self.num_shards = None
self.labels = labels
if self.labels is None or len(self.labels) == 0:
raise ValueError(f"{self} expects non empty labels list")
self.parser = parsers.make_parser(
labels=labels,
name=parser,
unk_id=unk_index,
blank_id=blank_index,
do_normalize=normalize,
)
self.eos_id = eos_id
self.bos_id = bos_id
self.sample_rate = sample_rate
self.pipe = Pipeline(
batch_size=batch_size,
num_threads=num_threads,
device_id=self.device_id,
exec_async=True,
exec_pipelined=True,
)
has_preprocessor = preprocessor_cfg is not None
if has_preprocessor:
if preprocessor_cfg.cls == "nemo.collections.asr.modules.AudioToMelSpectrogramPreprocessor":
feature_type = "mel_spectrogram"
elif preprocessor_cfg.cls == "nemo.collections.asr.modules.AudioToMFCCPreprocessor":
feature_type = "mfcc"
else:
raise ValueError(
f"{self} received an unexpected preprocessor configuration: {preprocessor_cfg.cls}."
f" Supported preprocessors are: AudioToMelSpectrogramPreprocessor, AudioToMFCCPreprocessor"
)
# Default values taken from AudioToMelSpectrogramPreprocessor
params = preprocessor_cfg.params
self.dither = params['dither'] if 'dither' in params else 0.0
self.preemph = params['preemph'] if 'preemph' in params else 0.97
self.window_size_sec = params[
'window_size'] if 'window_size' in params else 0.02
self.window_stride_sec = params[
'window_stride'] if 'window_stride' in params else 0.01
self.sample_rate = params[
'sample_rate'] if 'sample_rate' in params else sample_rate
self.window_size = int(self.window_size_sec * self.sample_rate)
self.window_stride = int(self.window_size_sec * self.sample_rate)
normalize = params[
'normalize'] if 'normalize' in params else 'per_feature'
if normalize == 'per_feature': # Each freq channel independently
self.normalization_axes = (1, )
elif normalize == 'all_features':
self.normalization_axes = (0, 1)
else:
raise ValueError(
f"{self} received {normalize} for the normalize parameter."
f" It must be either 'per_feature' or 'all_features'.")
self.window = None
window_name = params['window'] if 'window' in params else None
torch_windows = {
'hamming': torch.hamming_window,
'blackman': torch.blackman_window,
'bartlett': torch.bartlett_window,
}
if window_name is None or window_name == 'hann':
self.window = None # Hann is DALI's default
elif window_name == 'ones':
self.window = torch.ones(self.window_size)
else:
try:
window_fn = torch_windows.get(window_name, None)
self.window = window_fn(self.window_size, periodic=False)
except:
raise ValueError(
f"{self} received {window_name} for the window parameter."
f" It must be one of: ('hann', 'ones', 'hamming', 'blackman', 'bartlett', None)."
f" None is equivalent to 'hann'.")
self.n_fft = params[
'n_fft'] if 'n_fft' in params else None # None means default
self.n_mels = params['n_mels'] if 'n_mels' in params else 64
self.n_mfcc = params['n_mfcc'] if 'n_mfcc' in params else 64
features = params['features'] if 'features' in params else 0
if features > 0:
if feature_type == 'mel_spectrogram':
self.n_mels = features
elif feature_type == 'mfcc':
self.n_mfcc = features
# TODO Implement frame splicing
if 'frame_splicing' in params:
assert params[
'frame_splicing'] == 1, "Frame splicing is not implemented"
self.freq_low = params['lowfreq'] if 'lowfreq' in params else 0.0
self.freq_high = params[
'highfreq'] if 'highfreq' in params else self.sample_rate / 2.0
self.log_features = params['log'] if 'log' in params else True
# We want to avoid taking the log of zero
# There are two options: either adding or clamping to a small value
self.log_zero_guard_type = params[
'log_zero_guard_type'] if 'log_zero_guard_type' in params else 'add'
if self.log_zero_guard_type not in ["add", "clamp"]:
raise ValueError(
f"{self} received {self.log_zero_guard_type} for the "
f"log_zero_guard_type parameter. It must be either 'add' or "
f"'clamp'.")
