Development of a speech enhancement dual-microphone noise reduction system utilizing TMS320C6713

Chao Min Wu, Zeng Fong Chen, Ting An Liu

研究成果: 書貢獻/報告類型會議論文篇章同行評審

2 引文 斯高帕斯(Scopus)

摘要

The purpose of this research was to add a speech enhancement process that could further improve speech intelligibility and the performance of automatic scene classification and automatching noise reduction system after the application of the adaptive directional microphone strategy. The speech enhancement system is divided into two parts: one is the noise-estimation strategy and the other the speech-estimation function. Noise-estimation algorithms used in the research are: minimum statistics (MS), minima-controlled recursive averaging (MCRA), improved(IMCRA), MCRA-Loizou (MCRA-L), constrained variance spectral smoothing (CVS), forward-backward MCRA(MCRA-FB);speech-estimation function: Wiener filter, maximum-likelihood (ML), log-spectral amplitude (LSA), maximum a posteriori amplitude (MAPA). In this research, The MATLAB program was first used to simulate the speech enhancement system to evaluate the quality of output speech signal under different signal-to-noise ratio (SNR) conditions, and then to select the best combination of the speech enhancement system. Finally, the selected speech enhancement system was implemented with automatic scene classification and auto-matching noise reduction system in TMS320C6713 DSP Starter Kit (Texas Instruments, Dallas, Texas, USA), and compared with the output signal in the original noise reduction system. To show the performance of the selected speech enhancement system, the objective perceptual evaluation of speech quality (PESQ) approach was further used to estimate the quality of speech with the SNR range from +30 and -30dB. Our simulated results showed that the PESQ score was increased by 0.45 when the speech enhancement CVS with MAPA was used for the input signal with 30 dB SNR and by 0.65 for 10 dB SNR. However, for the hardware implementation, only the speech enhancement MCRA with MAPA was used for real-time processing. The PESQ score was increased by 0.27 for the input signal with 0 dB SNR. With the speech enhancement system, our overall hardware implementation could effectively reduce speech distortion and improve speech intelligibility.

原文???core.languages.en_GB???
主出版物標題ICSV 2016 - 23rd International Congress on Sound and Vibration
主出版物子標題From Ancient to Modern Acoustics
發行者International Institute of Acoustics and Vibrations
ISBN(電子)9789609922623
出版狀態已出版 - 2016
事件23rd International Congress on Sound and Vibration, ICSV 2016 - Athens, Greece
持續時間: 10 7月 201614 7月 2016

出版系列

名字ICSV 2016 - 23rd International Congress on Sound and Vibration: From Ancient to Modern Acoustics

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???event.eventtypes.event.conference???23rd International Congress on Sound and Vibration, ICSV 2016
國家/地區Greece
城市Athens
期間10/07/1614/07/16

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