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488 lines
22 KiB
Python
488 lines
22 KiB
Python
# Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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from typing import Callable, Dict, List, Optional, Tuple, Union
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import torch
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from nemo.collections.asr.data.feature_to_label import _audio_feature_collate_fn
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from nemo.collections.asr.parts.preprocessing.feature_loader import ExternalFeatureLoader
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from nemo.collections.asr.parts.preprocessing.features import normalize_batch
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from nemo.collections.asr.parts.preprocessing.segment import ChannelSelectorType
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from nemo.collections.asr.parts.utils.vad_utils import load_speech_segments_from_rttm
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from nemo.collections.common import tokenizers
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from nemo.collections.common.parts.preprocessing import collections, parsers
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from nemo.core.classes import Dataset
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from nemo.core.neural_types import AcousticEncodedRepresentation, LabelsType, LengthsType, NeuralType
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class ASRFeatureManifestProcessor:
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def __init__(
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self,
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manifest_filepath: str,
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parser: Union[str, Callable],
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max_duration: Optional[float] = None,
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min_duration: Optional[float] = None,
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max_utts: int = 0,
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bos_id: Optional[int] = None,
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eos_id: Optional[int] = None,
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pad_id: int = 0,
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index_by_file_id: bool = False,
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):
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self.parser = parser
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self.collection = collections.ASRFeatureText(
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manifests_files=manifest_filepath,
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parser=parser,
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min_duration=min_duration,
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max_duration=max_duration,
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max_number=max_utts,
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index_by_file_id=index_by_file_id,
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)
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self.eos_id = eos_id
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self.bos_id = bos_id
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self.pad_id = pad_id
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def process_text_by_id(self, index: int) -> Tuple[List[int], int]:
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sample = self.collection[index]
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return self.process_text_by_sample(sample)
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def process_text_by_file_id(self, file_id: str) -> Tuple[List[int], int]:
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manifest_idx = self.collection.mapping[file_id][0]
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sample = self.collection[manifest_idx]
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return self.process_text_by_sample(sample)
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def process_text_by_sample(self, sample: collections.ASRAudioText.OUTPUT_TYPE) -> Tuple[List[int], int]:
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t, tl = sample.text_tokens, len(sample.text_tokens)
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if self.bos_id is not None:
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t = [self.bos_id] + t
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tl += 1
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if self.eos_id is not None:
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t = t + [self.eos_id]
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tl += 1
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return t, tl
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class _FeatureTextDataset(Dataset):
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"""
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Dataset that loads tensors via a json file containing paths to audio feature files, transcripts,
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durations (in seconds) and optional RTTM files. Each new line is a different sample. Example below:
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{"feature_filepath": "/path/to/audio_feature.pt", "text_filepath": "/path/to/audio.txt",
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"rttm_filepath": "/path/to/audio_rttm.rttm", "duration": 23.147}
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...
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{"feature_filepath": "/path/to/audio_feature.pt", "text": "the transcription", "offset": 301.75, "duration": 0.82, "utt":
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"utterance_id", "ctm_utt": "en_4156", "side": "A"}
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Args:
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manifest_filepath (str): Path to manifest json as described above. Can be comma-separated paths.
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parser: Str for a language specific preprocessor or a callable.
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normalize (bool): whether and where to normalize feature, must be one of [None, "post_norm", "pre_norm"]
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normalize_type (Union[str, dict]): how to normalize feature, see `nemo.collections.asr.parts.preprocessing.features.normalize_batch`
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use_rttm (bool): whether to use RTTM files if there is any, default to False
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rttm_mode (str): how to use RTTM files, must be one of ['mask', 'drop'], default to 'mask'
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feat_min_len (int): minimum length of feature when rttm_mode=deop, default to 4.
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feat_mask_val (Optional[float]): value used to mask features with RTTM files, default to None to use zero mel-spectralgram
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frame_unit_time_secs (float): time in seconds for each frame
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sample_rate (int): Sample rate to resample loaded audio to
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int_values (bool): If true, load samples as 32-bit integers. Defauts to False.