self.log_zero_guard_value = params[
'log_zero_guard_value'] if 'log_zero_guard_value' in params else 1e-05
if isinstance(self.log_zero_guard_value, str):
if self.log_zero_guard_value == "tiny":
self.log_zero_guard_value = torch.finfo(torch.float32).tiny
elif self.log_zero_guard_value == "eps":
self.log_zero_guard_value = torch.finfo(torch.float32).eps
else:
raise ValueError(
f"{self} received {self.log_zero_guard_value} for the log_zero_guard_type parameter."
f"It must be either a number, 'tiny', or 'eps'")
self.mag_power = params['mag_power'] if 'mag_power' in params else 2
if self.mag_power != 1.0 and self.mag_power != 2.0:
raise ValueError(
f"{self} received {self.mag_power} for the mag_power parameter."
f" It must be either 1.0 or 2.0.")
self.pad_to = params['pad_to'] if 'pad_to' in params else 16
self.pad_value = params[
'pad_value'] if 'pad_value' in params else 0.0
with self.pipe:
audio, transcript = dali.fn.nemo_asr_reader(
name="Reader",
manifest_filepaths=manifest_filepath.split(','),
dtype=dali.types.FLOAT,
downmix=True,
sample_rate=float(self.sample_rate),
min_duration=min_duration,
max_duration=max_duration,
read_sample_rate=False,
read_text=True,
random_shuffle=shuffle,
shard_id=self.shard_id,
num_shards=self.num_shards,
pad_last_batch=True,
)
transcript_len = dali.fn.shapes(
dali.fn.reshape(transcript, shape=[-1]))
transcript = dali.fn.pad(transcript)
# Extract nonsilent region, if necessary
if trim:
# Need to extract non-silent region before moving to the GPU
roi_start, roi_len = dali.fn.nonsilent_region(audio,
cutoff_db=-60)
audio = audio.gpu() if self.device == 'gpu' else audio
audio = dali.fn.slice(audio,
roi_start,
roi_len,
normalized_anchor=False,
normalized_shape=False,
axes=[0])
else:
audio = audio.gpu() if self.device == 'gpu' else audio
if not has_preprocessor:
# No preprocessing, the output is the audio signal
audio = dali.fn.pad(audio)
audio_len = dali.fn.shapes(dali.fn.reshape(audio, shape=[-1]))
self.pipe.set_outputs(audio, audio_len, transcript,
transcript_len)
else:
# Additive gaussian noise (dither)
if self.dither > 0.0:
gaussian_noise = dali.fn.normal_distribution(
device=self.device)
audio = audio + self.dither * gaussian_noise
# Preemphasis filter
if self.preemph > 0.0:
audio = dali.fn.preemphasis_filter(
audio, preemph_coeff=self.preemph)
# Power spectrogram
spec = dali.fn.spectrogram(audio,
nfft=self.n_fft,
window_length=self.window_size,
window_step=self.window_stride)
if feature_type == 'mel_spectrogram' or feature_type == 'mfcc':
# Spectrogram to Mel Spectrogram
spec = dali.fn.mel_filter_bank(
spec,
sample_rate=self.sample_rate,
nfilter=self.n_mels,
normalize=True,
freq_low=self.freq_low,
freq_high=self.freq_high,
)
# Mel Spectrogram to MFCC
if feature_type == 'mfcc':
spec = dali.fn.mfcc(spec, n_mfcc=self.n_mfcc)
# Logarithm
if self.log_zero_guard_type == 'add':
spec = spec + self.log_zero_guard_value
spec = dali.fn.to_decibels(spec,
multiplier=math.log(10),
reference=1.0,
cutoff_db=math.log(
self.log_zero_guard_value))
# Normalization
spec = dali.fn.normalize(spec, axes=self.normalization_axes)
# Extracting the length of the spectrogram
shape_start = dali.types.Constant(np.array([1],
dtype=np.float32),
device='cpu')
shape_len = dali.types.Constant(np.array([1],
dtype=np.float32),
device='cpu')
spec_len = dali.fn.slice(
dali.fn.shapes(spec),
shape_start,
shape_len,
normalized_anchor=False,
normalized_shape=False,
axes=(0, ),
)
# Pads feature dimension to be a multiple of `pad_to` and the temporal dimension to be as big as the largest sample (shape -1)
spec = dali.fn.pad(spec,
fill_value=self.pad_value,
axes=(0, 1),
align=(self.pad_to, 1),
shape=(1, -1))
self.pipe.set_outputs(spec, spec_len, transcript, transcript_len)
