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MERaLiON/sea_audiobench_datasets_TLoc

SEA-SpeechBench — TLoc (Temporal Localization) This dataset is the temporal-localization (TLoc) task of SEA-SpeechBench, a large-scale multitask benchmark for speech understanding across Southeast Asia. It contains 6,736 evaluation examples across five languages, each pairing an audio recording of 30–180 seconds with an instruction and a reference answer. Given the recording and the instruction, a model must identify when a described utterance occurs. Quick start… See the full description on the dataset page: https://huggingface.co/datasets/MERaLiON/sea_audiobench_datasets_TLoc.

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SEA-SpeechBench — TLoc (Temporal Localization)

This dataset is the temporal-localization (TLoc) task of SEA-SpeechBench, a large-scale multitask benchmark for speech understanding across Southeast Asia. It contains 6,736 evaluation examples across five languages, each pairing an audio recording of 30–180 seconds with an instruction and a reference answer.

Given the recording and the instruction, a model must identify when a described utterance occurs.

Quick start

Requires datasets>=4.0, which decodes audio through torchcodec and needs a system FFmpeg (versions 4–7):

bash
pip install "datasets[audio]"
python
from datasets import load_dataset

ds = load_dataset(
    "MERaLiON/sea_audiobench_datasets_TLoc",
    "tloc_yodas2_en_30",
    split="train",
)

row = ds[0]
print(row["instruction"])
print(row["answer"])
print(row["language"], row["audio_length"])

# `context` is a torchcodec AudioDecoder, not a dict
samples = row["context"].get_all_samples()
waveform = samples.data          # torch.Tensor, shape (num_channels, num_samples)
sr = samples.sample_rate         # 16000
print(waveform.shape, sr)

If you need a numpy array — most feature extractors take one — squeeze the channel dimension (all TLoc audio is mono):

python
audio = samples.data.squeeze(0).numpy()

Subsets

All data lives in a single train split and is intended for evaluation only. Config names follow tloc_{source}_{language}_{context_length_seconds}.

ConfigSourceLangCtx (s)ExamplesNominal hrs
tloc_sg_streets_30SG Streetsen30950.8
tloc_sgpccsc_long_30ASR-SgpCCSCzh305124.3
tloc_yodas2_en_30YODAS2en301,0008.3
tloc_yodas2_id_30YODAS2id301,0008.3
tloc_yodas2_th_30YODAS2th301,0008.3
tloc_yodas2_vi_30YODAS2vi304894.1
tloc_yodas2_vi_60YODAS2vi603415.7
tloc_yodas2_vi_120YODAS2vi1201635.4
tloc_yodas2_vi_180YODAS2vi180623.1
tloc_yodas2_zh_30YODAS2zh301,0008.3
tloc_yodas2_zh_60YODAS2zh6075412.6
tloc_yodas2_zh_120YODAS2zh1202719.0
tloc_yodas2_zh_180YODAS2zh180492.5
Total6,736~80.7

Data fields

FieldTypeDescription
contextAudio(sampling_rate=16000)The audio recording.
instructionstringThe instruction given to the model.
answerstringThe reference answer.
audio_lengthfloat64Duration of context, in seconds.
languagestringLanguage code of the subset.

See the paper for the instruction and answer formats, how segments and annotations were constructed, and the evaluation protocol.

Source data

TLoc introduces no new recordings. All audio comes from existing corpora; the contribution is the segmentation, instructions, and reference answers.

SourceLangUpstream license
YODAS2en, id, zh, th, viCC BY 3.0
SG StreetsenNot specified
ASR-SgpCCSCzhCC BY-NC-ND 4.0

Citation

If you use TLoc, please cite our benchmark and the source corpora.

This benchmark

bibtex
@inproceedings{liao2026seaspeechbench,
  title     = {{SEA-SpeechBench}: A Large-Scale Multitask Benchmark for
               Speech Understanding Across Southeast Asia},
  author    = {Liao, Jingyi and Zhang, Wenyu and Liu, Zhuohan and
               He, Yingxu and Lin, Geyu and Zou, Xunlong and Sun, Shuo and
               Alsagoff, Syed Ali Redha and Aw, Ai Ti},
  booktitle = {Proceedings of the 2026 Conference on Empirical Methods in
               Natural Language Processing (EMNLP)},
  year      = {2026}
}

Source corpora

bibtex
@inproceedings{li2023yodas,
  title        = {YODAS: Youtube-Oriented Dataset for Audio and Speech},
  author       = {Li, Xinjian and Takamichi, Shinnosuke and Saeki, Takaaki
                  and Chen, William and Shiota, Sayaka and Watanabe, Shinji},
  booktitle    = {2023 IEEE Automatic Speech Recognition and Understanding
                  Workshop (ASRU)},
  pages        = {1--8},
  year         = {2023},
  organization = {IEEE}
}

@misc{sgpccsc,
  title  = {{ASR-SgpCCSC: A Singapore Chinese Conversational Speech Corpus}},
  author = {Magic Data Tech},
  year   = {2024},
  url    = {https://magichub.com/datasets/[SLUG]/}
}

@inproceedings{khassanov2019enriching,
  title     = {Enriching Rare Word Representations in Neural Language Models
               by Embedding Matrix Augmentation},
  author    = {Khassanov, Yerbolat and Zeng, Zhiping and Pham, Van Tung and
               Xu, Haihua and Chng, Eng Siong},
  booktitle = {Interspeech 2019},
  pages     = {3505--3509},
  year      = {2019},
  month     = sep,
  publisher = {ISCA},
  doi       = {10.21437/Interspeech.2019-1858},
  url       = {http://dx.doi.org/10.21437/Interspeech.2019-1858}
}