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augmentor (nemo.collections.asr.parts.perturb.AudioAugmentor): An AudioAugmentor object used to augment loaded audio
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max_duration (float): If audio exceeds this length, do not include in dataset
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min_duration (float): If audio is less than this length, do not include in dataset
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max_utts (int): Limit number of utterances
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trim (bool): whether or not to trim silence. Defaults to False
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bos_id (int): Id of beginning of sequence symbol to append if not None
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eos_id (int): Id of end of sequence symbol to append if not None
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pad_id (int): Id of pad symbol. Defaults to 0
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return_sample_id (bool): whether to return the sample_id as a part of each sample
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channel_selector (int | Iterable[int] | str): select a single channel or a subset of channels from multi-channel audio. If set to `'average'`, it performs averaging across channels. Disabled if set to `None`. Defaults to `None`. Uses zero-based indexing.
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"""
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ZERO_LEVEL_SPEC_DB_VAL = -16.635 # Log-Melspectrogram value for zero signal
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NORM_MODES = ["pre_norm", "post_norm"]
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RTTM_MODES = ["mask", "drop"]
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@property
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def output_types(self) -> Optional[Dict[str, NeuralType]]:
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"""Returns definitions of module output ports."""
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return {
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'features': NeuralType(('B', 'D', 'T'), AcousticEncodedRepresentation()),
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'feature_length': NeuralType(tuple('B'), LengthsType()),
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'transcripts': NeuralType(('B', 'T'), LabelsType()),
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'transcript_length': NeuralType(tuple('B'), LengthsType()),
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'sample_id': NeuralType(tuple('B'), LengthsType(), optional=True),
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}
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def __init__(
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self,
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manifest_filepath: str,
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parser: Union[str, Callable],
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normalize: Optional[str] = "post_norm",
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normalize_type: Union[str, dict] = "per_feature",
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use_rttm: bool = False,
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rttm_mode: str = "mask",
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feat_min_len: int = 4,
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feat_mask_val: Optional[float] = None,
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frame_unit_time_secs: float = 0.01,
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sample_rate: Optional[int] = 16000,
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augmentor: 'nemo.collections.asr.parts.perturb.FeatureAugmentor' = None,
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max_duration: Optional[int] = None,
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min_duration: Optional[int] = None,
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max_utts: int = 0,
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trim: bool = False,
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bos_id: Optional[int] = None,
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eos_id: Optional[int] = None,
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pad_id: int = 0,
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return_sample_id: bool = False,
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channel_selector: Optional[ChannelSelectorType] = None,
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):
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if type(manifest_filepath) == str:
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manifest_filepath = manifest_filepath.split(",")
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self.sample_rate = sample_rate
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self.normalize = normalize
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self.normalize_type = normalize_type
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self.use_rttm = use_rttm
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self.rttm_mode = rttm_mode
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if self.use_rttm and self.rttm_mode not in self.RTTM_MODES:
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raise ValueError(f"`rttm_mode` must be one of {self.RTTM_MODES}, got `{rttm_mode}` instead")
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self.feat_min_len = feat_min_len
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if feat_mask_val is not None:
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self.feat_mask_val = feat_mask_val
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elif normalize == "pre_norm":
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self.feat_mask_val = 0.0 # similar to SpectralAugmentation
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else:
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self.feat_mask_val = self.ZERO_LEVEL_SPEC_DB_VAL
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if normalize is not None and normalize not in self.NORM_MODES:
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raise ValueError(f"`normalize` must be one of {self.NORM_MODES}, got `{normalize}` instead")
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self.frame_unit_time_secs = frame_unit_time_secs
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self.manifest_processor = ASRFeatureManifestProcessor(
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manifest_filepath=manifest_filepath,
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parser=parser,
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max_duration=max_duration,
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min_duration=min_duration,
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max_utts=max_utts,
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bos_id=bos_id,
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eos_id=eos_id,
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pad_id=pad_id,
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)
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self.featurizer = ExternalFeatureLoader(augmentor=augmentor)
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self.trim = trim
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self.return_sample_id = return_sample_id
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self.channel_selector = channel_selector
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def get_manifest_sample(self, sample_id):
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return self.manifest_processor.collection[sample_id]
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def __getitem__(self, index):
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sample = self.manifest_processor.collection[index]
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offset = sample.offset
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if offset is None:
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offset = 0