# Building DALI pipeline
self.pipe.build()
if has_preprocessor:
output_names = [
'processed_signal', 'processed_signal_len', 'transcript_raw',
'transcript_raw_len'
]
else:
output_names = [
'audio', 'audio_len', 'transcript_raw', 'transcript_raw_len'
]
last_batch_policy = LastBatchPolicy.DROP if drop_last else LastBatchPolicy.PARTIAL
self._iter = DALIPytorchIterator(
[self.pipe],
output_map=output_names,
reader_name="Reader",
last_batch_policy=last_batch_policy,
dynamic_shape=True,
auto_reset=True,
)
# TODO come up with a better solution
class DummyDataset:
def __init__(self, parent):
self.parent = parent
def __len__(self):
return self.parent.size
self.dataset = DummyDataset(self) # Used by NeMo
def reset(self):
self._iter.reset()
def __iter__(self):
return self
def next(self):
return self.__next__()
@property
def size(self):
return self._iter.size
def __len__(self):
return len(self._iter)
def __next__(self):
outputs = self._iter.next()
assert len(outputs) == 1
out = outputs[0]
text_raw_len = out['transcript_raw_len'].numpy()
text_raw = out['transcript_raw'].numpy()
text_tokens = []
text_tokens_len = []
max_len = 0
batch_size = text_raw.shape[0]
for i, text in enumerate(text_raw):
n = text_raw_len[i][0]
tbytes = str(text[:n].tobytes(), encoding='utf8')
ttokens = self.parser(tbytes)
if self.bos_id is not None:
ttokens = [self.bos_id] + ttokens
if self.eos_id is not None:
ttokens = ttokens + [self.eos_id]
ttokens_len = len(ttokens)
text_tokens_len.append(ttokens_len)
text_tokens.append(ttokens)
if ttokens_len > max_len:
max_len = ttokens_len
transcript_out = torch.zeros(batch_size, max_len, dtype=torch.long)
for i, n in enumerate(text_tokens_len):
transcript_out[i, :n] = torch.tensor(text_tokens[i],
dtype=torch.long)
transcript_len_out = torch.tensor(text_tokens_len, dtype=torch.long)
out['transcript'] = transcript_out
out['transcript_len'] = transcript_len_out
return DALIOutputs(out)
class _AudioTextDALIDataset(Iterator):
"""
NVIDIA DALI pipeline that loads tensors via one or more manifest files where each line containing a sample descriptor in JSON,
including audio files, transcripts, and durations (in seconds).
Here's an example:
{"audio_filepath": "/path/to/audio.wav", "text_filepath": "/path/to/audio.txt", "duration": 23.147}
...
{"audio_filepath": "/path/to/audio.wav", "text": "the transcription", "offset": 301.75, "duration": 0.82, "utt":
"utterance_id", "ctm_utt": "en_4156", "side": "A"}
Args:
manifest_filepath: Path to manifest file with the format described above. Can be comma-separated paths.
device (str): Determines the device type to be used for preprocessing. Allowed values are: 'cpu', 'gpu'.
batch_size (int): Number of samples in a batch.
parser (str, callable): A str for an inbuilt parser, or a callable with signature f(str) -> List[int].
sample_rate (int): Sample rate to resample loaded audio to.
num_threads (int): Number of CPU processing threads to be created by the DALI pipeline.
max_duration (float): Determines the maximum allowed duration, in seconds, of the loaded audio files.
min_duration (float): Determines the minimum allowed duration, in seconds, of the loaded audio files.
bos_id (int): Id of beginning of sequence symbol to append if not None
eos_id (int): Id of end of sequence symbol to append if not None
pad_id (int): Id used to pad the input. Defaults to 0 if not provided.
trim (bool): If True, it will extract the nonsilent region of the loaded audio signal.
shuffle (bool): If set to True, the dataset will shuffled after loading.
drop_last (bool): If set to True, the last batch will be dropped if incomplete. This will be the case when the shard size is not divisible by the batch size.
