• Journal of the Institute of Convergence Signal Processing
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• v.4 no.4
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• pp.23-29
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• 2003

In this paper, we propose an idea for the improved 3-D sound system using conventional stereo headphones to obtain a better sound diffusion from the mono-sound recorded at an anechoic chamber. We use the HRTF(Head Related Transfer Function) for the sound localization and the wavelet filter bank with time delay for the sound diffusion. And we test the modified HRTF with the various sampling rate. We investigate the effects of the 3-D sound depending on the length of time delay at lowest frequency band. Also the correlation coefficient of the signals between the left channel and the right channel is measured to identify the sound diffusion. At last we obtain the diffusion sound using Cool Edit for reverberation.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.81-86
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• 2014

This paper presents a study on the effectiveness of using the spectral and the temporal octave-band order statistics for musical genre classification. In order to represent the relative disposition of the harmonic and non-harmonic components, we utilize the octave-band order statistics of power spectral distribution. Experiments on the widely used two music datasets were performed; the results show that the octave-band order statistics improve genre classification accuracy by 2.61 % for one dataset and 8.9 % for another dataset compared with the mel-frequency cepstral coefficients and the octave-band spectral contrast. Experimental results show that the octave-band order statistics are promising for musical genre classification.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.11
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• pp.184-193
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• 1998

In this paper, a real-time acoustic echo canceller is implemented using only PC's CPU without extra help from a DSP chip. The adaptive digital filter is designed efficiently so that it can be implemented in real-time and has a proper cancellation performance. It is proposed that a new double talk detector consumes a small computational complexity and guarantees the fast detection and robust operation. The real-time acoustic echo canceller consists of the full-duplex sound card and 166 MHz Pentium PC, and requires less than 10% CPU time.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.41-44
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• 1999

본 논문에서는 PC 응용을 위한 고음질의 한국어 text-to-speech(TTS) 합성 시스템을 개발하였다. 개발된 시스템의 합성방식으로는 음의 고저 조절, 인접음 사이의 연결 처리 및 음색제어 등에서 기존의 PSOLA 방식에 비해 장점을 가지는 정현파 모델 기반의 방식을 채택하였고, 자연스러운 운율 모델링을 위하여 통계적 기법중의 하나인 Classification and regression tree(CART) 방법을 사용하였다. 또한 음소 경계의 불연속성 문제를 줄이기 위한 합성단위로 초성-중성 및 종성 단위를 사용하였고, 다양한 음색표현이 가능하도록 음색제어 기능을 갖추었다. 그리고, 표준 Speech Application Program Interface(SAPI)를 준용한 TTS engine 형태로 구현함으로써 PC 상에서의 응용 프로그램 개발 편의성을 높였다. 합성음의 청취평가 결과 음질의 우수성 및 음색제어 기능의 유효성을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.10 no.2
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• pp.23-28
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• 1991

This paper is a study on isolated word recognition by independent speaker, we propose DTW speech recognition system by modified dynamic averaging method as reference pattern. 57 city names are selected as recognition vocabulary and 2th LPC cepstrum coefficients are used as the feature parameter. In this paper, besides recognition experiment using modified dynamic averaging method as reference pattern, we perform recognition experiments using causal method, dynamic averaging method, linear averaging method and clustering method with the same data in the same conditions for comparison with it. Through the experiment result, it is proved that recogntion rate by DTW using modified dynamic averaging method is the best as 97.6 percent.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.40-46
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• 2004

The equivalent source method (ESM) uses two groups of equivalent source positions. One group includes the first order images of the sound source inside the enclosure. The positions of the other group are usually on a spherical surface some distance outside the enclosure. A proper selection of the positions for the far field sources could greatly improve the performance of the modeling accuracy and reduce the number of the sources to achieve the required accuracy. This study uses optimally distributed far field source positions on the surface of enlarged version of the rectangular enclosure instead of using typical spherical distribution. Simulations using various sizes of the box shaped distribution are executed and optimal positions are searched using an optimization technique from the larger number of candidate positions. The results of using these far field source positions are compared and analyzed.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3
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• pp.79-87
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• 1999

In this paper, we reconstruct an algorithm of sound image control for real-time processing and implement a real-time system using digital signal processing board based on TMS320C40. The performance of real-time sound image control system was evaluated by a listening test. The results of the test showed that localized sound image can be perceived by headphone and speaker, and result of test by headphone was better than that of speaker. In the results of test of elevation perception, result of perception between left and right were better than those for differentiating between front and back, and the moving sound image was better than the fixed sound image.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.576-584
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• 2002

This paper proposes a new denosing method using wavelet packet, to reject unknown external noise and white gaussian ambient noise for measuring the transient noise which is one of the important elements for ship classification. The previous denosing method applied the same wavelet threshold at each node of multi-single sensors for rejecting white noise is not adequate in the underwater environment existing lots of external noises. The proposed algorithm of this paper applies a modified soft-threshold to each node according to the discriminated threshold so as to reject unknown external noise and white gaussian ambient noise. It is verified by numerical simulation that the SNR is increased more than 25㏈. And the simulation results are confirmed through sea-trial using multi-single sensors.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.5-11
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• 1995

In this paper, we design the new high-speed seek algorithm which can access the data of disk (in 3.5-inch optical disk drive). When seek the target track on optical disk in high-speed using the ref-T. E(Reference Tracking Error) and sample pulse signal, it's always possible high-speed seek without re-check and compensation for error by the ref_T. E and sample pulse signal. The performance test result is represented as follow ; it is satify the average seek time of 28mses.

• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.23-32
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• 1989

The calibration of 1 inch standard condenser microphone is done by the reciprocity calibration method. There are two kinds of reciprocity calibration, free-field calibration and pressure calibration. The pressure sensitivities of the three 1 inch condenser microphones are determined by pressure calibration. The accuracy of the pressure sensitivity of the microphone depends on the accuracies of the voltage and dimension measurements as well as the various corrections for the coupler. If the individual accuracies for the measurements and corrections are achieved, it is estimated that the over-all accuracy is approximately 0.05dB at low and middle frequencies decreasing to about 0.ldB over 10kHz.