• 대한전기학회:학술대회논문집
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• 대한전기학회 2008년도 학술대회 논문집 정보 및 제어부문
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• pp.290-291
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• 2008

The most popular speech feature used in speech recognition today is the Mel-Frequency Cepstral Coefficients (MFCC) algorithm, which could reflect the perception characteristics of the human ear more accurately than other parameters. This paper adopts MFCC and its first order difference, which could reflect the dynamic character of speech signal, as synthetical parametric representation. Furthermore, we quote Dynamic Time Warping (DTW) algorithm to search match paths in the pattern recognition process. We use the software "GoldWave" to record English digitals in the lab environments and the simulation results indicate the algorithm has higher recognition accuracy than others using LPCC, etc. as character parameters in the experiment for Digital Isolated Word Recognition (DIWR) system.

• 한국산학기술학회논문지
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• 제10권4호
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• pp.805-810
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• 2009

본 논문은 음주와 비음주 상태를 판별하기 위한 포어먼트 변화의 특징에 관한 연구이다. 단음절의 실험을 통하여 음주 음성신호에 비하여 비음주 음성신호의 F1, F2, F3의 포어먼트가 높게 나타나는 것을 확인하였으며, 또한 포어먼트는 음주와 비음주 상태를 구별하는데 매우 유효하다는 것을 알 수 있었다.

• 한국정보통신학회논문지
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• 제5권3호
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• pp.443-449
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• 2001

잡음환경에서의 음성인식은 실제의 환경에서의 음성인식에서 매우 중요한 애로기술로써 이를 해결하기 위한 연구는 꾸준히 연구되고 있다. 따라서 본 연구는 음성인식분야에서 가장 많이 사용하고 있는 HMM처리 시잡음처리의 문제점을 주파수 가중치 부가 HMM으로 해결하는 방법을 제안하고 그 성능을 인식실험을 통하여 검토하였다. 그 결과 SS처리를 함께 사용하는 $MCE-\mu$, MCE-$\rho$가 가장 잡음에 강한 방식임을 알 수 있었다.

• 한국통신학회논문지
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• 제15권11호
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• pp.937-944
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• 1990

DSBC 음성 부호화기의 성능 개선에 관한 두 방법을 제안하였다. 첫째는 계산량이 많은 종래의 비트할당을 수정함으로써 계산량을 줄일 수 있는 방법이고 둘째는 비전송 대역 재생시 백색잡음 주입으로 인한 허상 문제를 제거하기 위한 부대역 웨이팅 방법이다. 시뮬레이션 겨로가 검토된 방법은 음성 출력의 성능 향상에 응용할 수 있음을 나타내었다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1987년도 전기.전자공학 학술대회 논문집(II)
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• pp.1322-1325
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• 1987

In this study, the excitation pattern, which can be sensated by a man having hearing loss due to the damage of inner ear, is extracted, and the procedure of the auditory speech signal processing is simulated with the computer. Therefore, the excitation pattern is extracted by the neural tuning model satisfying the physiological characteristic of the inner ear and by the infor.ation extracted from speech signal. The firing pattern is also extracted by inputting this excitation pattern to the auditory neural model. With this extracted firing pattern, the possibility that the patient can sensate the speech signal is studied by the computer simulation.

• 한국음향학회지
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• 제10권4호
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• pp.12-18
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• 1991

음성 신호처리 분야에서 기본주파수를 정확히 검출하는 것이 아주 중요하다. 주파수 영역에서 피치검출 방법의 문제점은 대체로 배경잡음이나 제 1 포먼트에 의하여 발생한다. 그러므로, 본 논문에서는 스펙트럼 AMDF 함수를 이용하여 잡음의 영향이나 제 1 포먼트의 영향을 줄이는 주파수영역 피치검출 앨고리즘을 제안하였다. 여러 가지 컴퓨터 시뮬레이션 결과 제안한 앨고리즘이 기본주파수 검출에 효과적으로 나타났다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 신호처리소사이어티 추계학술대회 논문집
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• pp.45-48
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• 2003

In general, speech coding methods are classified into the following three categories: the waveform coding, the source coding and the hybrid coding. Fast speaking is possible to encode with a few information compared with slow speaking rate. In case of speaking rate, low frequency band is more important than high frequency band while listening. Speech vocoding technique is developing to way with low bit rate and complexity and high sound quality. the CELP type of vocoder support very good sound quality with low bit rate but these vocoders don't consider about the speaking rate. When we consider speaking rate and encode the frame depending on the speaking rate, the bit rate is able to reduce the bit rate than the conventional vocoder. We propose the technique to estimate the speaking rate and applied PSOLA technique in case of the frame of slow speaking rate. As a result of simulation bit rate can be reduced about 300 bps.

• 대한전자공학회논문지
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• 제18권1호
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• pp.41-50
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• 1981

The human speech contains many redundancies. To economize communication channel or memory size for a computerized synthesis of human voices, it is necessary to compress the data before sending. We have treated human speech organ as an eighth order dynamic system which is time varying as the person speaks. Using an anaylyzer of our design, each eight parameters are obtained for the vowels [아], [어], [오], [우], [으], [이], [애], and (외) of korean language with considerable discrepancies between persons. Supplying those parameters to a synthesizer which we have made, we have sucoeeded in the simulation of human speech for the above mentioned vowels of Korean language and observed that they bear all the features of the original speakers.

• The Journal of the Acoustical Society of Korea
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• 제25권4E호
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• pp.142-147
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• 2006

Blind signal separation is known as a powerful tool for enhancing noisy speech in many real world environments. In this paper, it is demonstrated that the performance of blind signal separation can be further improved by combining with a null beamformer (NBF). Cascading the blind source separation with null beamforming is equivalent to the decomposition of the received signals into the direct parts and reverberant parts. Investigation of beam patterns of the null beamformer and blind signal separation reveals that directional null of NBF reduces mainly direct parts of the unwanted signals whereas blind signal separation reduces reverberant parts. Further, it is shown that the decomposition of received signals can be exploited to solve the local stability problem. Therefore, faster and improved separation can be obtained by removing the direct parts first by null beamforming. Simulation results using real office recordings confirm the expectation.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2006년도 춘계 학술대회 발표논문집
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• pp.33-36
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• 2006

This paper describes a speech recognizer implemented on PDAs. The recognizer consists of feature extraction module, search module and utterance verification module. It can recognize 37 words that can be used in the telematics application and fixed-point operation is performed for real-time processing. Simulation results show that recognition accuracy is 94.5% for the in-vocabulary words and 56.8% for the out-of-task words.