• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2004.04a
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• pp.466-469
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• 2004

In this paper, real-time voice change method using pitch change technique is proposed to change one's voice to the other voice. For this purpose, sampling rate change method using DFT (Discrete Fourier Transform) method and time scale modification method using SOLA (Synchronized Overlap and Add) method is combined to change pitch. In order to evaluate the performance of the proposed method, voice transformation experiments were conducted. Experimental results showed that original speech signal is changed to the other speech signal in which original speaker's identity is difficult to find. The system is implemented using TI TMS320C6711DSK board to verify the system runs in real time.

• Phonetics and Speech Sciences
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• v.3 no.2
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• pp.89-96
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• 2011

In this paper, details of the design and implementation of a voice source simulator using Simulink and Matlab are discussed. This simulator is an implementation by model-based design concept. Voice sources can be analyzed and manipulated through various factors by choosing options from GUI input and selecting pre-defined blocks or user created ones. This kind of simulation tool can simplify the procedure of analyzing speech signals for various purposes such as voice quality analysis, pathological voice analysis, and speech coding. Also, basic analysis functions are supported to compare the original signal and the manipulated ones.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.799-800
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• 2006

This thesis implemented security systems of voice awareness which is higher accessible than existing security system using biological authentication system and is inexpensive in module of security device, and has an advantage in usability. Proposed the security systems of voice awareness implemented algorithm for characteristic extraction of inputted speaker's voice signal verification, and also implemented database of access control that is founded on extractible output. And a security system of voice awareness has a function of an authority of access control to system.

• Journal of Digital Convergence
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• v.11 no.6
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• pp.169-174
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• 2013

Speech recognition, the problem of performance degradation is the difference between the model training and recognition environments. Silence features normalized using the method as a way to reduce the inconsistency of such an environment. Silence features normalized way of existing in the low signal-to-noise ratio. Increase the energy level of the silence interval for voice and non-voice classification accuracy due to the falling. There is a problem in the recognition performance is degraded. This paper proposed a robust speech detection method in noisy environments using a silence feature normalization and voice energy maximize. In the high signal-to-noise ratio for the proposed method was used to maximize the characteristics receive less characterized the effects of noise by the voice energy. Cepstral feature distribution of voice / non-voice characteristics in the low signal-to-noise ratio and improves the recognition performance. Result of the recognition experiment, recognition performance improved compared to the conventional method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.112-118
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• 2012

In daily life, there are many information broadcasting by using voice information systems. If surrounding noises are mixed with the information signals, the clarity of the signal become down graded too much to understand. Surrounding noises are not uniformed, but very irregular signals always changing. Therefore, it is very hard to control the output signals along with the irregular signals. This paper suggests a method to change the level of the voice information adapting to the surround noise in the white noise environment. The surround noise level is measured by subtracting the stored output voice signal from the voice signal degraded by the noise. The noise is used to estimation of SNR. And, the method to change the output level of voice signal adapting to the noise level. The suggested adaptive voice information system has the advantage to improve listeners' speech perception and to use amplifier's energy effectively.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.73-76
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• 2006

유비쿼터스 컴퓨팅에서 중요한 기술 중 하나로 평가되는 음성인식 및 합성기술은 인간과 컴퓨터의 상호 작용에 있어 가장 편리하고 보편적인 방법이다. 음성인식 및 합성기술을 이용한 인간과 컴퓨터 상호작용 기반의 애플리케이션의 개발을 위해 음성 확장성 생성 언어(VoiceXML)을 이용하면 음성 인식 및 합성에 관한 전문 지식이 없어도 애플리케이션 제작을 쉽게 할 수 있다는 장점이 있어서 음성인식 및 합성기술의 인프라 구축과 저변 확대를 목적으로 일부 국내 업체들은 VoiceXML을 이용한 음성 애플리케이션을 제작하고 실험할 수 있도록 VoiceXML 실험 환경을 제공한다. 본 논문에서는 기존에 공개된 실험 환경을 소개하고, 다양한 실험 환경 제공을 위해 기존에 있던 Linux기반의 실험 환경과는 다른 Windows NT기반의 IIS(Internet Information Service)를 이용한 VoiceXML실험 환경을 제안하고 구현하였다. 그 결과 ASP(Active Server Page)와 ADO(ActiveX Data Object)를 이용한 VoiceXML음성 애플리케이션 실험이 가능한 환경을 구축하였고, 사용자 평가 결과 제안한 방법이 유효하다는 것을 확인하였다.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1004-1016
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• 2007

In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.1
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• pp.46-54
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• 2002

The world was opened by communication network because of fast improvement and diffusion of information communication. And information was effected in important factor that control economy improvement of the country. The country should improve the information security system because of necessity to maintain its information security independently. Therefore we have used the SEED cipher algorithm and designed the cipher chip of the voice band signal using the Xilinx Co. XCV300PQ240 chip. At the result we designed the voice signal cipher chip of the maximum frequency 47.895MHz and the total equivalent gate 27,285.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.285-288
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• 2001

In the speech analysis, to estimate formant center frequencies exactly is very important. If we know formant frequencies, we can expect which pronunciation is uttered. Generally, the magnitude of first formant frequency in voiced speech is 10dB more than other formant frequency. So, the shape of voice signal in time domain is affected by mainly first formant. Therefore we can get first formant frequency roughly by using ZCR(Zero Cross Rate). In this paper, we proposed the improvement method to get first formant frequency by using ZCR. We did autocorrelation before getting ZCR. This procedure makes voice signal smooth so, first formant in voice signal is emphasized. As a result of this method, we got more exact ZCR and first formant frequency. Conventional method of formant estimate is done in frequency domain but proposed method is done in time domain. So, this is very simple.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.51-55
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• 2010

Tele-marketing service has been provided through voice media in a several places such as modern call centers. In modern call centers, they are trying to measure their service quality, and one of the measuring method is a extracting speaker's feeling information in their voice. In this study, it is proposed to analyze speaker's voice in order to extract their feeling information. For this purpose, a person's feeling is categorized by analyzing several types of signal parameters in the voice signal. A person's feeling can be categorized in four different states: joy, sorrow, excitement, and normality. In a normal condition, excited or angry state can be major factor of service quality. In this paper, it is proposed to select a conversation with problems by extracting the speaker's feeling information based on pitches and amplitudes of voice.