If set to False and the size of dataset is not divisible by the batch size, then the last batch will be smaller.
device_id (int): Index of the GPU to be used (local_rank). Only applicable when device == 'gpu'. Defaults to 0.
global_rank (int): Worker rank, used for partitioning shards. Defaults to 0.
world_size (int): Total number of processes, used for partitioning shards. Defaults to 1.
preprocessor_cfg (DictConfig): Preprocessor configuration. Supports AudioToMelSpectrogramPreprocessor and AudioToMFCCPreprocessor.
return_sample_id (bool): whether to return the sample_id as a part of each sample (not supported yet).
"""
def __init__(
self,
manifest_filepath: str,
device: str,
batch_size: int,
parser: Union[str, Callable],
audio_tar_filepaths: Optional[Union[str, List[str]]] = None,
audio_tar_index_filepaths: Optional[Union[str, List[str]]] = None,
sample_rate: int = 16000,
num_threads: int = 4,
max_duration: float = 0.0,
min_duration: float = 0.0,
bos_id: Optional[int] = None,
eos_id: Optional[int] = None,
pad_id: int = 0,
trim: bool = False,
shuffle: bool = False,
drop_last: bool = False,
shard_strategy: str = "scatter",
device_id: int = 0,
global_rank: int = 0,
world_size: int = 1,
preprocessor_cfg: DictConfig = None,
return_sample_id: bool = False,
):
self.drop_last = drop_last # used by lr_scheduler
if return_sample_id:
raise ValueError(
"Currently DALI data layers don't support returning the sample_id and return_sample_id can not be enabled."
)
self.return_sample_id = return_sample_id
if not HAVE_DALI:
raise ModuleNotFoundError(
f"{self} requires NVIDIA DALI to be installed. "
f"See: https://docs.nvidia.com/deeplearning/dali/user-guide/docs/installation.html#id1"
)
if device not in ('cpu', 'gpu'):
raise ValueError(
f"{self} received an unexpected device argument {device}. Supported values are: 'cpu', 'gpu'"
)
device_id = device_id if device == 'gpu' else None
self.batch_size = batch_size # Used by NeMo
self.device = device
self.device_id = device_id
if world_size > 1:
self.shard_id = global_rank
self.num_shards = world_size
else:
self.shard_id = None
self.num_shards = None
self.eos_id = eos_id
self.bos_id = bos_id
self.sample_rate = sample_rate
self.pipe = Pipeline(
batch_size=batch_size,
num_threads=num_threads,
device_id=self.device_id,
exec_async=True,
exec_pipelined=True,
)
has_preprocessor = preprocessor_cfg is not None
if has_preprocessor:
if preprocessor_cfg._target_ == "nemo.collections.asr.modules.AudioToMelSpectrogramPreprocessor":
feature_type = "mel_spectrogram"
elif preprocessor_cfg._target_ == "nemo.collections.asr.modules.AudioToMFCCPreprocessor":
feature_type = "mfcc"
else:
raise ValueError(
f"{self} received an unexpected preprocessor configuration: {preprocessor_cfg._target_}."
