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| import io |
| import os |
| from dataclasses import dataclass |
| from pathlib import Path |
| from typing import Dict, List, Optional, Tuple |
|
|
| import librosa |
| import soundfile as sf |
| import torch.utils.data |
|
|
| from nemo.collections.asr.data.audio_to_text import expand_sharded_filepaths |
| from nemo.collections.asr.parts.preprocessing.segment import available_formats as valid_sf_formats |
| from nemo.collections.asr.parts.utils.manifest_utils import read_manifest |
| from nemo.collections.tts.parts.preprocessing.feature_processors import FeatureProcessor |
| from nemo.collections.tts.parts.utils.tts_dataset_utils import ( |
| filter_dataset_by_duration, |
| get_weighted_sampler, |
| load_audio, |
| resample_batch, |
| sample_audio, |
| stack_tensors, |
| ) |
| from nemo.core.classes import Dataset, IterableDataset |
| from nemo.utils import logging |
| from nemo.utils import webdataset as wds |
| from nemo.utils.distributed import webdataset_split_by_workers |
|
|
| VALID_FILE_FORMATS = ';'.join(['wav', 'mp3', 'flac', 'opus'] + [fmt.lower() for fmt in valid_sf_formats.keys()]) |
|
|
|
|
| @dataclass |
| class DatasetMeta: |
| manifest_path: Path |
| audio_dir: Path |
| sample_weight: float = 1.0 |
| audio_tar_filepaths: Optional[List[str]] = None |
|
|
|
|
| @dataclass |
| class DatasetSample: |
| dataset_name: str |
| manifest_entry: dict |
| audio_dir: Path |
|
|
|
|
| def audio_collate_fn(batch: List[dict], resample_rates: Optional[Tuple[int]] = None): |
| dataset_name_list = [] |
| audio_filepath_list = [] |
| audio_list = [] |
| audio_len_list = [] |
|
|
| for example in batch: |
| dataset_name_list.append(example["dataset_name"]) |
| audio_filepath_list.append(example["audio_filepath"]) |
| audio_list.append(example["audio"]) |
| audio_len_list.append(example["audio_len"]) |
|
|
| batch_audio_len = torch.IntTensor(audio_len_list) |
| audio_max_len = int(batch_audio_len.max().item()) |
|
|
| batch_audio = stack_tensors(audio_list, max_lens=[audio_max_len]) |
|
|
| if resample_rates: |
| batch_audio, batch_audio_len = resample_batch( |
| audio=batch_audio, |
| audio_len=batch_audio_len, |
| input_sample_rate=resample_rates[0], |
| output_sample_rate=resample_rates[1], |
| ) |
|
|
| batch_dict = { |
| "dataset_names": dataset_name_list, |
| "audio_filepaths": audio_filepath_list, |
| "audio": batch_audio, |
| "audio_lens": batch_audio_len, |
| } |
|
|
| return batch_dict |
|
|
|
|
| def preprocess_manifest( |
| dataset_name: str, |
| dataset: DatasetMeta, |
| min_duration: float, |
| max_duration: float, |
| ): |
| entries = read_manifest(dataset.manifest_path) |
| filtered_entries, total_hours, filtered_hours = filter_dataset_by_duration( |
| entries=entries, min_duration=min_duration, max_duration=max_duration |
| ) |
|
|
| logging.info(dataset_name) |
| logging.info(f"Original # of files: {len(entries)}") |
| logging.info(f"Filtered # of files: {len(filtered_entries)}") |
| logging.info(f"Original duration: {total_hours:.2f} hours") |
| logging.info(f"Filtered duration: {filtered_hours:.2f} hours") |
|
|
| samples = [] |
| sample_weights = [] |
| for entry in filtered_entries: |
| sample = DatasetSample(dataset_name=dataset_name, manifest_entry=entry, audio_dir=Path(dataset.audio_dir)) |
| samples.append(sample) |
| sample_weights.append(dataset.sample_weight) |
|
|
| return samples, sample_weights |
|
|
|
|
| class VocoderDataset(Dataset): |
| """ |
| Class for processing and loading Vocoder training examples. |
| |
| Args: |
| dataset_meta: Dict of dataset names (string) to dataset metadata. |
| sample_rate: Sample rate to load audio as. If the audio is stored at a different sample rate, then it will |
| be resampled using librosa. |
| resample_rate: Optional sample rate to resample to, using torch-based resampling. |
| n_samples: Optional int, if provided then n_samples samples will be randomly sampled from the full |
| audio file. |
| weighted_sampling_steps_per_epoch: Optional int, If provided, then data will be sampled (with replacement) based on |
