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Revert "Convert dataset to Parquet (#4)"

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This reverts commit 01983bc6601bb765424785c0b41a3a2486469038.

README.md CHANGED
@@ -1,4 +1,5 @@
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  ---
 
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  annotations_creators:
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  - expert-generated
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  language_creators:
@@ -9,6 +10,7 @@ license:
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  - cc-by-4.0
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  multilinguality:
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  - monolingual
 
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  size_categories:
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  - 1K<n<10K
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  source_datasets:
@@ -16,10 +18,22 @@ source_datasets:
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  task_categories:
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  - automatic-speech-recognition
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  task_ids: []
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- paperswithcode_id: arabic-speech-corpus
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- pretty_name: Arabic Speech Corpus
 
 
 
 
 
 
 
 
 
 
 
 
 
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  dataset_info:
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- config_name: clean
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  features:
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  - name: file
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  dtype: string
@@ -33,38 +47,16 @@ dataset_info:
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  dtype: string
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  - name: orthographic
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  dtype: string
 
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  splits:
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  - name: train
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- num_bytes: 1527815416.966
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  num_examples: 1813
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  - name: test
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- num_bytes: 99851729.0
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  num_examples: 100
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- download_size: 1347643373
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- dataset_size: 1627667145.966
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- configs:
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- - config_name: clean
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- data_files:
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- - split: train
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- path: clean/train-*
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- - split: test
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- path: clean/test-*
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- default: true
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- train-eval-index:
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- - config: clean
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- task: automatic-speech-recognition
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- task_id: speech_recognition
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- splits:
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- train_split: train
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- eval_split: test
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- col_mapping:
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- file: path
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- text: text
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- metrics:
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- - type: wer
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- name: WER
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- - type: cer
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- name: CER
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  ---
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  # Dataset Card for Arabic Speech Corpus
 
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  ---
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+ pretty_name: Arabic Speech Corpus
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  annotations_creators:
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  - expert-generated
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  language_creators:
 
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  - cc-by-4.0
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  multilinguality:
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  - monolingual
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+ paperswithcode_id: arabic-speech-corpus
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  size_categories:
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  - 1K<n<10K
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  source_datasets:
 
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  task_categories:
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  - automatic-speech-recognition
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  task_ids: []
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+ train-eval-index:
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+ - config: clean
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+ task: automatic-speech-recognition
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+ task_id: speech_recognition
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+ splits:
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+ train_split: train
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+ eval_split: test
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+ col_mapping:
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+ file: path
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+ text: text
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+ metrics:
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+ - type: wer
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+ name: WER
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+ - type: cer
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+ name: CER
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  dataset_info:
 
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  features:
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  - name: file
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  dtype: string
 
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  dtype: string
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  - name: orthographic
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  dtype: string
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+ config_name: clean
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  splits:
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  - name: train
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+ num_bytes: 1002365
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  num_examples: 1813
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  - name: test
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+ num_bytes: 65784
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  num_examples: 100
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+ download_size: 1192302846
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+ dataset_size: 1068149
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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  ---
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  # Dataset Card for Arabic Speech Corpus
arabic_speech_corpus.py ADDED
@@ -0,0 +1,145 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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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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+ """Arabic Speech Corpus"""
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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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+ @phdthesis{halabi2016modern,
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+ title={Modern standard Arabic phonetics for speech synthesis},
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+ author={Halabi, Nawar},
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+ year={2016},
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+ school={University of Southampton}
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+ }
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+ """
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+
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+ _DESCRIPTION = """\
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+ This Speech corpus has been developed as part of PhD work carried out by Nawar Halabi at the University of Southampton.
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+ The corpus was recorded in south Levantine Arabic
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+ (Damascian accent) using a professional studio. Synthesized speech as an output using this corpus has produced a high quality, natural voice.
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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://en.arabicspeechcorpus.com/arabic-speech-corpus.zip"
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+
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+
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+ class ArabicSpeechCorpusConfig(datasets.BuilderConfig):
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+ """BuilderConfig for ArabicSpeechCorpu."""
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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(ArabicSpeechCorpusConfig, self).__init__(version=datasets.Version("2.1.0", ""), **kwargs)
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+
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+
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+ class ArabicSpeechCorpus(datasets.GeneratorBasedBuilder):
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+ """ArabicSpeechCorpus dataset."""
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+
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+ BUILDER_CONFIGS = [
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+ ArabicSpeechCorpusConfig(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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+ "text": datasets.Value("string"),
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+ "audio": datasets.Audio(sampling_rate=48_000),
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+ "phonetic": datasets.Value("string"),
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+ "orthographic": 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):
100
+ archive_path = dl_manager.download_and_extract(_URL)
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+ archive_path = os.path.join(archive_path, "arabic-speech-corpus")
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+ return [
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+ datasets.SplitGenerator(name="train", gen_kwargs={"archive_path": archive_path}),
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+ datasets.SplitGenerator(name="test", gen_kwargs={"archive_path": os.path.join(archive_path, "test set")}),
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+ ]
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+
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+ def _generate_examples(self, archive_path):
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+ """Generate examples from a Librispeech archive_path."""
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+ lab_dir = os.path.join(archive_path, "lab")
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+ wav_dir = os.path.join(archive_path, "wav")
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+ if "test set" in archive_path:
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+ phonetic_path = os.path.join(archive_path, "phonetic-transcript.txt")
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+ else:
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+ phonetic_path = os.path.join(archive_path, "phonetic-transcipt.txt")
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+
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+ orthographic_path = os.path.join(archive_path, "orthographic-transcript.txt")
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+
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+ phonetics = {}
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+ orthographics = {}
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+
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+ with open(phonetic_path, "r", encoding="utf-8") as f:
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+ for line in f:
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+ wav_file, phonetic = line.split('"')[1::2]
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+ phonetics[wav_file] = phonetic
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+
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+ with open(orthographic_path, "r", encoding="utf-8") as f:
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+ for line in f:
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+ wav_file, orthographic = line.split('"')[1::2]
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+ orthographics[wav_file] = orthographic
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+
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+ for _id, lab_name in enumerate(sorted(os.listdir(lab_dir))):
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+ lab_path = os.path.join(lab_dir, lab_name)
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+ lab_text = open(lab_path, "r", encoding="utf-8").read()
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+
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+ wav_name = lab_name[:-4] + ".wav"
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+ wav_path = os.path.join(wav_dir, wav_name)
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+
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+ example = {
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+ "file": wav_path,
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+ "audio": wav_path,
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+ "text": lab_text,
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+ "phonetic": phonetics[wav_name],
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+ "orthographic": orthographics[wav_name],
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+ }
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+ yield str(_id), example
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