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Physiological acoustics is a very hot topic in modern acoustic research, which is to study the hearing mechanism and the utterance of both humans and animals. It could be divided into two main aspects: physical acoustics of the ear and physiological acoustics. In physiological acoustics, there are some common research methods, such as objective experimental testing, subjective feelings evaluation survey statistical method, building the physical acoustic model and numerical simulation methods, etc. The authors are researching the accurate mathematical model of equal loudness curves, critical band and masking effects, by applying the holographic concept with several biological factors which are required to build a standard model. The cochlear emission information should be extracted from wavelet analysis method and two hearing protection technologies are being developed by band shielding.

In this paper, we propose an active hearing protection system for workers in an extreme noise industrial environment. For the active hearing protection, we utilize ANC (Active and speech enhancement technology. Active hearing protection provides face to face communication that voice is passed through and the noise can be blocked. For this object we propose a method of estimating a secondary path of ANC accordance with user's hearing protection wearing characteristics and a speech enhancement algorithm to improve speech intelligibility with the ANC and the De-Noised speech signal. An adaptive RMC (Residual Music Canceler) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. We obtained results that the secondary path can be accurately estimated and high quality of music sound can be consistently obtained in a time-varying secondary path situation. Most of the ANC system use the LMS algorithm to design a secondary path filter. However the convergence time of the filter is prolonged in the conventional method because the initial coefficient values of the secondary path filter is set to 0. To solve this problem, in this paper we propose a method of shortening the convergence time of the filter by setting the coefficient values determined in advance as an initial value of the secondary pass filter. With the proposed method the error is decreased from the initial stage of the secondary path filter, whereby the convergence of the ANC system is increased. In the experiments by using a commercially available headset it was confirmed that the proposed method is faster than the conventional method for convergence of ANC system. By applying the ANC and the De-Noising system in real-time DSP (Digital Signal Processor), we have developed an active hearing protection.