Wav2Vec 2 audio processing
I am currently usingfacebook/wav2vec2-basemodel for an audio classification task. My code is based on the official HF Audio Classificationtutorial.In this tutorial, audios are sampled using asample_rateof 16.000.  This means that 1 second of audio results in an array of length 16.000. Also,Wav2Vec2maximum 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?