# Wav2Vec 2 audio processing

**URL:** <https://discuss.huggingface.co/t/wav2vec-2-audio-processing/90127>\
**Category:** Models\
**Created:** [June 3, 2024, 7:28pm UTC](https://discuss.huggingface.co/t/wav2vec-2-audio-processing/90127 "2024-06-03T19:28:05Z")\
**Posts on this page:** 1\
**Page:** 1

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**Author:** ![alejopaullier](https://sea2.discourse-cdn.com/hellohellohello/user_avatar/discuss.huggingface.co/alejopaullier/32/27750_2.png) [@alejopaullier](https://discuss.huggingface.co/u/alejopaullier)\
**Post date:** [June 3, 2024, 7:28pm UTC](https://discuss.huggingface.co/t/wav2vec-2-audio-processing/90127/1 "2024-06-03T19:28:05Z")

</div>

I am currently using `facebook/wav2vec2-base` model for an audio classification task. My code is based on the official HF Audio Classification [tutorial](https://huggingface.co/docs/transformers/en/tasks/audio_classification).

In this tutorial, audios are sampled using a `sample_rate` of 16.000. This means that 1 second of audio results in an array of length 16.000. Also, `Wav2Vec2` maximum input length is 150.000 as far as I know.

Does this mean that, without any chunking, the model can only process audios just up to ~10 seconds?

If you have longer audios, which i guess is usually the case, what strategies can be applied to mitigate this issue?