• Journal of Korea Multimedia Society
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• v.4 no.1
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• pp.56-66
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• 2001

The spectral subtraction algorithm using a criterion based on the human perception has been recently developed. The speech processed with Virag's algorithm sounds more pleasant to a human listener than those obtained by the classical methods. However, Virag's algorithm requires a robust voice activity detector (VAD). In the ESS (extended spectral subtraction) algorithm without VAD, the residual noise becomes more noticeable as the SNR decrease. In this paper we propose a new speech enhancement method, the combination of Wiener filter and spectral subtraction based on noise masking characteristics in the human auditory system. There is no need of VAD because the noise can be successively updated even during speech activity using Wiener filter. The adjustment of the subtraction parameter based on the masking threshold makes the residual noise inaudible. The proposed method has been compared with conventional spectral subtraction algorithms. Objective and subjective evaluation of the proposed system is performed with several noise types having different time-frequency distributions. The application of objective measures, the study of the speech spectrograms, as well as subjective listening tests, confirm that the enhanced speech with proposed algorithm is more pleasant to a human listener.

• Journal of KSNVE
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• v.10 no.2
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• pp.345-352
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• 2000

Recently, it becomes to be very important to reduce the cabin noise of passenger ship, according to the trend of speedy and luxury ship. The noise reduction and control techniques should be considered as important factors from the viewpoint of the sound problem of cabin. Therefore, ship designer has to improve the sound quality as well as to redece the sound pressure level in cabins. In this paper, for the new approach of these problems, we tried to find the trends of noise and sound quality of high-speed coastal passenger ships. Loudness, roughness, fluctuation strength, and sharpness are selected as the parameters for the evaluation of sound quality. The parameters are calculated by using the sound measured in cabin while the ship is running. Furthermore we tried to find the trend of each parameter in cabins and compare with that of sound pressure level. As results, we find that the loudness is linearly proportional to sound pressure level. But, the other parameters show different trends which may be caused by ship motion on the wave and fluctuation of propelling power.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.513-516
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• 2001

마이크로폰을 배열을 이용한 음성개선 기법 중에서 저주파 영역에서의 높은 상관성과 고주파 영역에서의 spatial aliasing을 동시에 고려하기 위하여 대수적인 선형 마이크로폰 배열을 사용하고 웨이브렛 도메인에서의 Wiener 필터에 기반한 postfiltering을 수행하는 방법이 제안된 바 있는데[l], 본 논문에서는 이 방법의 문제점을 분석하고 해결방안을 제시하였다. 제안한 알고리즘을 사용하여 시뮬레이션한 결과, 마이크에 입력되는 음성신호의 SNR이 0dB일 때와 l0dB일 때, 기존의 알고리즘에 비해 약 1.7dB와 2.5dB의 성능개선이 있었으며, 청취실험을 통해서도 음질의 향상을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.12 no.3
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• pp.62-67
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• 1993

부호 여기된 선형예측 음성보코더들은 4800bps의 낮은 전송율에서도 좋은 음질을 제공한다. CELP 형 보코더의 피치검색법의 문제점중에 하나는 피치 검출시의 조오류에 의해 예측이득이 저하된다는 점이다. 본 논문에서 우리는 CELP 보코더의 피치 예측이득을 개선하는 한 새로운 피치검색법을 제안하였다. 제안한 방법은 예비피치들을 검출하여 이들중 피치 예측이득이 최대인 값을 선정하는 방법을 적용하였다. 이 방법을 여러 화자의 발성에 대해 적용한 결과 피치 예측이득율 6.1% 정도 개선할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.2
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• pp.61-67
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• 2000

Recent progress in digital and VLSI technology has offered new possibility fer noticeable advance of hearing aids. Yet, environmental noise remains one of the major problems to hearing aid users. This paper describes results which speech recognition performance and speech discrimination performance was measured for listeners with sensorineural hearing loss, while listeners in speech-band noise. In addition, to ameliorate hearing-aided environments of hearing impaired listeners, environmental noise reduction using speech enhancement techniques are investigated as a front-end of conventional hearing aids. Speech enhancement techniques are implemented in a realtime system equipped with DSP board. The clinical test results suggest that the speech enhancement technique may work in synergy with gain functions fer the greater SNR improvement as the preprocessing algorithm of digital hearing aids.

• Annual Conference of KIPS
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• 2018.10a
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• pp.673-676
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• 2018

4차 산업혁명이 도래하면서 정보 통신 기술 및 융합 기술의 발전에 힘입어 소리 데이터를 이용한 연구가 활발하게 진행되고 있다. 소리 데이터를 이용한 학술적 프로토타입 연구들을 실제 환경에서 운용하기 위해서는 소리 취득 시 발생하는 다양한 잡음 환경에서도 원시 데이터(raw data)에 근접한 정보를 취득할 수 있는 시스템의 강인함이 보장되어야 한다. 본 논문에서는 SEGAN(Speech Enhancement Generative Adversarial Network) 모델을 활용하여, 전처리 및 후처리 과정이 필요 없이 원시 데이터를 대상으로 하는 end-to-end 방식의 소리 음질 향상 시스템을 제안한다. 제안하는 시스템은, 축산업 분야의 돼지 호흡기 질병 소리 데이터를 이용하여 실험하였으며, 여러 가지 잡음 상황(인위적인 잡음, 실제 환경 잡음)에서 소리 음질이 개선됨을 실험적으로 검증하였다.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.17 no.6
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• pp.154-159
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• 2003

In this paper, we proposed variable threshold dual rate ADPCM coding method which adapts two coding rates of the standard ADPCM of ITU G.726 for speech quality improvement at a comparably low coding rates. The ZCR(Zero Crossing Rate) is computed for speecd data and under the noisy environment, noise data dominant region showed higher ZCR and speech data dominant region showed lower ZCR. The speech data with the higher ZCR is encoded by low coding rate for reduced coded data and the speech data with the lower ZCR is encoded by high coding rate for speech quality improvements. For coded data, 2 bits are assigned for low coding rate of 16[Kbps] and 5 bits are is assigned for high coding rate of 40[Kbps]. Through the simulation, the proposed idea is evaluated and shown that the variable dual rate ADPCM coding technique shows the qood speech quality at low coding rate.

• Journal of KSNVE
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• v.11 no.2
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• pp.196-199
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• 2001

차량 실내소음의 주요 음원은 엔진과 그 주변부에 있다. 엔진에서 발생되는 소음을 차단하여 차실내 소음수준을 개선하기 위하여 엔진룸과 차실과 경계면을 이루고 있는 대시(dash)부의 차음성능이 중요하다. 특히, 음질을 결정하는 주파수 대역인 250Hz 이상의 소음을 제어하기 위하여 다양한 대시 판넬 뿐만 아니라 대시 판넬 전후에 다층구조의 차음재를 부착한다.(중략)

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.2
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• pp.108-116
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• 2014

With the increasing the importance of emotional quality of vehicle, the sound quality of systems with electric motor components has become increasingly important. Electric motors are used for windows, seats, sun roof, mirrors, steering columns, windshield wiper and climate control blowers, etc. In this paper, a study was conducted to identify sound quality factors that contribute to customer's satisfaction and preference of the window lift system. Jury test for subjective evaluation was carried out and sound quality index was developed. Averaged sound pressure level and sharpness were significant factors when glass moves down. Also, maximum loudness at stop section and averaged loudness were significant factor when glass moves up. Next, noise source identification was carried out using beam forming method during glass transferred section and impulsive noise at stop section. Several improvement methods were applied using the source identification result. And finally, the degree of sound quality improvement was judged using sound quality index.