--- dataset_info: features: - name: file dtype: string - name: audio struct: - name: array sequence: float64 - name: path dtype: string - name: sampling_rate dtype: int64 - name: text dtype: string - name: speaker_id dtype: int64 - name: chapter_id dtype: int64 - name: id dtype: string splits: - name: train.clean.100 num_bytes: 1623641436 num_examples: 1000 - name: train.clean.360 num_bytes: 1572285643 num_examples: 1000 - name: train.other.500 num_bytes: 1502809029 num_examples: 1000 - name: validation.clean num_bytes: 65591952 num_examples: 100 - name: validation.other num_bytes: 76760504 num_examples: 100 - name: test.clean num_bytes: 85852252 num_examples: 100 - name: test.other num_bytes: 58550856 num_examples: 100 download_size: 1181170369 dataset_size: 4985491672 configs: - config_name: default data_files: - split: train.clean.100 path: data/train.clean.100-* - split: train.clean.360 path: data/train.clean.360-* - split: train.other.500 path: data/train.other.500-* - split: validation.clean path: data/validation.clean-* - split: validation.other path: data/validation.other-* - split: test.clean path: data/test.clean-* - split: test.other path: data/test.other-* --- # Condensed LibriSpeech ASR This dataset is a condensed version of the [LibriSpeech ASR dataset](https://huggingface.co/datasets/openslr/librispeech_asr), created by subsampling approximately 10% of the original data from each split. It is intended for quick experimentation, prototyping, and debugging when working with Automatic Speech Recognition (ASR) tasks. ## Dataset Details - **Original Dataset:** [LibriSpeech ASR](https://huggingface.co/datasets/openslr/librispeech_asr) - **Condensation Ratio:** Approximately 10% of the full dataset - **Splits Included:** - `train.clean.100` - `train.clean.360` - `train.other.500` - `validation.clean` - `validation.other` - `test.clean` - `test.other` For each split, a default number of examples was extracted: - **Training Splits:** 1,000 examples each - **Validation/Test Splits:** 100 examples each ## Data Format Each sample in the dataset contains the following fields: - **file:** A path to the original audio file (FLAC format). - **audio:** A dictionary containing: - `path`: Path to the audio file. - `array`: The decoded audio waveform as a NumPy array. - `sampling_rate`: The sampling rate (typically 16 kHz). - **text:** The transcription corresponding to the audio. - **id:** A unique identifier for the sample. - **speaker_id:** A unique identifier for the speaker. - **chapter_id:** An identifier corresponding to the audiobook chapter. ## How Was This Dataset Created? The condensed dataset was generated by streaming the full [LibriSpeech ASR dataset](https://huggingface.co/datasets/openslr/librispeech_asr) using the Hugging Face Datasets library and selecting approximately 10% of each split. This process preserves the original structure and fields, enabling seamless use with models and workflows designed for LibriSpeech. ## Usage Example Below is a Python snippet to load and inspect the dataset: ```python from datasets import load_dataset # Load the condensed dataset from the Hugging Face Hub dataset = load_dataset("nyalpatel/condensed_librispeech_asr") # Access a specific split (e.g., test.clean) test_dataset = dataset["test.clean"] # Display the first example in the test set print(test_dataset[0])