| the sample weights provided in the dataset metadata. If None, then sample weights will be ignored. |
| feature_processors: Optional, list of feature processors to run on training examples. |
| min_duration: Optional float, if provided audio files in the training manifest shorter than 'min_duration' |
| will be ignored. |
| max_duration: Optional float, if provided audio files in the training manifest longer than 'max_duration' |
| will be ignored. |
| trunc_duration: Optional int, if provided audio will be truncated to at most 'trunc_duration' seconds. |
| volume_norm: Whether to apply volume normalization to loaded audio. |
| """ |
|
|
| def __init__( |
| self, |
| dataset_meta: Dict, |
| sample_rate: int, |
| resample_rate: Optional[int] = None, |
| n_samples: Optional[int] = None, |
| weighted_sampling_steps_per_epoch: Optional[int] = None, |
| feature_processors: Optional[Dict[str, FeatureProcessor]] = None, |
| min_duration: Optional[float] = None, |
| max_duration: Optional[float] = None, |
| trunc_duration: Optional[float] = None, |
| volume_norm: bool = False, |
| ): |
| super().__init__() |
|
|
| self.sample_rate = sample_rate |
| if resample_rate and self.sample_rate != resample_rate: |
| self.resample_rates = [sample_rate, resample_rate] |
| else: |
| self.resample_rates = None |
|
|
| self.n_samples = n_samples |
| self.trunc_duration = trunc_duration |
| self.volume_norm = volume_norm |
| self.weighted_sampling_steps_per_epoch = weighted_sampling_steps_per_epoch |
| self.load_precomputed_mel = False |
|
|
| if feature_processors: |
| logging.info(f"Found feature processors {feature_processors.keys()}") |
| self.feature_processors = list(feature_processors.values()) |
| else: |
| self.feature_processors = [] |
|
|
| self.data_samples = [] |
| self.sample_weights = [] |
| for dataset_name, dataset_info in dataset_meta.items(): |
| dataset = DatasetMeta(**dataset_info) |
| samples, weights = preprocess_manifest( |
| dataset_name=dataset_name, |
| dataset=dataset, |
| min_duration=min_duration, |
| max_duration=max_duration, |
| ) |
| self.data_samples += samples |
| self.sample_weights += weights |
|
|
| def get_sampler(self, batch_size: int, world_size: int) -> Optional[torch.utils.data.Sampler]: |
| if not self.weighted_sampling_steps_per_epoch: |
| return None |
|
|
| sampler = get_weighted_sampler( |
| sample_weights=self.sample_weights, |
| batch_size=batch_size, |
| world_size=world_size, |
| num_steps=self.weighted_sampling_steps_per_epoch, |
| ) |
| return sampler |
|
|
| def __len__(self): |
| return len(self.data_samples) |
|
|
| def __getitem__(self, index): |
| data = self.data_samples[index] |
|
|
| if self.n_samples: |
| audio_array, _, audio_filepath_rel = sample_audio( |
| manifest_entry=data.manifest_entry, |
| audio_dir=data.audio_dir, |
| sample_rate=self.sample_rate, |
| n_samples=self.n_samples, |
| volume_norm=self.volume_norm, |
| ) |
| else: |
| audio_array, _, audio_filepath_rel = load_audio( |
| manifest_entry=data.manifest_entry, |
| audio_dir=data.audio_dir, |
| sample_rate=self.sample_rate, |
| max_duration=self.trunc_duration, |
| volume_norm=self.volume_norm, |
| ) |
| audio = torch.tensor(audio_array, dtype=torch.float32) |
| audio_len = audio.shape[0] |
|
|
| example = { |
| "dataset_name": data.dataset_name, |
| "audio_filepath": audio_filepath_rel, |
| "audio": audio, |
| "audio_len": audio_len, |
| } |
|
|
| for processor in self.feature_processors: |
| processor.process(example) |
|
|
| return example |
|
|
| def collate_fn(self, batch): |
| return audio_collate_fn(batch, resample_rates=self.resample_rates) |
|
|
|
|
| class TarredVocoderDataset(IterableDataset): |
| """ |
| A similar Dataset to the VocoderDataset, but loads tarred audio files. |
| |
| Accepts a single comma-separated JSON manifest file (in the same style as for the VocoderDataset), |
| as well as the path(s) to the tarball(s) containing the wav files. Each line of the manifest should |
