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librispeech_asr.py.lock ADDED
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librispeech_asr_dummy.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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+ from __future__ import absolute_import, division, print_function
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+
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+ import glob
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+ import os
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+
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+ import datasets
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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.
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+
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+ Note that in order to limit the required storage for preparing this dataset, the audio
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+ is stored in the .flac format and is not converted to a float32 array. To convert, the audio
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+ file to a float32 array, please make use of the `.map()` function as follows:
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+
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+
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+ ```python
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+ import soundfile as sf
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+
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+ def map_to_array(batch):
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+ speech_array, _ = sf.read(batch["file"])
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+ batch["speech"] = speech_array
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+ return batch
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+
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+ dataset = dataset.map(map_to_array, remove_columns=["file"])
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+ ```
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+ """
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+
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+ _URL = "http://www.openslr.org/12"
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+ _DL_URL = "https://s3.amazonaws.com/datasets.huggingface.co/librispeech_asr/2.1.0/"
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+ _DL_URL = "https://s3.amazonaws.com/datasets.huggingface.co/librispeech_asr/2.1.0/"
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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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+ }
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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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+ BUILDER_CONFIGS = [
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+ LibrispeechASRConfig(name="clean", description="'Clean' speech."),
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+ LibrispeechASRConfig(name="other", description="'Other', more challenging, 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.features.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=("speech", "text"),
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+ homepage=_URL,
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+ citation=_CITATION,
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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_and_extract(_DL_URLS[self.config.name])
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+ return [
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+ datasets.SplitGenerator(name=datasets.Split.VALIDATION, gen_kwargs={"archive_path": archive_path["dev"], "split_name": f"dev_{self.config.name}"}),
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+ ]
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+
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+ def _generate_examples(self, archive_path, split_name):
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+ """Generate examples from a Librispeech archive_path."""
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+ transcripts_glob = os.path.join(archive_path, split_name, "*/*/*.txt")
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+ for transcript_file in glob.glob(transcripts_glob):
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+ path = os.path.dirname(transcript_file)
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+ with open(os.path.join(path, transcript_file)) as f:
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+ for line in f:
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+ line = line.strip()
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+ key, transcript = line.split(" ", 1)
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+ audio_file = f"{key}.flac"
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+ speaker_id, chapter_id = [int(el) for el in key.split("-")[:2]]
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+ example = {
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+ "id": key,
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+ "speaker_id": speaker_id,
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+ "chapter_id": chapter_id,
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+ "file": os.path.join(path, audio_file),
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+ # "audio": os.path.join(path, audio_file),
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+ "text": transcript,
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+ }
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+ yield key, example
librispeech_asr_dummy.py.lock ADDED
File without changes