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features = self.featurizer.process(sample.feature_file)
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f, fl = features, torch.tensor(features.shape[1]).long()
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t, tl = self.manifest_processor.process_text_by_sample(sample=sample)
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# Feature normalization
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if self.normalize is None:
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if self.use_rttm and sample.rttm_file:
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f = self.process_features_with_rttm(f, offset, sample.rttm_file, self.feat_mask_val)
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elif self.normalize == "post_norm":
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# (Optional) Masking based on RTTM file
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if self.use_rttm and sample.rttm_file:
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f = self.process_features_with_rttm(f, offset, sample.rttm_file, self.feat_mask_val)
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f = self.normalize_feature(f)
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else: # pre-norm
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f = self.normalize_feature(f)
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# (Optional) Masking based on RTTM file
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if self.use_rttm and sample.rttm_file:
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f = self.process_features_with_rttm(f, offset, sample.rttm_file, self.feat_mask_val)
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if self.return_sample_id:
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output = f, fl, torch.tensor(t).long(), torch.tensor(tl).long(), index
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else:
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output = f, fl, torch.tensor(t).long(), torch.tensor(tl).long()
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return output
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def process_features_with_rttm(self, features, offset, rttm_file, mask_val):
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segments = load_speech_segments_from_rttm(rttm_file)
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new_features = features.clone()
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sid, fid = 0, 0
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for i in range(features.size(1)):
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t = offset + i * self.frame_unit_time_secs
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while sid < len(segments) - 1 and segments[sid][1] < t:
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sid += 1
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if segments[sid][1] == 0 or t < segments[sid][0] or t > segments[sid][1]:
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# not in speech segment
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if self.rttm_mode == "drop":
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# drop the frame
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continue
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else:
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# mask the frame with specified value
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new_features[:, i] = mask_val
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fid += 1
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else:
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# in speech segment
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new_features[:, fid] = features[:, i]
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fid += 1
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if fid < self.feat_min_len and self.rttm_mode == "drop":
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new_features[:, : self.feat_min_len] = mask_val
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return new_features[:, : self.feat_min_len]
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return new_features[:, :fid]
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def __len__(self):
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return len(self.manifest_processor.collection)
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def _collate_fn(self, batch):
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return _audio_feature_collate_fn(
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batch, feat_pad_val=self.feat_mask_val, label_pad_id=self.manifest_processor.pad_id
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)
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def normalize_feature(self, feat):
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"""
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Args:
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feat: feature tensor of shape [M, T]
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"""
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feat = feat.unsqueeze(0) # add batch dim
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feat, _, _ = normalize_batch(feat, torch.tensor([feat.size(-1)]), self.normalize_type)
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return feat.squeeze(0) # delete batch dim
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class FeatureToCharDataset(_FeatureTextDataset):
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"""
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Dataset that loads tensors via a json file containing paths to audio feature
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files, transcripts, durations (in seconds) and optional RTTM files. Each new line is a
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different sample. Example below:
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{"feature_filepath": "/path/to/audio_feature.pt", "text_filepath":
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"/path/to/audio.txt", "duration": 23.147, "rttm_filepath": "/path/to/audio_rttm.rttm",}
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...
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{"feature_filepath": "/path/to/audio_feature.pt", "text": "the
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transcription", "offset": 301.75, "duration": 0.82, "utt":
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"utterance_id", "ctm_utt": "en_4156", "side": "A"}
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Args:
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manifest_filepath (str): Path to manifest json as described above. Can
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be comma-separated paths.
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labels (str): String containing all the possible characters to map to
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normalize (str): how to normalize feature, must be one of [None, "post_norm", "pre_norm"]
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normalize_type (Union[str, dict]): how to normalize feature, see `nemo.collections.asr.parts.preprocessing.features.normalize_batch`
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use_rttm (bool): whether to use RTTM files if there is any, default to False
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rttm_mode (str): how to use RTTM files, must be one of ['mask', 'drop'], default to 'mask'
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feat_min_len (int): minimum length of feature, default to 4
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feat_mask_val (Optional[float]): value used to mask features with RTTM files, default to None to use zero mel-spectralgram
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frame_unit_time_secs: time in seconds for each frame
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sample_rate (int): Sample rate to resample loaded audio to
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int_values (bool): If true, load samples as 32-bit integers. Defauts to False.