| contain the information for one audio file, and duration of audio. |
| |
| Valid formats for the audio_tar_filepaths argument include: |
| (1) a single string that can be brace-expanded, e.g. 'path/to/audio.tar' or 'path/to/audio_{1..100}.tar.gz', or |
| (2) a list of file paths that will not be brace-expanded, e.g. ['audio_1.tar', 'audio_2.tar', ...]. |
| |
| See the WebDataset documentation for more information about accepted data and input formats. |
| |
| If using multiple processes the number of shards should be divisible by the number of workers to ensure an |
| even split among workers. If it is not divisible, logging will give a warning but training will proceed. |
| In addition, if using mutiprocessing, each shard MUST HAVE THE SAME NUMBER OF ENTRIES after filtering |
| is applied. We currently do not check for this, but your program may hang if the shards are uneven! |
| |
| Additionally, please note that the len() of this DataLayer is assumed to be the length of the manifest |
| after filtering. An incorrect manifest length may lead to some DataLoader issues down the line. |
| |
| Args: |
| dataset_meta: Dict of dataset names (string) to dataset metadata. |
| audio_tar_filepaths: Either a list of audio tarball filepaths, or a |
| string (can be brace-expandable). |
| sample_rate: Sample rate to load audio as. If the audio is stored at a different sample rate, then it will |
| be resampled. |
| n_samples: Optional int, if provided then n_samples samples will be randomly sampled from the full |
| audio file. |
| shuffle_n (int): How many samples to look ahead and load to be shuffled. |
| See WebDataset documentation for more details. |
| Defaults to 0. |
| min_duration: Optional float, if provided audio files in the training manifest shorter than 'min_duration' |
| will be ignored. |
| max_duration: Optional float, if provided audio files in the training manifest longer than 'max_duration' |
| will be ignored. |
| trunc_duration: Optional int, if provided audio will be truncated to at most 'trunc_duration' seconds. |
| feature_processors: Optional, list of feature processors to run on training examples. |
| shard_strategy (str): Tarred dataset shard distribution strategy chosen as a str value during ddp. |
| - `scatter`: The default shard strategy applied by WebDataset, where each node gets |
| a unique set of shards, which are permanently pre-allocated and never changed at runtime. |
| - `replicate`: Optional shard strategy, where each node gets all of the set of shards |
| available in the tarred dataset, which are permanently pre-allocated and never changed at runtime. |
| The benefit of replication is that it allows each node to sample data points from the entire |
| dataset independently of other nodes, and reduces dependence on value of `shuffle_n`. |
| |
| .. warning:: |
| Replicated strategy allows every node to sample the entire set of available tarfiles, |
| and therefore more than one node may sample the same tarfile, and even sample the same |
| data points! As such, there is no assured guarantee that all samples in the dataset will be |
| sampled at least once during 1 epoch. Scattered strategy, on the other hand, on specific |
| occasions (when the number of shards is not divisible with ``world_size``), will not sample |
| the entire dataset. For these reasons it is not advisable to use tarred datasets as validation |
| or test datasets. |
| 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 0. |
| """ |
|
|
| def __init__( |
| self, |
| dataset_meta: Dict, |
| sample_rate: int, |
| n_samples: Optional[int] = None, |
| shuffle_n: int = 0, |
| min_duration: float = 0.1, |
| max_duration: Optional[float] = None, |
| trunc_duration: Optional[float] = None, |
| feature_processors: Optional[Dict[str, FeatureProcessor]] = None, |
| shard_strategy: str = "scatter", |
| global_rank: int = 0, |
| world_size: int = 2, |
| **kwargs, |
| ): |
| super().__init__() |
|
|
| if len(kwargs) > 0: |
| logging.warning( |
| f"Arguments {kwargs.keys()} does not support for TarredVocoderDataset, they will be ignored." |
