• Journal of Biomedical Engineering Research
• /
• v.27 no.1
• /
• pp.22-29
• /
• 2006

This paper proposes an efficient array beamforming method using spatial matched filtering for ultrasound imaging. In the proposed method, ultrasound waves are transmitted from an array subaperture with fixed transmit focus as in conventional array imaging. At receive, radio frequency (RF) echo signals from each receive channel are passed through a spatial matched filter that is constructed based on the system transmit-receive spatial impulse response. The filtered echo signals are then summed. The filter remaps and spatially registers the acoustic energy from each element so that the pulse-echo impulse response of the summed output is focused with acceptably low side lobes. Analytical beam pattern analysis and simulation results using a linear array show that the proposed spatial filtering method can provide more improved spatial resolution and contrast-to-noise ratio (CNR) compared with conventional dynamic receive focusing (DRF) method by implementing two-way dynamically focused beam pattern throughout the field.

• The Journal of the Acoustical Society of Korea
• /
• v.23 no.6
• /
• pp.437-445
• /
• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.6
• /
• pp.373-383
• /
• 2012

The performance of underwater acoustic communication system is sensitive to the Doppler shift and ISI(Inter-Symbol Interference). Therefore, the simulation algorithm needs to consider time-spread due to multipath arrivals which cause the ISI, and time-varying Doppler shift along with moving source and receiver. For this purpose, VirTEX(Virtual Time series EXperiment) based on Ray model has been developed. In this paper, VirTEX is used to compare the characteristics of ocean waveguide from the experimental data and illustrate the performance. The CIR(Channel Impulse Response) that characterizes the multipath arrivals with representative time-spread due to multipath arrivals is compared between numerically simulated and experimental probe signal. Also, the communication performance analysis for BER(Bit Error Rate) is compared between numerically simulated and experimental data signal. As a result, VirTEX can be useful as a simulation tool for evaluating the performance of underwater acoustic communication system.

• The Transactions of The Korean Institute of Electrical Engineers
• /
• v.56 no.3
• /
• pp.629-631
• /
• 2007

In hands-free full-duplex communication systems, acoustic signals picked up by the microphones can be mixed with echo signals as well as noises, which may result in poor performance of the corresponding communication system. Also, the system performance may decrease further if the reverberation occurs since it is harder to estimate the impulse response of the demixing system. For blind source separation (BSS) in such cases, a time-frequency masking approach can be employed to separate undesired echo signals and noises, but, permutation ambiguities also should be solved for the echo cancellation. In this paper, we propose a new acoustic echo cancellation (AEC) approach utilizing the time-frequency masking and higher-order statistics, whereby a desired signal selection, based on coherence and third-order statistics (i.e., kurtosis), is introduced along with output signal normalization. Simulation results demonstrate that the proposed approach yields better echo and noise cancellation performances than the conventional AEC approaches.

• Journal of Sensor Science and Technology
• /
• v.19 no.6
• /
• pp.490-496
• /
• 2010

In this study, in order to overcome the weakness of acoustic field analysis which is generally used for ultrasonic probe performance evaluation, automatic acoustic field measurement system and evaluation parameters were proposed. The comparisons between acoustic field simulation and measured acoustic distribution data of normal and abnormal channels were conducted to evaluate the availability of proposed system and evaluation parameters. First, the impulse response characteristic of sample probe was investigated to classify the normal elements and abnormal elements. And then, normal channels and abnormal channels with abnormal element were chosen. The suggested 12 evaluation parameters were calculated using the acoustic fields of these channels. The availability of proposed automatic acoustic field measurement system and evaluation parameters was confirmed. And the performance evaluation of ultrasonic probe using acoustic field analysis could be easier and faster.