f" Supported preprocessors are: AudioToMelSpectrogramPreprocessor, AudioToMFCCPreprocessor"
)
# Default values taken from AudioToMelSpectrogramPreprocessor
params = preprocessor_cfg
self.dither = params['dither'] if 'dither' in params else 0.0
self.preemph = params['preemph'] if 'preemph' in params else 0.97
self.window_size_sec = params[
'window_size'] if 'window_size' in params else 0.02
self.window_stride_sec = params[
'window_stride'] if 'window_stride' in params else 0.01
self.sample_rate = params[
'sample_rate'] if 'sample_rate' in params else sample_rate
self.window_size = int(self.window_size_sec * self.sample_rate)
self.window_stride = int(self.window_stride_sec * self.sample_rate)
normalize = params[
'normalize'] if 'normalize' in params else 'per_feature'
if normalize == 'per_feature': # Each freq channel independently
self.normalization_axes = (1, )
elif normalize == 'all_features':
self.normalization_axes = (0, 1)
else:
raise ValueError(
f"{self} received {normalize} for the normalize parameter."
f" It must be either 'per_feature' or 'all_features'.")
self.window = None
window_name = params['window'] if 'window' in params else 'hann'
torch_windows = {
'hann': torch.hann_window,
'hamming': torch.hamming_window,
'blackman': torch.blackman_window,
'bartlett': torch.bartlett_window,
'none': None,
}
if window_name == 'ones':
window_tensor = torch.ones(self.window_size)
else:
try:
window_fn = torch_windows.get(window_name, None)
except:
raise ValueError(
f"{self} received '{window_name}' for the window parameter."
f" It must be one of: ('hann', 'ones', 'hamming', 'blackman', 'bartlett', None)."
f" None is equivalent to 'hann'.")
window_tensor = window_fn(
self.window_size, periodic=False) if window_fn else None
self.window = window_tensor.numpy().tolist(
) if window_tensor is not None else None
self.n_fft = params['n_fft'] if 'n_fft' in params else 2**math.ceil(
math.log2(self.window_size))
self.n_mels = params['n_mels'] if 'n_mels' in params else 64
self.n_mfcc = params['n_mfcc'] if 'n_mfcc' in params else 64
features = params['features'] if 'features' in params else 0
if features > 0:
if feature_type == 'mel_spectrogram':
self.n_mels = features
elif feature_type == 'mfcc':
self.n_mfcc = features
# TODO Implement frame splicing
if 'frame_splicing' in params:
assert params[
'frame_splicing'] == 1, "Frame splicing is not implemented"
self.freq_low = params['lowfreq'] if 'lowfreq' in params else 0.0
self.freq_high = params[
'highfreq'] if 'highfreq' in params else self.sample_rate / 2.0
self.log_features = params['log'] if 'log' in params else True
# We want to avoid taking the log of zero
# There are two options: either adding or clamping to a small value
self.log_zero_guard_type = params[
'log_zero_guard_type'] if 'log_zero_guard_type' in params else 'add'
if self.log_zero_guard_type not in ["add", "clamp"]:
raise ValueError(
f"{self} received {self.log_zero_guard_type} for the "
f"log_zero_guard_type parameter. It must be either 'add' or "
f"'clamp'.")
self.log_zero_guard_value = (params['log_zero_guard_value']
if 'log_zero_guard_value' in params
else 2**-24)
if isinstance(self.log_zero_guard_value, str):
if self.log_zero_guard_value == "tiny":
self.log_zero_guard_value = torch.finfo(torch.float32).tiny
elif self.log_zero_guard_value == "eps":
self.log_zero_guard_value = torch.finfo(torch.float32).eps
else:
raise ValueError(
f"{self} received {self.log_zero_guard_value} for the log_zero_guard_type parameter."
f"It must be either a number, 'tiny', or 'eps'")
self.mag_power = params['mag_power'] if 'mag_power' in params else 2
if self.mag_power != 1.0 and self.mag_power != 2.0:
raise ValueError(
f"{self} received {self.mag_power} for the mag_power parameter."
f" It must be either 1.0 or 2.0.")
