• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.67-75
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• 1997

In this paper, A method is proposed to analyze the noise figure for some parameters based on IS-98 recommendation when a RF receiver for the CDMA mobile station is designed. Simulation results show that the maximum noise figure of the receiver, which fits in the receiving sensitivity of the mobile station, is about 11dB, and this value is smaller about 3dB than the existing specification. In addition, we have tested the relationship between frame error rate and Eb/Nt of traffic channel for each coding rate. according to the speed changes of the traffic channel. In order to prove IMD to be most important variable in LNA we have tested IMD spurious in case that LNA always turns on and in case that LNA turns ON/OFF.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.520-525
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• 2009

The conventional studies about an adaptive beamformer assumed that the interference signals are stationary, so they used time-average of signals or Least Mean Squares. However, these methods showed low performance of canceling the non-stationary interferences. In this paper, the MAFF-RLS algorithm is developed in order to cancel non-stationary interferences, and the GSC structure using this algorithm is proposed. Furthermore, the performance of the MAFF-RLS beamformer is verified by simulation using MATLAB. This simulation results show the performance of the proposed beamformer is better than that of the SMI and the conventional RLS beamformer.

• Journal of Advanced Navigation Technology
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• v.24 no.5
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• pp.393-407
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• 2020

In order to reduce the threat by illegal unmanned aerial vehicles, a tracking system based on sound was implemented. There are three main points to the drone acoustic tracking method. First, it scans the space through variable beam formation to find a sound source and records the sound using a microphone array. Second, it classifies it into a hidden Markov model (HMM) to find out whether the sound source exists or not, and finally, the sound source is In the case of a drone, a sound source recorded and stored as a tracking reference signal based on an adaptive beam pattern is used. The simulation was performed in both the ideal condition without background noise and interference sound and the non-ideal condition with background noise and interference sound, and evaluated the tracking performance of illegal drones. The drone tracking system designed the criteria for determining the presence or absence of a drone according to the improvement of the search distance performance according to the microphone array performance and the degree of sound pattern matching, and reflected in the design of the speech reading circuit.

• Journal of Digital Convergence
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• v.11 no.10
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• pp.367-372
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• 2013

In Speech recognition system, there is a problem with phoneme in the model training and it cause a stored mode regeneration process which come into being appear time and more costs. In this paper, we propose the methode of likely phoneme model schema using decision tree clustering. Proposed system has a robust and correct sound model which system apply the decision tree clustering methode form generate model, therefore this system reduce the regeneration process and provide a retrieve the phoneme unit in probability model. Also, this proposed system provide a additional likely phoneme model and configured robust correct sound model. System performance as a result of represent vocabulary dependence recognition rate of 98.3%, vocabulary independence recognition rate of 98.4%.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.774-780
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• 2009

This paper proposes a transmitted power saving method for underwater acoustic sensors considering the acoustic wave propagation characteristic that propagation loss increases more rapidly in higher frequency band. In the proposed scheme, sensor nodes are divided into a few groups based on the distance between sink node and the sensor node, and each group uses its own frequency band. The node group with longer distance uses lower frequency and the node group with shorter distance uses higher frequency. By means of such a distance-dependent frequency allocation, all sensor nodes are able to maintain a certain target signal-to-noise ratio (SNR), but also save transmitted power. In addition, the optimum size of node group is obtained, and also a frequency allocation algorithm is proposed accordingly. Numerical results show that the proposed scheme saves transmitted power by more 10 dB comparing non-grouping methods.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.270-277
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• 2010

In this paper, we proposed an algorithm that improves tone quality of noisy audio signals by using ICA(Independent Component Analysis) algorithm and perceptual filters. Many algorithms have been proposed to eliminate the noise from the audio signals, such as spectral subtraction method, perceptual filter, etc. The perceptual filter uses a noise that is acquired from silent ranges in the input signal. In this case, the improvement rate of tone quality decreases if the noise energy is changed by the environmental variation in a signal frame. But the proposed method estimates a noise that is changed at each frame using ICA algorithm. The estimated noise is applied to perceptual filter. To show the performance of the proposed algorithm, several tests are performed to various input signals. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR), noise-to-mask ratio (NMR) and Degradation Category Rating (DCR) test.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.303-313
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• 2010

This paper describes an acoustic communication receiver structure, which is designed for QPSK (Quadrature Phase Shift Keying) signal with 25 kHz carrier frequency and 5 kHz symbol rate, and takes samples from received signal at 100 kHz sampling rate. Based on the described receiver structure, optimum design parameters, such as number of taps of FF (Feed-Forward) and FB (Feed-Back) filters and forgetting factor of RLS (Recursive Least-Square) algorithm, of joint equalizer are determined to minimize the BER (Bit Error Rate) performance of the joint equalizer output symbols when the acquisition data in shallow water using implemented acoustic transducers is decimated at a rate of 2:1 and then enforced to the input of receiver. The transmission distances are 1.4 km, 2.9 km, and 4.7 km. Analysis results show that the optimum number of taps of FF and FB filters are different according to the distance between source and destination, but the optimum or near optimum value of forgetting factor is 0.997. Therefore, we can reach a conclusion that the proper receiver structure could change the number of taps of FF and FB filters with the fixed forgetting factor 0.997 according to the transmission distance. Another analysis result is that there are an acceptable performance degradation when the 16-tap-length simple filter is used as a low-pass filter of receiver instead of 161-tap-length matched filter.

• Journal of the Korea Computer Industry Society
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• v.2 no.8
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• pp.1063-1068
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• 2001

In this paper, we designed a high resolution six channel DSB modulator of Acousto-Optic effect generator make use of DDS technology. Also, configured seamless connection for other instruments to use IEEE-488 bus interface. We programmed the device driver for DDS and DAC control by 80C51 assembler language. And, high resolution 6 channel DSB modulator has improved the important characteristics of that the frequency tuning range, the resolution, the switching time. This DSB modulator system can use high precision frequency synthesizer for instruments.

• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.6
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• pp.58-61
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• 1983

This is a Loudness Control Circuit in an audio amplifier controlled by feedback type volume control variable resistors. This circuit consists of Bridged Twin T network and a ordinary variable resistor. The variably resistor acts not only as a volume control by varying feedback qupntity, but also as Loudness Control through the characteristics variation by Sound Level. This new Loudness Control Circuit showed ideal compensation characteristics that agree computer simulation and measured datas.

• The Journal of the Acoustical Society of Korea
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• v.26 no.1E
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• pp.27-32
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• 2007

Hydraulic actuator and electro-magnetic liner actuator have been used as typical active vibration control methods. However these methods have many kinds of disadvantages such as causing space limit, difficult maintenance, complicate structures, etc. The purpose of this paper was to study on the possibility of active vibration control using piezoelectric transducer. Piezoelectric transducer generated a vibration and GIC (General Impedance Converter) amplifier was adopted to give adjustable vibration signal to transducer and high amplitude of vibration. Resonance frequency of piezoelectric transducer was controlled by GIC amplifier and higher amplitude of vibration was achieved. Finally active vibration control using piezoelectric transducer was performed.