• The Journal of the Acoustical Society of Korea
• /
• v.9 no.2
• /
• pp.42-52
• /
• 1990

A method of measuring impulse response of acoustic system, two kinds of cross-correlation methods (the direct correlation method and the M-sequence modulation correlation method) were applied. According to the direct correlation method, by using stationary random noise source and by calculating the cross-correlation function between the sourece and the output signal, equivalent impulse response can be obtained not being influenced by the back ground noises. By applying this method, the measurement of echo-time patterns in rooms and oblique incident sound absorption characteristics of sound absorbing materials was carried out. In the case of the M-sequence modulation correlation method which was contrived by Aoshima, an intermittent random nosie modulated by M-sequence signal is used as the source signal, and the cross-correlation function between the M-sequence signal and the squared output signal is calculated. According to this method, equivalent energy impulse response (squared impulse response) of te propagation system can be obtained without being influenced by the back ground noises and the air fluctuation caused by wind. As the applcaition of this measuring method to the architectural acoustics, the meaurements of echo-time patterns, reverberation decays and sound pressure lev디 distributions in rooms and sound insulation efficiencies in buildings were carried out. From these experimental studies, it has been found that this M-sequence modulation correlation method is markedly useful especially for the field masurement of sound insulation under high back ground noise condition.

• Journal of Korea Multimedia Society
• /
• v.13 no.7
• /
• pp.960-972
• /
• 2010

In this paper, we present a software for placing instruments on stage based on the acoustic properties of the concert hall. In order to simulate the changes in sound depending on the positions of the instruments, we incorporated the idea of location-based reverberation effect which can be realized through the convolution of instrument sounds with the impulse responses from the respective instrument positions. And we developed a software with a real-time convolution engine which enables the user to conveniently simulate the resulting sound of various instrument placements. The software was tested with the impulse response data measured at two concert halls of the National Center for Korean Traditional Performing Arts and Korean traditional instrument sounds. Results of these experiments show that simulated reverberation effects properly represent the spatial placement of instruments on stage.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.39 no.5
• /
• pp.556-563
• /
• 2002

In this paper, a novel method for designing an M-channel, causal, stable IIR cosine-modulated filter bank (CMFB) with near PR property is proposed. The IIR prototype filter is designed with a simple constraint using lattice stucture with 1st order allpass filter components. The IIR prototype filter which is designed by the proposed method has higher stopband attenuation and sharper roll-off characteristic than the one which is designed by the previously proposed method with similar complexity. The proposed M-channel IIR CMFB which is designed from this IIR prototype filter is applied to subband acoustic echo canceller (AEC). We obtained about 15dB higher ERLE using this subband AEC than when M-channel FIR subband AEC with similar complexity.

• The Journal of the Acoustical Society of Korea
• /
• v.38 no.5
• /
• pp.494-510
• /
• 2019

Underwater acoustic communication channel parameters consist of impulse response, delay spreading, scattering function, coherence bandwidth, frequency selective fading, coherence time and time variant magnitude fading statistics on which communication system modem and channel coding are designed. These parameters are influenced by sound velocity profile, platform motion and sea surface roughness in given acoustical oceanography condition. In this paper, channel model based on phasor, channel simulator, measurement and analysis method of channel parameters are given in a fixed source-to-receiver system and the parameters are analyzed using shallow water experimental data. For two different source-to-receiver ranges of 300 m and 600 m, the parameters are characterized by three multipaths such as a direct, a surface reflection path with time variant scattering and a bottom reflection path. The results present a channel modelling method of a fixed source source-to-receiver system, channel parameters measurement and analysis methods and a system design and performance assessment method in shallow water.

• Journal of Sensor Science and Technology
• /
• v.18 no.1
• /
• pp.33-41
• /
• 2009

A multilevel acoustic Fresnel lens (MAFL) for the ultrasonic imaging transducer of which center frequency is approximately 5.MHz was newly designed and fabricated. The phase level of the lens was 64, and the focal length and the aperture width were 30.mm and 11.mm, respectively. The characteristics of impulse response, acoustic field and imaging performance of the transducer attached the lens were compared with the transducer attached a conventional refraction type acoustic lens (RAL). The results show that the center frequency, the loop sensitivity, and the focal depth of the MAFL transducer were higher or larger than those of the RAL transducer by approximately 0.2.MHz, 1.4.dB, and 2.mm, respectively. Consequently, it was shown that the brighter acoustic images with higher lateral resolution and the increased imaging performance for deep targets can be obtained by using the MAFL transducer.