self.pad_to = max(params['pad_to'],
1) if 'pad_to' in params else 16
self.pad_value = params[
'pad_value'] if 'pad_value' in params else 0.0
with self.pipe:
if audio_tar_filepaths is None and audio_tar_index_filepaths is None:
audio, indices = dali.fn.readers.nemo_asr(
name="Reader",
manifest_filepaths=manifest_filepath.split(','),
dtype=dali.types.FLOAT,
downmix=True,
sample_rate=float(self.sample_rate),
min_duration=min_duration,
max_duration=max_duration,
read_sample_rate=False,
read_text=False,
read_idxs=True,
random_shuffle=shuffle,
shard_id=self.shard_id,
num_shards=self.num_shards,
pad_last_batch=True,
)
self.is_tarred_dataset = False
elif audio_tar_filepaths is not None and audio_tar_index_filepaths is not None:
audio_tar_filepaths = expand_audio_filepaths(
audio_tar_filepaths,
shard_strategy=shard_strategy,
world_size=world_size,
global_rank=global_rank)
audio_tar_index_filepaths = expand_audio_filepaths(
audio_tar_index_filepaths,
shard_strategy=shard_strategy,
world_size=world_size,
global_rank=global_rank,
)
if len(audio_tar_filepaths) != len(
audio_tar_index_filepaths) and len(
audio_tar_index_filepaths) != 0:
raise ValueError(
f"Number of filepaths provided for `audio_tar_filepaths` must match "
f"`audio_tar_index_filepaths`. Got {len(audio_tar_filepaths)} audio_tar_filepaths and "
f"{len(audio_tar_index_filepaths)} audio_tar_index_filepaths."
)
tar_file = dali.fn.readers.webdataset(
paths=audio_tar_filepaths,
index_paths=audio_tar_index_filepaths,
name="Reader",
ext=["wav"],
missing_component_behavior="error",
random_shuffle=shuffle,
shard_id=self.shard_id,
num_shards=self.num_shards,
pad_last_batch=True,
)
audio, _ = dali.fn.decoders.audio(
tar_file,
dtype=dali.types.FLOAT,
downmix=True,
sample_rate=float(self.sample_rate),
)
indices = dali.fn.get_property(tar_file, key="source_info")
indices = dali.fn.pad(indices)
self.is_tarred_dataset = True
else:
raise RuntimeError(
"When using DALI datasets, either `audio_tar_filepaths` "
"and `audio_tar_index_filepaths` should either both be None (sequential dataset)"
"or provided (tarred dataset).")
# Extract nonsilent region, if necessary
if trim:
# Need to extract non-silent region before moving to the GPU
roi_start, roi_len = dali.fn.nonsilent_region(audio,
cutoff_db=-60)
audio = audio.gpu() if self.device == 'gpu' else audio
audio = dali.fn.slice(audio,
roi_start,
roi_len,
normalized_anchor=False,
normalized_shape=False,
axes=[0])
else:
audio = audio.gpu() if self.device == 'gpu' else audio
if not has_preprocessor:
# No preprocessing, the output is the audio signal
audio_len = dali.fn.shapes(dali.fn.reshape(audio, shape=[-1]))
audio = dali.fn.pad(audio)
self.pipe.set_outputs(audio, audio_len, indices)
else:
# Additive gaussian noise (dither)
if self.dither > 0.0:
gaussian_noise = dali.fn.random.normal(audio)
audio = audio + self.dither * gaussian_noise
# Preemphasis filter
if self.preemph > 0.0:
audio = dali.fn.preemphasis_filter(
audio, preemph_coeff=self.preemph, border='zero')
# Power spectrogram
spec = dali.fn.spectrogram(
audio,
nfft=self.n_fft,
window_length=self.window_size,
window_step=self.window_stride,
window_fn=self.window,
)
if feature_type == 'mel_spectrogram' or feature_type == 'mfcc':
# Spectrogram to Mel Spectrogram
spec = dali.fn.mel_filter_bank(
spec,
sample_rate=self.sample_rate,
nfilter=self.n_mels,
normalize=True,
freq_low=self.freq_low,
freq_high=self.freq_high,
)
# Mel Spectrogram to MFCC
if feature_type == 'mfcc':
spec = dali.fn.mfcc(spec, n_mfcc=self.n_mfcc)
# Logarithm
if self.log_zero_guard_type == 'add':
spec = spec + self.log_zero_guard_value
spec = dali.fn.to_decibels(spec,
