• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.36-42
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• 1996

In this paper, we have proposed the robust estimation of HMM parameters which is based on CDDN(codeword dependent distance normalization)and codeword weighting by distance. The proposed method has used a distance normalization based on the characteristics of a codeword dependent distribution and have computed fuzzy contributions of codeword to a input vector with a fuzzy objective function. From experimental results, we have shown the effectiveness of the proposed method in that the correction rate of the proposed method is improved 4.5% over the conventional FVQ based method. Especially, the application of distance weighting to smoothing of output probability is improved the performance of 2.5% compared to distance based codeword weighting.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.374-380
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• 2010

It is known that the problem of FIR filtering with noisy input and output data can be solved by a total least squares (TLS) estimation. It is also known that the performance of the TLS estimation is very sensitive to the ratio between the variances of the input and output noises. In this paper, we propose a convex combination algorithm between the ordinary recursive LS based TLS (RTLS) and the ordinary recursive LS (RLS). This combined algorithm is robust to the noise variance ratio and has almost the same complexity as the RTLS. Simulation results show that the proposed algorithm performs near TLS in noise variance ratio ${\gamma}{\approx}1$ and that it outperforms TLS and LS in the rage of 2 < $\gamma$ < 20. Consequently, the practical workability of the TLS method applied to noisy data has been significantly broadened.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1
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• pp.1-9
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• 2009

This paper deals with the realization of multi-channel live karaoke. In this study, 6-channel MP3 decoding and tempo/key scaling was operated in real time by using the TMS320C6713 DSP, which is 32 bit floating-point DSP made by TI Co. The 6 channel consists of front L/R instrument, rear L/R instrument, melody, and woofer. In case of the 4 channel, rear L/R instrument can be replaced with drum L/R channel. And the final output data is generated as adjusted to a 5.1 channel speaker. The SOLA algorithm was applied for tempo scaling, and key scaling was done with interpolation and decimation in the time domain. Drum channel was excluded in key scaling by separating instruments into drums and non-drums, and in processing SOLA, high-quality tempo scaling was made possible by differentiating SOLA frame size, which was optimized for real-time process. The use of 6 channels allows the composition of various channels, and the multi-channel audio system of this study can be effectively applied at any place where live music is needed.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.1-7
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• 2008

The paper is to solve the problem of existing sound system, which has difficulties of system organization and the increase of additional install cost and unfriendly interior. To solve the existing system, we drew the new sound system based on PLC and studied it. A transmitter and a receiver were designed using the PLC chip INT5500CS. Sound system was configured with a CD player that sound signals are sent from the transmitter and a speaker connected to the receiver. For analysis of characteristics of this system, a USBPre external sound card and Smaart Live 5 which is a PC-based sound measuring program were added. As a result of our experiment, the measured signal level is $2{\sim}3$[dB] lower than reference signal, latency is 16.69[ms] and the specific character of coherency is bad in high frequency band. Otherwise, this system transmits and receives signals over 90[%] in good condition as a result of measuring pink noise, frequency(1kHz), and phase, magnitude. In view of the result so far achieved, the system designed our team has excellent performance, it resolves defect of existing audio signal transmition system.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.408-416
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• 2021

In the ultrasonic dispersion, in order to avoid direct contact of the radiation surface of ultrasonic transducers with a liquid sample, the liquid sample is separated by a glass container and it receives ultrasonic energy through an acoustic medium. The transmission efficiency of the ultrasonic energy in the multi-layered ultrasonic system is an important factor. In this study, we suggested a method that can improve the ultrasonic energy transfer efficiency by using a propylene glycol solution as a liquid matching layer in the multi-layered acoustic system. In this method, a propylene glycol solution was filled between the Langevin-type ultrasonic transducer and the luminol solution and the sonoluminescence phenomena in the luminol solution, which is caused by nonlinear effect of high power ultrasound radiated from the transducer, was examined by using a Photo Multiplier Tube (PMT). The transmission efficiency depending on the concentration of propylene glycol solution was observed, and we can see that as the concentration of the propylene glycol solution increased, the matching effect increased while the acoustic attenuation increased. It was confirmed that there is an optimal concentration compromised these two conflicting conditions, and the optimum concentration of the propylene glycol solution was determined experimentally.