| ) |
|
|
| self.sample_rate = sample_rate |
| self.n_samples = n_samples |
|
|
| if trunc_duration: |
| self.trunc_samples = int(trunc_duration * self.sample_rate) |
| else: |
| self.trunc_samples = None |
|
|
| if feature_processors: |
| logging.info(f"Found feature processors {feature_processors.keys()}") |
| self.feature_processors = list(feature_processors.values()) |
| else: |
| self.feature_processors = [] |
|
|
| self.data_samples = [] |
| self.audio_tar_filepaths = [] |
| for dataset_name, dataset_info in dataset_meta.items(): |
| audio_tar_filepaths = dataset_info.audio_tar_filepaths |
| self.audio_tar_filepaths += [audio_tar_filepaths] |
| dataset = DatasetMeta(**dataset_info) |
| samples, _ = preprocess_manifest( |
| dataset_name=dataset_name, |
| dataset=dataset, |
| min_duration=min_duration, |
| max_duration=max_duration, |
| ) |
| self.data_samples += samples |
|
|
| self.file_id_to_sample_map = {} |
| for sample in self.data_samples: |
| file_id = os.path.splitext(os.path.basename(sample.manifest_entry["audio_filepath"]))[0] |
| if file_id not in self.file_id_to_sample_map: |
| self.file_id_to_sample_map[file_id] = sample |
| else: |
| raise ValueError( |
| f"Duplicate file_id {file_id} found in manifest {sample.manifest_entry['audio_filepath']}" |
| ) |
|
|
| logging.info(f"world size: {world_size}") |
| audio_tar_filepaths = expand_sharded_filepaths( |
| sharded_filepaths=audio_tar_filepaths, |
| global_rank=global_rank, |
| world_size=world_size, |
| shard_strategy=shard_strategy, |
| ) |
|
|
| self._dataset = wds.DataPipeline( |
| wds.SimpleShardList(urls=audio_tar_filepaths), |
| webdataset_split_by_workers, |
| wds.shuffle(shuffle_n), |
| wds.tarfile_to_samples(), |
| wds.rename(audio=VALID_FILE_FORMATS, key='__key__'), |
| wds.to_tuple('audio', 'key'), |
| self._filter, |
| wds.map(self._build_sample), |
| ) |
|
|
| def _filter(self, iterator): |
| class FilteredIterator: |
| def __init__(self, file_id_to_sample_map): |
| self.iterator = iterator |
| self.file_id_to_sample_map = file_id_to_sample_map |
|
|
| def __iter__(self): |
| return self |
|
|
| def __next__(self): |
| while True: |
| audio_bytes, audio_filename = next(self.iterator) |
| file_id = os.path.splitext(os.path.basename(audio_filename))[0] |
| if file_id in self.file_id_to_sample_map: |
| return audio_bytes, audio_filename |
|
|
| return FilteredIterator(self.file_id_to_sample_map) |
|
|
| def _build_sample(self, tup): |
| audio_bytes, audio_filename = tup |
| file_id = os.path.splitext(os.path.basename(audio_filename))[0] |
| data = self.file_id_to_sample_map[file_id] |
|
|
| audio_array, sr = sf.read(file=io.BytesIO(audio_bytes), dtype='float32') |
| if sr != self.sample_rate: |
| logging.warning( |
| f"Sample rate of {sr} does not match target sample rate of {self.sample_rate}. Resampling audio." |
| ) |
| audio_array = librosa.core.resample(audio_array, orig_sr=sr, target_sr=self.sample_rate) |
|
|
| audio_array = torch.from_numpy(audio_array) |
| if self.n_samples: |
| len_audio = audio_array.shape[0] |
| if len_audio > self.n_samples: |
| start = torch.randint(0, len_audio - self.n_samples, (1,)) |
| audio_array = audio_array[start : start + self.n_samples] |
| else: |
| audio_array = audio_array[: self.n_samples] |
|
|
| if self.trunc_samples: |
| audio_array = audio_array[: self.trunc_samples] |
|
|
| audio_len = torch.tensor(audio_array.shape[0]) |
|
|
| example = { |
| "dataset_name": data.dataset_name, |
| "audio_filepath": audio_filename, |
| "audio": audio_array, |
| "audio_len": audio_len, |
| } |
|
|
| for processor in self.feature_processors: |
| processor.process(example) |
|
|
| return example |
|
|
| def get_sampler(self, batch_size: int, world_size: int): |
| """ |
| Currently sampler is not supported for tarred dataset. |
| """ |
| return None |
|
|
| def collate_fn(self, batch): |
| return audio_collate_fn(batch) |
|
|
| def __iter__(self): |
| return self._dataset.__iter__() |
|
|
| def __len__(self): |
| return len(self.file_id_to_sample_map) |
|
|