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Create librispeech_asr.py

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  1. librispeech_asr.py +161 -0
librispeech_asr.py ADDED
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+ # coding=utf-8
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+ # Copyright 2021 The TensorFlow Datasets Authors and the HuggingFace Datasets Authors.
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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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+
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+ # Lint as: python3
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+ """Librispeech automatic speech recognition dataset."""
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+
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+
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+ import os
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+
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+ import datasets
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+ from datasets.tasks import AutomaticSpeechRecognition
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+
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+
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+ _CITATION = """\
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+ @inproceedings{panayotov2015librispeech,
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+ title={Librispeech: an ASR corpus based on public domain audio books},
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+ author={Panayotov, Vassil and Chen, Guoguo and Povey, Daniel and Khudanpur, Sanjeev},
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+ booktitle={Acoustics, Speech and Signal Processing (ICASSP), 2015 IEEE International Conference on},
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+ pages={5206--5210},
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+ year={2015},
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+ organization={IEEE}
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+ }
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+ """
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+
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+ _DESCRIPTION = """\
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+ LibriSpeech is a corpus of approximately 1000 hours of read English speech with sampling rate of 16 kHz,
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+ prepared by Vassil Panayotov with the assistance of Daniel Povey. The data is derived from read
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+ audiobooks from the LibriVox project, and has been carefully segmented and aligned.87
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+ """
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+
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+ _URL = "http://www.openslr.org/12"
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+ _DL_URL = "http://www.openslr.org/resources/12/"
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+
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+
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+ _DL_URLS = {
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+ "clean": {
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+ "dev": _DL_URL + "dev-clean.tar.gz",
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+ "train.100": _DL_URL + "train-clean-100.tar.gz",
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+ },
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+ }
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+
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+
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+ class LibrispeechASRConfig(datasets.BuilderConfig):
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+ """BuilderConfig for LibriSpeechASR."""
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+
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+ def __init__(self, **kwargs):
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+ """
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+ Args:
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+ data_dir: `string`, the path to the folder containing the files in the
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+ downloaded .tar
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+ citation: `string`, citation for the data set
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+ url: `string`, url for information about the data set
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+ **kwargs: keyword arguments forwarded to super.
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+ """
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+ super(LibrispeechASRConfig, self).__init__(version=datasets.Version("2.1.0", ""), **kwargs)
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+
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+
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+ class LibrispeechASR(datasets.GeneratorBasedBuilder):
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+ """Librispeech dataset."""
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+
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+ DEFAULT_WRITER_BATCH_SIZE = 256
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+ DEFAULT_CONFIG_NAME = "all"
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+ BUILDER_CONFIGS = [
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+ LibrispeechASRConfig(name="clean", description="'Clean' speech."),
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+ ]
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+
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+ def _info(self):
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+ return datasets.DatasetInfo(
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+ description=_DESCRIPTION,
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+ features=datasets.Features(
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+ {
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+ "file": datasets.Value("string"),
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+ "audio": datasets.Audio(sampling_rate=16_000),
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+ "text": datasets.Value("string"),
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+ "speaker_id": datasets.Value("int64"),
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+ "chapter_id": datasets.Value("int64"),
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+ "id": datasets.Value("string"),
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+ }
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+ ),
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+ supervised_keys=("file", "text"),
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+ homepage=_URL,
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+ citation=_CITATION,
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+ task_templates=[AutomaticSpeechRecognition(audio_column="audio", transcription_column="text")],
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+ )
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+
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+ def _split_generators(self, dl_manager):
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+ archive_path = dl_manager.download(_DL_URLS[self.config.name])
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+ # (Optional) In non-streaming mode, we can extract the archive locally to have actual local audio files:
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+ local_extracted_archive = dl_manager.extract(archive_path) if not dl_manager.is_streaming else {}
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+
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+ if self.config.name == "clean":
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+ train_splits = [
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+ datasets.SplitGenerator(
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+ name="train.100",
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+ gen_kwargs={
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+ "local_extracted_archive": local_extracted_archive.get("train.100"),
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+ "files": dl_manager.iter_archive(archive_path["train.100"]),
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+ },
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+ ),
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+ ]
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+ dev_splits = [
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+ datasets.SplitGenerator(
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+ name=datasets.Split.VALIDATION,
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+ gen_kwargs={
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+ "local_extracted_archive": local_extracted_archive.get("dev"),
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+ "files": dl_manager.iter_archive(archive_path["dev"]),
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+ },
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+ )
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+ ]
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+
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+ return train_splits + dev_splits
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+
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+ def _generate_examples(self, files, local_extracted_archive):
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+ """Generate examples from a LibriSpeech archive_path."""
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+ key = 0
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+ audio_data = {}
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+ transcripts = []
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+ for path, f in files:
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+ if path.endswith(".flac"):
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+ id_ = path.split("/")[-1][: -len(".flac")]
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+ audio_data[id_] = f.read()
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+ elif path.endswith(".trans.txt"):
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+ for line in f:
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+ if line:
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+ line = line.decode("utf-8").strip()
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+ id_, transcript = line.split(" ", 1)
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+ audio_file = f"{id_}.flac"
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+ speaker_id, chapter_id = [int(el) for el in id_.split("-")[:2]]
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+ audio_file = (
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+ os.path.join(local_extracted_archive, audio_file)
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+ if local_extracted_archive
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+ else audio_file
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+ )
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+ transcripts.append(
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+ {
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+ "id": id_,
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+ "speaker_id": speaker_id,
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+ "chapter_id": chapter_id,
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+ "file": audio_file,
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+ "text": transcript,
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+ }
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+ )
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+ if audio_data and len(audio_data) == len(transcripts):
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+ for transcript in transcripts:
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+ audio = {"path": transcript["file"], "bytes": audio_data[transcript["id"]]}
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+ yield key, {"audio": audio, **transcript}
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+ key += 1
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+ audio_data = {}
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+ transcripts = []