• The Journal of the Acoustical Society of Korea
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• v.23 no.4
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• pp.316-323
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• 2004

Recent trend of installation of highly efficient and high Power engine in vehicles has increased complaints about exterior noise being made while travelling. To examine the sources of outer noise of vehicles controlled by regulations. using lead wrapping method. Pass-by noise tests have been conducted as opening each defined source one by one. The sources of outer noise have been found and the rate of noise contribution has been produced. The results of the tests have been applied to put noise-reducing objects in the order, and counter plans effective to reduce noise have been devised.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.161-164
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• 1999

본 논문에서는 표적 추적 시스템에서 센서 어레이에 입사되는 표적신호에 대한 센서 출력 신호간의 지연시간 추정(TDE:Time Delay Estimation)을 위해 백색상호 상관법(CSP:Cross-Power Spectrum Phase Analysis)을 이용한다. 그러나 음파의 다중경로 전달특성 및 배경잡음의 영향으로 인해 CSP계수는 많은 클러터(Clutter)를 포함하게 되고 결국 방위 추정 오차의 요인이 된다. 따라서 센서 어레이 중심좌표를 기준으로 대칭 배열된 센서쌍(Pair)에 대한 CSP계수를 동기가산 하여 실제 표적 방향 이외의 방향정보를 제거하는 방법을 제안한다. 시간에 따라 각도를 변침하는 표적에 대한 표적기동분석 (BOTMA:Bearings Only Target Motion Analysis)을 위해 매 관측시간마다 동기가산을 행한 CSP결과를 누적하여 방위각 궤적을 형성하였을 때 시간 Window에 따라 약간의 차이는 있지만 약 10dB의 궤적 추적 성능을 확인하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.105-108
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• 1999

TTS(Text-To-Speech) 시스템 합성음성의 자연감을 개선하기 위해 하나의 언어에 대해 존재하는 운율 법칙을 정확히 구현해야 한다. 존재하는 운율 법칙을 추출하기 위해서는 방대한 분량의 언어 자료 구축이 필요하다. 그러나 이 방법은 존재하는 운율 현상이 포함된 언어자료에 대해 완벽한 운율을 파악할 수 없으므로 합성음성의 질을 좋게 할 수 없다. 본 논문은 한국어 음성의 운율을 학습하기 위해 2개의 인공 신경망을 제안한다. 하나의 신경망으로 문장의 각 음소에 대한 피치 변화를 학습시키는 것이며, 다른 하나는 에너지 변화를 학습하도록 하였다. 신경망은 BP 신경망을 이용하며 11개의 음소를 나타내기 위해 11개의 입력과, 중간 음소의 피치와 에너지 변화곡선을 근사하는 다항식 계수를 출력하도록 하였다. 신경망시스템의 학습과 평가에 앞서, 음성학적 균형잡힌 고립단어를 기반으로 의미있는 문장을 구성하였다. 문장을 남자 화자로 하여금 읽게 하고 녹음하여 음성 DB를 구축하였다. 음성 DB에 대해 각 음소의 운율 정보를 수집하여 신경망에 맞는 목표 패턴과 훈련 패턴을 작성하였다. 이 목표 패턴은 회귀분석을 통한 추세선을 이용해 피치와 에너지에 대한 2차 다항식계수로 구성하였다. 본 논문은 목표패턴에 맞는 신경망을 학습시켜 좋은 결과를 얻었다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.339-342
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• 2000

본 논문에서는 DSP와 RISC 마이크로 콘트롤러의 결합으로 구성된 이중 코어 DSP를 이용하여 휴대장치에 적합한 저전력 MPEC-2 계층-III 복호화기의 구조를 제안하고 실시간 시스템을 구현하였다. 제안된 시스템은 디지털 오디오 데이터 처리부와 시스템 제어 정보처리부로 나누어 병렬처리가 가능한 구조이다. 디지털 오디오데이터 처리부에서는 DSP의 강력한 산술연산기능으로 MPEG 복호화 알고리듬을 수행하며 시스템 제어부에서는 마이크로 콘트롤러의 장점인 저가, 저전력의 제어 기능으로 사용자 인터페이스 및 파일 관리, 비트스트림 제어를 담당하도록 구성된다. 입력부에서는 Multi Meadia Card(MMC)를 지원하고, PC와 호환 가능하도록 파일 관리 시스템으로 운용되며 직렬 통신의 데이터 전송과 16비트 해상도 및 최대 48kHz 표본화주파수로 스테레오 출력이 가능하다. 구현된 시스템은 이중 코어를 이용하여 DSP의 연산량 및 동작속도의 감소로 인한 저가, 저전력의 효과로 인해 휴대장치에 적합하다.