multiplier=math.log(10),
reference=1.0,
cutoff_db=math.log(
self.log_zero_guard_value))
# Normalization
spec = dali.fn.normalize(spec,
axes=self.normalization_axes,
epsilon=1e-5**2,
ddof=1)
# Extracting the length of the spectrogram
spec_len = dali.fn.slice(dali.fn.shapes(spec),
1,
1,
axes=(0, ))
# Pads feature dimension to be a multiple of `pad_to` and the temporal dimension to be as big as the largest sample (shape -1)
spec = dali.fn.pad(spec,
fill_value=self.pad_value,
axes=(0, 1),
align=(self.pad_to, 1),
shape=(1, -1))
self.pipe.set_outputs(spec, spec_len, indices)
x = time.time()
# Building DALI pipeline
self.pipe.build()
y = time.time()
logging.info(f"Time for pipe.build() : {(y - x)} seconds")
if has_preprocessor:
output_names = [
'processed_signal', 'processed_signal_len', 'manifest_indices'
]
else:
output_names = ['audio', 'audio_len', 'manifest_indices']
x = time.time()
last_batch_policy = LastBatchPolicy.DROP if drop_last else LastBatchPolicy.PARTIAL
self._iter = DALIPytorchIterator(
[self.pipe],
output_map=output_names,
reader_name="Reader",
last_batch_policy=last_batch_policy,
dynamic_shape=True,
auto_reset=True,
)
y = time.time()
logging.info(
f"Time for DALIPytorchIterator to initialize : {(y - x)} seconds")
# TODO come up with a better solution
class DummyDataset:
def __init__(self, parent):
self.parent = parent
def __len__(self):
return self.parent.size
self.dataset = DummyDataset(self) # Used by NeMo
x = time.time()
self.manifest_processor = ASRManifestProcessor(
manifest_filepath=manifest_filepath,
parser=parser,
max_duration=max_duration,
min_duration=min_duration,
max_utts=0,
bos_id=bos_id,
eos_id=eos_id,
pad_id=pad_id,
index_by_file_id=self.is_tarred_dataset,
)
y = time.time()
logging.info(
f"Time to build nemo manifest processor - {(y - x)} seconds")
def reset(self):
self._iter.reset()
def __iter__(self):
return self
def next(self):
return self.__next__()
@property
def size(self):
return self._iter.size
def __len__(self):
return len(self._iter)
def __next__(self):
outputs = self._iter.next()
assert len(outputs) == 1
dali_out = outputs[0]
manifest_indices = dali_out['manifest_indices'].numpy()
out = {}
out_names = [
'processed_signal', 'processed_signal_len', 'audio', 'audio_len'
]
for out_name in out_names:
if out_name in dali_out:
out[out_name] = dali_out[out_name].detach().clone()
text_tokens = []
text_tokens_len = []
max_len = 0
batch_size = manifest_indices.shape[0]
for i, manifest_index in enumerate(manifest_indices):
if not self.is_tarred_dataset:
# Loose-file dataset. Index is integer based.
manifest_index = manifest_index[0]
text, text_length = self.manifest_processor.process_text_by_id(
manifest_index)
else:
# Tarred-file dataset. Index is filename based.
resolved_manifest_indices = manifest_index.tobytes().decode(
).split(":")
resolved_manifest_index = resolved_manifest_indices[
2] # we require just the filename segment
resolved_manifest_index = os.path.splitext(
resolved_manifest_index)[0] # we dont need file extension
text, text_length = self.manifest_processor.process_text_by_file_id(
resolved_manifest_index)
text_tokens_len.append(text_length)
text_tokens.append(text)
if text_length > max_len:
max_len = text_length
transcript_out = torch.full([batch_size, max_len],
fill_value=self.manifest_processor.pad_id,
dtype=torch.long)
for i, n in enumerate(text_tokens_len):
transcript_out[i, :n] = torch.tensor(text_tokens[i],
dtype=torch.long)
transcript_len_out = torch.tensor(text_tokens_len, dtype=torch.long)
out['transcript'] = transcript_out
out['transcript_len'] = transcript_len_out
return DALIOutputs(out)