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augmentor (nemo.collections.asr.parts.perturb.AudioAugmentor): An AudioAugmentor
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object used to augment loaded audio
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max_duration: If audio exceeds this length, do not include in dataset
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min_duration: If audio is less than this length, do not include
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in dataset
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max_utts: Limit number of utterances
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blank_index: blank character index, default = -1
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unk_index: unk_character index, default = -1
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bos_id: Id of beginning of sequence symbol to append if not None
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eos_id: Id of end of sequence symbol to append if not None
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return_sample_id (bool): whether to return the sample_id as a part of each sample
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channel_selector (int | Iterable[int] | str): select a single channel or a subset of channels from multi-channel audio. If set to `'average'`, it performs averaging across channels. Disabled if set to `None`. Defaults to `None`. Uses zero-based indexing.
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"""
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def __init__(
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self,
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manifest_filepath: str,
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labels: Union[str, List[str]],
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normalize: Optional[str] = "post_norm",
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normalize_type: Union[str, dict] = "per_feature",
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use_rttm: bool = False,
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rttm_mode: str = "mask",
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feat_min_len: int = 4,
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feat_mask_val: Optional[float] = None,
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frame_unit_time_secs: float = 0.01,
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sample_rate: Optional[int] = 16000,
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augmentor: 'nemo.collections.asr.parts.perturb.FeatureAugmentor' = None,
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max_duration: Optional[int] = None,
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min_duration: Optional[int] = None,
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max_utts: int = 0,
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blank_index: int = -1,
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unk_index: int = -1,
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trim: bool = False,
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bos_id: Optional[int] = None,
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eos_id: Optional[int] = None,
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pad_id: int = 0,
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parser: Union[str, Callable] = 'en',
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return_sample_id: bool = False,
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channel_selector: Optional[ChannelSelectorType] = None,
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):
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self.labels = labels
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parser = parsers.make_parser(
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labels=labels, name=parser, unk_id=unk_index, blank_id=blank_index, do_normalize=normalize
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)
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super().__init__(
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manifest_filepath=manifest_filepath,
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parser=parser,
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normalize=normalize,
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normalize_type=normalize_type,
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use_rttm=use_rttm,
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rttm_mode=rttm_mode,
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feat_min_len=feat_min_len,
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feat_mask_val=feat_mask_val,
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frame_unit_time_secs=frame_unit_time_secs,
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sample_rate=sample_rate,
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augmentor=augmentor,
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max_duration=max_duration,
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min_duration=min_duration,
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max_utts=max_utts,
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trim=trim,
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bos_id=bos_id,
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eos_id=eos_id,
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pad_id=pad_id,
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return_sample_id=return_sample_id,
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channel_selector=channel_selector,
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)
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class FeatureToBPEDataset(_FeatureTextDataset):
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"""
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Dataset that loads tensors via a json file containing paths to audio feature
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files, transcripts, durations (in seconds) and optional RTTM files. Each new line is a different sample.
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Example below:
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{"audio_filepath": "/path/to/audio.wav", "text_filepath":
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"/path/to/audio.txt", "duration": 23.147, "rttm_filepath": "/path/to/audio_rttm.rttm",}
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...
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{"audio_filepath": "/path/to/audio.wav", "text": "the
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transcription", "offset": 301.75, "duration": 0.82, "utt":
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"utterance_id", "ctm_utt": "en_4156", "side": "A"}
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In practice, the dataset and manifest used for character encoding and byte pair encoding
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are exactly the same. The only difference lies in how the dataset tokenizes the text in
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the manifest.
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Args:
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manifest_filepath (str): Path to manifest json as described above. Can
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be comma-separated paths.
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tokenizer: A subclass of the Tokenizer wrapper found in the common collection,
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nemo.collections.common.tokenizers.TokenizerSpec. ASR Models support a subset of
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all available tokenizers.
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normalize (str): how to normalize feature, must be one of [None, "post_norm", "pre_norm"]
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normalize_type (Union[str, dict]): how to normalize feature, see `nemo.collections.asr.parts.preprocessing.features.normalize_batch`
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use_rttm (bool): whether to use RTTM files if there is any, default to False
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rttm_mode (str): how to use RTTM files, must be one of ['mask', 'drop'], default to 'mask'
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feat_min_len (int): minimum length of feature, default to 4
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feat_mask_val (Optional[float]): value used to mask features with RTTM files, default to None to use zero mel-spectralgram
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frame_unit_time_secs: time in seconds for each frame
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sample_rate (int): Sample rate to resample loaded audio to
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int_values (bool): If true, load samples as 32-bit integers. Defauts to False.
