Audio Pre-Processing and Beamforming Implementation on Embedded Systems
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Published:2024-07-15
Issue:14
Volume:13
Page:2784
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ISSN:2079-9292
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Container-title:Electronics
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language:en
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Short-container-title:Electronics
Author:
Wang Jian-Hong1ORCID, Le Phuong Thi2ORCID, Kuo Shih-Jung3, Tai Tzu-Chiang4, Li Kuo-Chen5ORCID, Chen Shih-Lun6, Wang Ze-Yu1, Pham Tuan7ORCID, Li Yung-Hui8ORCID, Wang Jia-Ching3
Affiliation:
1. School of Computer Science and Technology, Shandong University of Technology, Zibo 255000, China 2. Department of Computer Science and Information Engineering, Fu Jen Catholic University, New Taipei City 24205, Taiwan 3. Department of Computer Science and Information Engineering, National Central University, Taoyuan City 320317, Taiwan 4. Department of Computer Science and Information Engineering, Providence University, Taichung 43301, Taiwan 5. Department of Information Management, Chung Yuan Christian University, Taoyuan City 320317, Taiwan 6. Department of Electronic Engineering, Chung Yuan Christian University, Taoyuan City 320317, Taiwan 7. Faculty of Digital Technology, University of Technology and Education, Da Nang 550000, Vietnam 8. AI Research Center, Hon Hai Research Institute, New Taipei City 236, Taiwan
Abstract
Since the invention of the microphone by Barina in 1876, there have been numerous applications of audio processing, such as phonographs, broadcasting stations, and public address systems, which merely capture and amplify sound and play it back. Nowadays, audio processing involves analysis and noise-filtering techniques. There are various methods for noise filtering, each employing unique algorithms, but they all require two or more microphones for signal processing and analysis. For instance, on mobile phones, two microphones located in different positions are utilized for active noise cancellation (one for primary audio capture and the other for capturing ambient noise). However, a drawback is that when the sound source is distant, it may lead to poor audio capture. To capture sound from distant sources, alternative methods, like blind signal separation and beamforming, are necessary. This paper proposes employing a beamforming algorithm with two microphones to enhance speech and implementing this algorithm on an embedded system. However, prior to beamforming, it is imperative to accurately detect the direction of the sound source to process and analyze the audio from that direction.
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