• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2016.10a
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• pp.770-772
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• 2016

본 논문에서는 음성신호처리 시스템에 유용하게 사용되는 음성신호의 특징 파라미터를 출력하는 스펙트럼 필터모델을 사용하여, 배경잡음 환경 하에서 음성신호 중의 잡음을 제거하는 알고리즘을 제안한다. 따라서 본 논문에서는 배경잡음을 제거할 때 고려해야 할 인간의 청각특성이 포함된 음성의 진폭 스펙트럼에 의한 청각필터의 특성을 도입한다. 본 논문의 실험에서 사용한 성능평가의 방법으로는 음절 명료도의 테스트에 적합한 주관적인 평가인 주파수 영역에서의 스펙트럼 왜곡률(Spectral Distortion, SD)을 사용하여 실험결과를 비교하고 고찰한다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.199-202
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• 2002

본 논문에서는 사람의 청각 모델을 기반으로 잡음에 의해 손상된 음성 신호로부터 잡음 신호의 마스킹 특성과 신호에너지의 지각(知覺)을 나타내는 임계대역(critical band)에서의 잡음 에너지에 대한 지각 패턴인 noise excitation pattern을 이용한 잡음 에너지 차감과 잡음 추정 오차에 의한 변형된 음성신호 내의 순음(tonal) 성분과 비순음(non-tonal)성분의 보정을 통해 효과적인 음성 품질의 개선을 위한 연구를 하였다.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.2
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• pp.171-178
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• 2016

An aircraft in flight and ATC(: Air Traffic Controllers) working in the Ground Control Center carry out a voice communication using the radio. Voice signal to be transmitted from the aircraft is received to a plurality of terrestrial sites around the country at the same time. The ATC receives the various quality of voice signal from the aircraft depending on the distance, speed, weather conditions and adjusted condition of the antenna and the radio. The ATC carries out a voice communication with aircraft in the optimal conditions finding the best voice signal. However, the present system chooses the values of the CD(: Carrier Dectect) which is determined to be superior to, based on the input voice level, as optimal channel. Thus this system can not be seen to select the optimal channel because it doesn't consider the effect of the noise which influences on the communication quality. In this paper, after removing the noise in the voice signal, we could give the digitized information and an improved voice signal quality, so that users can select an optimal channel. By using it, when operating the training eavesdropping system or the aircraft control, we can expect prevention accident and improvement of training performance by selecting the improved quality channel.

• Electronics and Telecommunications Trends
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• v.21 no.6 s.102
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• pp.77-85
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• 2006

2006년 5월 고품질 인터넷 전화(VoIP) 서비스에 사용할 수 있는 가변대역 광대역 음성 코덱 기술이 ITU-T SG16 회의에서 국제 표준으로 확정되었다. ITU-T에서 표준화된 G.729.1 코덱은 국내 IPR이 포함된 최초 음성 코덱 분야의 표준이다. 이 코덱은 인터넷 전화 서비스에서 가장 널리 사용되는 G.729를 기반으로 8-32kbps 범위 내에서 임베디드 형태로 비트열 계층을 쌓아가는 구조로 협대역 신호(300-3400Hz)부터 광대역 신호(50-7000Hz)를 압축, 복원한다. 이 기술은 기존의 인터넷 전화에서 사용하고 있는 코덱의 낮은 품질, 대역폭 확장, 품질 제어가 곤란한 단점을 개선하여 인터넷전화 서비스 시장을 활성화 할 것으로 기대된다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.05a
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• pp.285-290
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• 2001

본 논문에서는 멀티미디어 서비스 요소 중의 하나인 VoIP(Voice Over Internet Protocol)망에서의 음성 품질에 대한 평가를 위해 VoIP망에서 송화자 내용- 발생과정에 있어서 어느 정도 완전히 표현되었는가를 나타내는 송화품질과 음성의 전송계를 통해 수화자에게 전달되는 과정에서 왜곡이나 잡음 등의 방해요인에 의해 열화되는 정도를 나타내는 전송품질, 그리고 수화자가 청각에서 신호처리 과정을 거친 송화자의 내용을 어느 징도 이해할 수 있는지를 나타내는 수화품질에 대한 주관적 방법을 평가한 후 통화품질을 측정한 내용을 분석하여 그 원인과 개선책에 대한 방법을 제시하고자 한다.

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.5
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• pp.811-816
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• 2019

This paper first proposes a technique to separate clean speech signals and mixed noise signals by using an independent vector analysis algorithm of frequency band for 4 channel speech source signals with a noise. An improved output speech signal from the proposed independent vector analysis algorithm is obtained by using the cross-correlation between the signal outputs from the frequency domain delay-sum beamforming and the output signals separated from the proposed independent vector analysis algorithm. In the experiments, the proposed algorithm improves the maximum SNRs of 10.90dB and the segmental SNRs of 10.02dB compared with the frequency domain delay-sum beamforming algorithm for the input mixed noise speeches with 0dB and -5dB SNRs including white noise, respectively. Therefore, it can be seen from this experiment and consideration that the speech quality of this proposed algorithm is improved compared to the frequency domain delay-sum beamforming algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1845-1851
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• 2007

Background noise and room reverberation are two causes of degradation in speech in listening situations. Many algorithms developed to enhance reverberant speech. In this paper we propose a dereverberation method for enhancement of speech using modified the linear prediction(LP) residual in reverberant room condition. The proposed dereberberation method based on the fact that the signification excitation of the vocal tract system takes place at the instant of glottal closure in voiced speech. Our method used delay information form each sensor, and we need reverberant signals from 3 sensors. We obtain a new LP residual signal using modified IP residual combination which derived form weighting of the LP residual and the Hilbert transform of LP residual. The nature of the coherently added Hilbert envelop has several large amplitude spikes because of the effects of noise and reverberation. This residual of the clean speech is used to excite the time-varying all-pole filter to obtain the enhanced speech. We achieved simulation of proposed algorithm for performance analysis in reverberation environment. The proposed algorithm improves substantially the quality of reverberant speech.

• Journal of the Korea Society for Simulation
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• v.16 no.3
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• pp.75-87
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• 2007

As the VoIP service quality is getting better and many shortcomings are being overcome, users are getting interested in this service. Also, there are several additional features that provide a convenience to users such as presence service, instant messaging service and so on. But, as there are always two sides of rein, some security issues have users hesitate to make use of it. This paper deals with one of the issues, the VoIP spam problem. We took into account the signal pattern of voice message in spam call and we have constructed voice signal models of normal call, normal call with noise and spam call. Each voice signal case is inserted into our spam decision algorithm which detects the spam calls based on the amount of information in the call signal. We made use of the DEVS-$Java^{TM}$ for our modeling and simulation. The contribution of this work is in suggestion of a way to detect voice spam call signal and testing of the method using modeling and simulation methodology.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.70-77
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• 2017

Bandwidth extension is a technique to improve speech quality, intelligibility and naturalness, extending from the 300 ~ 3,400 Hz narrowband speech to the 50 ~ 7,000 Hz wideband speech. In this paper, an Artificial Bandwidth Extension (ABE) module embedded in the Opus audio decoder is designed using the information of narrowband speech to reduce the computational complexity of LPC (Linear Prediction Coding) and LSF (Line Spectral Frequencies) analysis and the algorithm delay of the ABE module. We proposed a spectral envelope extension method using DBN (Deep Belief Network), one of deep learning techniques, and the proposed scheme produces better extended spectrum than the traditional codebook mapping method.