|
|
augmentor (nemo.collections.asr.parts.perturb.AudioAugmentor): An AudioAugmentor
|
|
object used to augment loaded audio
|
|
max_duration: If audio exceeds this length, do not include in dataset
|
|
min_duration: If audio is less than this length, do not include
|
|
in dataset
|
|
max_utts: Limit number of utterances
|
|
trim: Whether to trim silence segments
|
|
use_start_end_token: Boolean which dictates whether to add [BOS] and [EOS]
|
|
tokens to beginning and ending of speech respectively.
|
|
return_sample_id (bool): whether to return the sample_id as a part of each sample
|
|
channel_selector (int | Iterable[int] | str): select a single channel or a subset of channels from multi-channel audio. If set to `'average'`, it performs averaging across channels. Disabled if set to `None`. Defaults to `None`. Uses zero-based indexing.
|
|
"""
|
|
|
|
def __init__(
|
|
self,
|
|
manifest_filepath: str,
|
|
tokenizer: 'nemo.collections.common.tokenizers.TokenizerSpec',
|
|
normalize: Optional[str] = "post_norm",
|
|
normalize_type: Union[str, dict] = "per_feature",
|
|
use_rttm: bool = False,
|
|
rttm_mode: str = "mask",
|
|
feat_min_len: int = 4,
|
|
feat_mask_val: Optional[float] = None,
|
|
frame_unit_time_secs: float = 0.01,
|
|
sample_rate: Optional[int] = 16000,
|
|
augmentor: 'nemo.collections.asr.parts.perturb.FeatureAugmentor' = None,
|
|
max_duration: Optional[int] = None,
|
|
min_duration: Optional[int] = None,
|
|
max_utts: int = 0,
|
|
use_start_end_token: bool = True,
|
|
trim: bool = False,
|
|
return_sample_id: bool = False,
|
|
channel_selector: Optional[ChannelSelectorType] = None,
|
|
):
|
|
if use_start_end_token and hasattr(tokenizer, "bos_id") and tokenizer.bos_id > 0:
|
|
bos_id = tokenizer.bos_id
|
|
else:
|
|
bos_id = None
|
|
|
|
if use_start_end_token and hasattr(tokenizer, "eos_id") and tokenizer.eos_id > 0:
|
|
eos_id = tokenizer.eos_id
|
|
else:
|
|
eos_id = None
|
|
|
|
if hasattr(tokenizer, "pad_id") and tokenizer.pad_id > 0:
|
|
pad_id = tokenizer.pad_id
|
|
else:
|
|
pad_id = 0
|
|
|
|
class TokenizerWrapper:
|
|
def __init__(self, tokenizer):
|
|
if isinstance(tokenizer, tokenizers.aggregate_tokenizer.AggregateTokenizer):
|
|
self.is_aggregate = True
|
|
else:
|
|
self.is_aggregate = False
|
|
self._tokenizer = tokenizer
|
|
|
|
def __call__(self, *args):
|
|
if isinstance(args[0], List) and self.is_aggregate:
|
|
t = []
|
|
for span in args[0]:
|
|
t.extend(self._tokenizer.text_to_ids(span['str'], span['lang']))
|
|
return t
|
|
|
|
t = self._tokenizer.text_to_ids(*args)
|
|
return t
|
|
|
|
super().__init__(
|
|
manifest_filepath=manifest_filepath,
|
|
parser=TokenizerWrapper(tokenizer),
|
|
normalize=normalize,
|
|
normalize_type=normalize_type,
|
|
use_rttm=use_rttm,
|
|
rttm_mode=rttm_mode,
|
|
feat_min_len=feat_min_len,
|
|
feat_mask_val=feat_mask_val,
|
|
frame_unit_time_secs=frame_unit_time_secs,
|
|
sample_rate=sample_rate,
|
|
augmentor=augmentor,
|
|
max_duration=max_duration,
|
|
min_duration=min_duration,
|
|
max_utts=max_utts,
|
|
trim=trim,
|
|
bos_id=bos_id,
|
|
eos_id=eos_id,
|
|
pad_id=pad_id,
|
|
return_sample_id=return_sample_id,
|
|
channel_selector=channel_selector,
|
|
)
|