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, 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
- 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 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:
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])