|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
"""Librispeech automatic speech recognition dataset.""" |
|
|
|
|
|
import os |
|
|
|
import datasets |
|
from datasets.tasks import AutomaticSpeechRecognition |
|
|
|
|
|
_CITATION = """\ |
|
@inproceedings{panayotov2015librispeech, |
|
title={Librispeech: an ASR corpus based on public domain audio books}, |
|
author={Panayotov, Vassil and Chen, Guoguo and Povey, Daniel and Khudanpur, Sanjeev}, |
|
booktitle={Acoustics, Speech and Signal Processing (ICASSP), 2015 IEEE International Conference on}, |
|
pages={5206--5210}, |
|
year={2015}, |
|
organization={IEEE} |
|
} |
|
""" |
|
|
|
_DESCRIPTION = """\ |
|
LibriSpeech is a corpus of approximately 1000 hours of read English speech with sampling rate of 16 kHz, |
|
prepared by Vassil Panayotov with the assistance of Daniel Povey. The data is derived from read |
|
audiobooks from the LibriVox project, and has been carefully segmented and aligned.87 |
|
""" |
|
|
|
_URL = "http://www.openslr.org/12" |
|
_DL_URL = "http://www.openslr.org/resources/12/" |
|
|
|
|
|
_DL_URLS = { |
|
"clean": { |
|
"dev": _DL_URL + "dev-clean.tar.gz", |
|
"train.100": _DL_URL + "train-clean-100.tar.gz", |
|
}, |
|
} |
|
|
|
|
|
class LibrispeechASRConfig(datasets.BuilderConfig): |
|
"""BuilderConfig for LibriSpeechASR.""" |
|
|
|
def __init__(self, **kwargs): |
|
""" |
|
Args: |
|
data_dir: `string`, the path to the folder containing the files in the |
|
downloaded .tar |
|
citation: `string`, citation for the data set |
|
url: `string`, url for information about the data set |
|
**kwargs: keyword arguments forwarded to super. |
|
""" |
|
super(LibrispeechASRConfig, self).__init__(version=datasets.Version("2.1.0", ""), **kwargs) |
|
|
|
|
|
class LibrispeechASR(datasets.GeneratorBasedBuilder): |
|
"""Librispeech dataset.""" |
|
|
|
DEFAULT_WRITER_BATCH_SIZE = 256 |
|
DEFAULT_CONFIG_NAME = "all" |
|
BUILDER_CONFIGS = [ |
|
LibrispeechASRConfig(name="clean", description="'Clean' speech."), |
|
] |
|
|
|
def _info(self): |
|
return datasets.DatasetInfo( |
|
description=_DESCRIPTION, |
|
features=datasets.Features( |
|
{ |
|
"file": datasets.Value("string"), |
|
"audio": datasets.Audio(sampling_rate=16_000), |
|
"text": datasets.Value("string"), |
|
"speaker_id": datasets.Value("int64"), |
|
"chapter_id": datasets.Value("int64"), |
|
"id": datasets.Value("string"), |
|
} |
|
), |
|
supervised_keys=("file", "text"), |
|
homepage=_URL, |
|
citation=_CITATION, |
|
task_templates=[AutomaticSpeechRecognition(audio_column="audio", transcription_column="text")], |
|
) |
|
|
|
def _split_generators(self, dl_manager): |
|
archive_path = dl_manager.download(_DL_URLS[self.config.name]) |
|
|
|
local_extracted_archive = dl_manager.extract(archive_path) if not dl_manager.is_streaming else {} |
|
|
|
if self.config.name == "clean": |
|
train_splits = [ |
|
datasets.SplitGenerator( |
|
name="train.100", |
|
gen_kwargs={ |
|
"local_extracted_archive": local_extracted_archive.get("train.100"), |
|
"files": dl_manager.iter_archive(archive_path["train.100"]), |
|
}, |
|
), |
|
] |
|
dev_splits = [ |
|
datasets.SplitGenerator( |
|
name=datasets.Split.VALIDATION, |
|
gen_kwargs={ |
|
"local_extracted_archive": local_extracted_archive.get("dev"), |
|
"files": dl_manager.iter_archive(archive_path["dev"]), |
|
}, |
|
) |
|
] |
|
|
|
return train_splits + dev_splits |
|
|
|
def _generate_examples(self, files, local_extracted_archive): |
|
"""Generate examples from a LibriSpeech archive_path.""" |
|
key = 0 |
|
audio_data = {} |
|
transcripts = [] |
|
for path, f in files: |
|
if path.endswith(".flac"): |
|
id_ = path.split("/")[-1][: -len(".flac")] |
|
audio_data[id_] = f.read() |
|
elif path.endswith(".trans.txt"): |
|
for line in f: |
|
if line: |
|
line = line.decode("utf-8").strip() |
|
id_, transcript = line.split(" ", 1) |
|
audio_file = f"{id_}.flac" |
|
speaker_id, chapter_id = [int(el) for el in id_.split("-")[:2]] |
|
audio_file = ( |
|
os.path.join(local_extracted_archive, audio_file) |
|
if local_extracted_archive |
|
else audio_file |
|
) |
|
transcripts.append( |
|
{ |
|
"id": id_, |
|
"speaker_id": speaker_id, |
|
"chapter_id": chapter_id, |
|
"file": audio_file, |
|
"text": transcript, |
|
} |
|
) |
|
if audio_data and len(audio_data) == len(transcripts): |
|
for transcript in transcripts: |
|
audio = {"path": transcript["file"], "bytes": audio_data[transcript["id"]]} |
|
yield key, {"audio": audio, **transcript} |
|
key += 1 |
|
audio_data = {} |
|
transcripts = [] |
|
|