This paper shows one and two dimensional active linear modeling of cochlear biomechanics using Hspice. The advantage of the Hspice modeling is that the cochlear biomechanics may be implemented into an analog Ie chip. This paper explains in detail how to transform the physical cochlear biomechanics to the electrical circuit model and how to represent the circuit in Hspice code. There are some circuit design rules to make the Hspice code to be executed properly.

The acoustical sources of intake and exhaust systems in fluid machines are often characterized by the source impedance and strength using linear frequency-domain modeling. In the case of the sources which are nonlinear and time-variant, however, the source parameters were sometimes incorrectly obtained. In this paper, the source model and direct measurement technique are modified in order to evaluate the effect due to nonlinear and periodically time-varying source character as well as the linear property of the reflectivity of in-duct fluid machine source. With a priori known kinematical information of the source, the types of nonlinear time-variant terms can be presumed by a simple physical model, in which there is practically no restriction on the form of the model. The concept of source impedance can be extendable by introducing the linear frequency response function for each nonlinear or time-variant input. Extending the conventional method and adapting the reverse MISO technique, it is possible to develop a direct method that can deal with the nonlinear time-variant source parameters. The proposed direct method has a novel feature that there is no restriction on the probability or spectral natures of the excited sound pressure data. The present method is verified by the simulated measurements for simplified fluid machines. It is thought that the proposed method would be useful in predicting the insertion loss or the radiated sound level from intake or exhaust systems.

Enhanced approach using computational and experimental method is proposed and performed to describe very well the behavior of loudspeaker than conventional method. Proposed procedure is composed of four parts. First, Thiele-Small parameters for test loudspeaker are identified by an electrical impedance method like as a delta mass method. Second part includes the processes to measure physical properties. Physical data like masses and thicknesses of loudspeaker's components are measured by an electrical precision scale and a digital vernier caliper. Third, the identified Thiele-Small parameters are proposed to be used as load boundary conditions for vibration analysis instead of electromagnetic circuit analysis to get a driving force upon bobbin part. Also, these parameters and physical data are used to modify physical properties required for computation to accommodate simulated sound pressure level with measured one for loudspeaker enclosure system. These data like as Young's modulus and thickness for a diaphragm are required for vibration analysis of loudspeaker but not measured accurately. Finally, it was investigated that simulated sound pressure level with full acoustic modeling including an acoustic port for test loudspeaker agreed with experimental result very well in the midrange frequency band(from 100 Hz to 2,000 Hz). In addition, several design parametric study is performed to grasp acoustical behaviors of loudspeaker system due to variations of diaphragm thicknesses and shapes of dust cap.

To investigate the equalizer performance in underwater acoustic communication m the presence of intersymbol interference (ISI) due to multipath, computer simulations are carried out in discrete multipath shallow water channels for three different horizontal ranges. For the purpose of computation simplicity, least mean square (LMS) algorithm is adopted both in linear equalizer and nonlinear equalizer, decision feedback equalizer (DFE) to cancel out ISI effects. Binary phase shift keying (BPSK) signals have been transmitted with high data rate of 2000bps through the use of equalization technique. The results demonstrate that equalization is an efficient way to achieve high transmission data rate in the shallow water channel.

In this paper, we present a phase coherent all-digital transceiver for underwater acoustic communication, which allows the system to reduce complexity and increase robustness in time variant underwater environments. It is designed in the digital domain except for transducers and amplifiers and implemented by using a multiple digital signal processors (DSPs) system. For phase coherent reception, conventional systems employed phase-locked loop (PLL) and delay-locked loop (DLL) for synchronization, but this paper suggests a frame synchronization scheme based on the quadrature receiver structure without using phase information. We show experimental results in the underwater anechoic basin at MOERI. The results show that the adaptive equalizer compensates frame synchronization error and the correction capability is dependent on the length of equalizer.

Two twin microphones may produce particular patterns of binaural directivity by time delays between twin microphones. The boundary element method (BEM) was used for the simulation of the sound pressure field around the head model in order to quantify the acoustic head effect. The sound pressure onto the microphone was calculated by the BEM to an incident sound pressure. Then a planar directivity pattern was formed by four sound pressure signals from four microphones. The optimal binaural directivity pattern may be achieved by adjusting time delays at each frequency while maintaining the forward beam pattern is relatively bigger than the backward beam pattern.