Add loading script
Browse files- asr_dummy.py +186 -0
asr_dummy.py
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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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# Lint as: python3
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"""SUPERB: Speech processing Universal PERformance Benchmark."""
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import glob
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import os
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import textwrap
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import datasets
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from datasets.tasks import AutomaticSpeechRecognition
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_CITATION = """\
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@article{DBLP:journals/corr/abs-2105-01051,
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author = {Shu{-}Wen Yang and
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Po{-}Han Chi and
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Yung{-}Sung Chuang and
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Cheng{-}I Jeff Lai and
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Kushal Lakhotia and
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Yist Y. Lin and
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Andy T. Liu and
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Jiatong Shi and
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Xuankai Chang and
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Guan{-}Ting Lin and
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Tzu{-}Hsien Huang and
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Wei{-}Cheng Tseng and
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Ko{-}tik Lee and
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Da{-}Rong Liu and
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Zili Huang and
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Shuyan Dong and
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Shang{-}Wen Li and
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Shinji Watanabe and
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Abdelrahman Mohamed and
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Hung{-}yi Lee},
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title = {{SUPERB:} Speech processing Universal PERformance Benchmark},
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journal = {CoRR},
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volume = {abs/2105.01051},
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year = {2021},
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url = {https://arxiv.org/abs/2105.01051},
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archivePrefix = {arXiv},
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eprint = {2105.01051},
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timestamp = {Thu, 01 Jul 2021 13:30:22 +0200},
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biburl = {https://dblp.org/rec/journals/corr/abs-2105-01051.bib},
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bibsource = {dblp computer science bibliography, https://dblp.org}
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}
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"""
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_DESCRIPTION = """\
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Self-supervised learning (SSL) has proven vital for advancing research in
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natural language processing (NLP) and computer vision (CV). The paradigm
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pretrains a shared model on large volumes of unlabeled data and achieves
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state-of-the-art (SOTA) for various tasks with minimal adaptation. However, the
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speech processing community lacks a similar setup to systematically explore the
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paradigm. To bridge this gap, we introduce Speech processing Universal
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PERformance Benchmark (SUPERB). SUPERB is a leaderboard to benchmark the
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performance of a shared model across a wide range of speech processing tasks
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with minimal architecture changes and labeled data. Among multiple usages of the
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shared model, we especially focus on extracting the representation learned from
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SSL due to its preferable re-usability. We present a simple framework to solve
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SUPERB tasks by learning task-specialized lightweight prediction heads on top of
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the frozen shared model. Our results demonstrate that the framework is promising
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as SSL representations show competitive generalizability and accessibility
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across SUPERB tasks. We release SUPERB as a challenge with a leaderboard and a
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benchmark toolkit to fuel the research in representation learning and general
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speech processing.
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Note that in order to limit the required storage for preparing this dataset, the
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audio is stored in the .flac format and is not converted to a float32 array. To
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convert, the audio file to a float32 array, please make use of the `.map()`
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function as follows:
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```python
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import soundfile as sf
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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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dataset = dataset.map(map_to_array, remove_columns=["file"])
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```
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"""
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class AsrDummybConfig(datasets.BuilderConfig):
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"""BuilderConfig for Superb."""
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def __init__(
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self,
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data_url,
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url,
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task_templates=None,
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**kwargs,
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):
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super(AsrDummybConfig, self).__init__(version=datasets.Version("1.9.0", ""), **kwargs)
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self.data_url = data_url
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self.url = url
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self.task_templates = task_templates
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class AsrDummy(datasets.GeneratorBasedBuilder):
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"""Superb dataset."""
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BUILDER_CONFIGS = [
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AsrDummybConfig(
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name="asr",
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description=textwrap.dedent(
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"""\
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ASR transcribes utterances into words. While PR analyzes the
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improvement in modeling phonetics, ASR reflects the significance of
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the improvement in a real-world scenario. LibriSpeech
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train-clean-100/dev-clean/test-clean subsets are used for
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training/validation/testing. The evaluation metric is word error
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rate (WER)."""
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),
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url="http://www.openslr.org/12",
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data_url="http://www.openslr.org/resources/12/",
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task_templates=[AutomaticSpeechRecognition(audio_file_path_column="file", transcription_column="text")],
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)
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]
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DEFAULT_CONFIG_NAME = "asr"
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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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"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=self.config.url,
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citation=_CITATION,
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task_templates=self.config.task_templates,
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)
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def _split_generators(self, dl_manager):
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_DL_URLS = {
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"test": self.config.data_url + "test-clean.tar.gz",
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}
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archive_path = dl_manager.download_and_extract(_DL_URLS)
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return [
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datasets.SplitGenerator(name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path["test"]}),
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]
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def _generate_examples(self, archive_path):
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"""Generate examples."""
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transcripts_glob = os.path.join(archive_path, "LibriSpeech", "*/*/*/*.txt")
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for transcript_file in sorted(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), "r", encoding="utf-8") 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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"text": transcript,
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}
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yield key, example
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