saisamarth commited on
Commit
727941c
·
1 Parent(s): 9cb0e1a

Create librispeech_asr.py

Browse files
Files changed (1) hide show
  1. librispeech_asr.py +155 -0
librispeech_asr.py ADDED
@@ -0,0 +1,155 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ # coding=utf-8
2
+ # Copyright 2021 The TensorFlow Datasets Authors and the HuggingFace Datasets Authors.
3
+ #
4
+ # Licensed under the Apache License, Version 2.0 (the "License");
5
+ # you may not use this file except in compliance with the License.
6
+ # You may obtain a copy of the License at
7
+ #
8
+ # http://www.apache.org/licenses/LICENSE-2.0
9
+ #
10
+ # Unless required by applicable law or agreed to in writing, software
11
+ # distributed under the License is distributed on an "AS IS" BASIS,
12
+ # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13
+ # See the License for the specific language governing permissions and
14
+ # limitations under the License.
15
+
16
+ # Lint as: python3
17
+ """Librispeech automatic speech recognition dataset."""
18
+
19
+
20
+ import os
21
+
22
+ import datasets
23
+ from datasets.tasks import AutomaticSpeechRecognition
24
+
25
+
26
+ _CITATION = """\
27
+ @inproceedings{panayotov2015librispeech,
28
+ title={Librispeech: an ASR corpus based on public domain audio books},
29
+ author={Panayotov, Vassil and Chen, Guoguo and Povey, Daniel and Khudanpur, Sanjeev},
30
+ booktitle={Acoustics, Speech and Signal Processing (ICASSP), 2015 IEEE International Conference on},
31
+ pages={5206--5210},
32
+ year={2015},
33
+ organization={IEEE}
34
+ }
35
+ """
36
+
37
+ _DESCRIPTION = """\
38
+ LibriSpeech is a corpus of approximately 1000 hours of read English speech with sampling rate of 16 kHz,
39
+ prepared by Vassil Panayotov with the assistance of Daniel Povey. The data is derived from read
40
+ audiobooks from the LibriVox project, and has been carefully segmented and aligned.87
41
+ """
42
+
43
+ _URL = "http://www.openslr.org/12"
44
+ _DL_URL = "http://www.openslr.org/resources/12/"
45
+
46
+
47
+ _DL_URLS = {
48
+ "clean": {
49
+ "dev": _DL_URL + "dev-clean.tar.gz",
50
+ },
51
+
52
+ }
53
+
54
+
55
+ class LibrispeechASRConfig(datasets.BuilderConfig):
56
+ """BuilderConfig for LibriSpeechASR."""
57
+
58
+ def __init__(self, **kwargs):
59
+ """
60
+ Args:
61
+ data_dir: `string`, the path to the folder containing the files in the
62
+ downloaded .tar
63
+ citation: `string`, citation for the data set
64
+ url: `string`, url for information about the data set
65
+ **kwargs: keyword arguments forwarded to super.
66
+ """
67
+ super(LibrispeechASRConfig, self).__init__(version=datasets.Version("2.1.0", ""), **kwargs)
68
+
69
+
70
+ class LibrispeechASR(datasets.GeneratorBasedBuilder):
71
+ """Librispeech dataset."""
72
+
73
+ DEFAULT_WRITER_BATCH_SIZE = 256
74
+ DEFAULT_CONFIG_NAME = "all"
75
+ BUILDER_CONFIGS = [
76
+ LibrispeechASRConfig(name="clean", description="'Clean' speech."),
77
+ LibrispeechASRConfig(name="other", description="'Other', more challenging, speech."),
78
+ LibrispeechASRConfig(name="all", description="Combined clean and other dataset."),
79
+ ]
80
+
81
+ def _info(self):
82
+ return datasets.DatasetInfo(
83
+ description=_DESCRIPTION,
84
+ features=datasets.Features(
85
+ {
86
+ "file": datasets.Value("string"),
87
+ "audio": datasets.Audio(sampling_rate=16_000),
88
+ "text": datasets.Value("string"),
89
+ "speaker_id": datasets.Value("int64"),
90
+ "chapter_id": datasets.Value("int64"),
91
+ "id": datasets.Value("string"),
92
+ }
93
+ ),
94
+ supervised_keys=("file", "text"),
95
+ homepage=_URL,
96
+ citation=_CITATION,
97
+ task_templates=[AutomaticSpeechRecognition(audio_column="audio", transcription_column="text")],
98
+ )
99
+
100
+ def _split_generators(self, dl_manager):
101
+ archive_path = dl_manager.download(_DL_URLS[self.config.name])
102
+ # (Optional) In non-streaming mode, we can extract the archive locally to have actual local audio files:
103
+ local_extracted_archive = dl_manager.extract(archive_path) if not dl_manager.is_streaming else {}
104
+
105
+ if self.config.name == "clean":
106
+
107
+ dev_splits = [
108
+ datasets.SplitGenerator(
109
+ name=datasets.Split.VALIDATION,
110
+ gen_kwargs={
111
+ "local_extracted_archive": local_extracted_archive.get("dev"),
112
+ "files": dl_manager.iter_archive(archive_path["dev"]),
113
+ },
114
+ )
115
+ ]
116
+
117
+ return dev_splits
118
+
119
+ def _generate_examples(self, files, local_extracted_archive):
120
+ """Generate examples from a LibriSpeech archive_path."""
121
+ key = 0
122
+ audio_data = {}
123
+ transcripts = []
124
+ for path, f in files:
125
+ if path.endswith(".flac"):
126
+ id_ = path.split("/")[-1][: -len(".flac")]
127
+ audio_data[id_] = f.read()
128
+ elif path.endswith(".trans.txt"):
129
+ for line in f:
130
+ if line:
131
+ line = line.decode("utf-8").strip()
132
+ id_, transcript = line.split(" ", 1)
133
+ audio_file = f"{id_}.flac"
134
+ speaker_id, chapter_id = [int(el) for el in id_.split("-")[:2]]
135
+ audio_file = (
136
+ os.path.join(local_extracted_archive, audio_file)
137
+ if local_extracted_archive
138
+ else audio_file
139
+ )
140
+ transcripts.append(
141
+ {
142
+ "id": id_,
143
+ "speaker_id": speaker_id,
144
+ "chapter_id": chapter_id,
145
+ "file": audio_file,
146
+ "text": transcript,
147
+ }
148
+ )
149
+ if audio_data and len(audio_data) == len(transcripts):
150
+ for transcript in transcripts:
151
+ audio = {"path": transcript["file"], "bytes": audio_data[transcript["id"]]}
152
+ yield key, {"audio": audio, **transcript}
153
+ key += 1
154
+ audio_data = {}
155
+ transcripts = []