Datasetpublic

changelinglab/librispeech-segment

LibriSpeech Segment English read-speech corpus with phone-level time alignments (Montreal Forced Aligner). Suitable for training and evaluating phone recognition and phonetic segmentation models. Sources Audio: LibriSpeech (OpenSLR 12) by Vassil Panayotov, Guoguo Chen, Daniel Povey, Sanjeev Khudanpur (2015). Phone alignments: anyspeech/librispeech_MFA_alignments. Splits Split Utterances train.clean.100 28,538 train.clean.360 104,008… See the full description on the dataset page: https://huggingface.co/datasets/changelinglab/librispeech-segment.

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Dataset Card

LibriSpeech Segment

English read-speech corpus with phone-level time alignments (Montreal Forced Aligner). Suitable for training and evaluating phone recognition and phonetic segmentation models.

Sources

—Audio: LibriSpeech (OpenSLR 12) by Vassil Panayotov, Guoguo Chen, Daniel Povey, Sanjeev Khudanpur (2015).
—Phone alignments: `anyspeech/librispeech_MFA_alignments`.

Splits

Split	Utterances
train.clean.100	28,538
train.clean.360	104,008
train.other.500	148,645
dev.clean	2,703
dev.other	2,864
test.clean	2,620
test.other	2,938

Split labels follow the LibriSpeech canonical naming.

Schema

Column	Type	Description
`utt_id`	string	Utterance id, e.g. `7635-105409-0022`
`audio`	Audio(16 kHz)	Embedded waveform bytes (decoded on access)
`text`	string	Word-level transcript (uppercase)
`phones`	sequence[string]	ARPABET phone tokens
`phone_starts`	sequence[float64]	Phone start times in seconds
`phone_ends`	sequence[float64]	Phone end times in seconds
`language`	string	`eng` (ISO 639-3)
`speaker_id`	string	LibriSpeech speaker id
`duration`	float64	Utterance duration in seconds
`split`	string	LibriSpeech split label

Phone inventory

Phones are ARPABET (e.g. DH, EH, R, AE, OW). Silence and pauses are marked with [SIL] intervals, kept in the alignment so boundary models can learn from them. [UNK] may also appear for OOV cases.

License

Released under the CC BY 4.0 license, matching the original LibriSpeech audio.

Citation

bibtex

@inproceedings{panayotov2015librispeech,
  title={Librispeech: an asr corpus based on public domain audio books},
  author={Panayotov, Vassil and Chen, Guoguo and Povey, Daniel and Khudanpur, Sanjeev},
  booktitle={2015 IEEE international conference on acoustics, speech and signal processing (ICASSP)},
  pages={5206--5210},
  year={2015},
  organization={IEEE}
}