changelinglab/librispeech-segment

--- license: cc-by-4.0 language: - en pretty_name: LibriSpeech Segment task_categories: - automatic-speech-recognition tags: - speech - phone-alignment - segmentation - english size_categories: - 100K<n<1M --- # 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](https://www.openslr.org/12/) (OpenSLR 12) by Vassil Panayotov, Guoguo Chen, Daniel Povey, Sanjeev Khudanpur (2015). - **Phone alignments**: [`anyspeech/librispeech_MFA_alignments`](https://huggingface.co/datasets/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} } ```--- license: CC BY 4.0 language: - en pretty_name: LibriSpeech Segment task_categories: - 自动语音识别（Automatic Speech Recognition） tags: - 语音 - 音素对齐（phone-alignment） - 语音分割（segmentation） - 英语 size_categories: - 100K<n<1M --- # LibriSpeech Segment 英语朗读语音语料库，带有**音素级时间对齐（phone-level time alignments）**（由蒙特利尔强制对齐器（Montreal Forced Aligner）生成），适用于音素识别模型与语音分割模型的训练与评估。 ## 数据来源 - **音频**：[LibriSpeech](https://www.openslr.org/12/)（OpenSLR 12），作者为Vassil Panayotov、Guoguo Chen、Daniel Povey、Sanjeev Khudanpur（2015年）。 - **音素对齐结果**：[`anyspeech/librispeech_MFA_alignments`](https://huggingface.co/datasets/anyspeech/librispeech_MFA_alignments)。 ## 拆分集 | 拆分标签 | 话语总数 | |-------------------|------------| | 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 | 拆分标签遵循LibriSpeech的标准命名规范。 ## 数据结构 | 列名 | 类型 | 描述 | |----------------|----------------------|------------------------------------------------------| | `utt_id` | 字符串 | 话语ID，例如 `7635-105409-0022` | | `audio` | 音频（16千赫（kHz）） | 嵌入式音频波形字节（访问时自动解码） | | `text` | 字符串 | 词级转录文本（全大写） | | `phones` | 字符串序列 | ARPABET音素标记（Token） | | `phone_starts` | 64位浮点数序列 | 音素起始时间（单位：秒） | | `phone_ends` | 64位浮点数序列 | 音素结束时间（单位：秒） | | `language` | 字符串 | `eng`（ISO 639-3 标准语言代码） | | `speaker_id` | 字符串 | LibriSpeech说话人ID | | `duration` | 64位浮点数 | 话语时长（单位：秒） | | `split` | 字符串 | LibriSpeech拆分集标签 | ## 音素表 音素采用ARPABET编码体系（例如 `DH`、`EH`、`R`、`AE`、`OW`）。静音与停顿以`[SIL]`区间标记，保留在对齐结果中以便边界模型学习。针对未登录词（Out-of-Vocabulary, OOV）场景，可能出现`[UNK]`标记。 ## 许可证 本数据集采用**CC BY 4.0**许可证发布，与原始LibriSpeech音频的许可证保持一致。 ## 引用 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} }