• Journal of the Korean Society of Safety
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• v.22 no.3 s.81
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• pp.7-12
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• 2007

This paper presents the development of an electrical fire detection system using digital temperature and current detectors in order to sound for electrical fire in advance. As the demand for electricity is increasing and industrial facilities are getting more complex and larger in size, the losses of human life and property are on the increase by electrical fires. In order to prevent electrical fires, it is required to find out fire signatures, or electric signal of the overcurrent and overheating. Therefore, in this paper, developed is an electrical fire detection system based on the detection of signal for overcurrent and overheating to prevent electrical accidents in advance that happen in electrical wires. The developed system gives an alarm by computer monitor, speaker system and mobile phone before electrical fires occur and give severe damages to human beings and properties, and the system can be implemented and supplied for business and residental buildings at a low price. The usefulness and validity of the system, also, verified in this paper by case study and experiments.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.143-146
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• 2004

일반적으로, 화상회의에서 화자의 위치를 검출하는 것은 음향 신호를 기반으로 이루어져 왔다. 그러나 물리적인 환경의 제약이나 화자 검출 시스템의 한계를 벗어나는 노이즈가 발생하는 경우에는 검출 시스템의 성능저하를 초래하게 된다. 본 논문에서는 음향 기반의 검출 시스템과 독립적으로, 혹은 상호 보완적으로 사용될 수 있는 영상 기반의 화자 검출 알고리즘에 대하여 제안하고자 한다. 화자의 위치에 관한 정보는 화상회의에 한층 사실감을 부여하는 3 차원 오디오 재생에 사용될 수 있다.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.39-40
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• 2007

For efficient interaction between human and robots, speech interface is a core problem especially in noisy and reverberant conditions. This paper analyzes main issues of spoken language interface for humanoid robots, such as sound source localization, voice activity detection, and speaker recognition.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.187-190
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• 2002

본 논문에서는 음성 신호에서 발성 화자가 바뀌는 시점을 자동적으로 찾아내는 화자변화 구간 검출에 대하여 연구하였다. 화자변화 검출을 위해서는 음성 신호에 나타나는 화자 개별성에 의한 차이만 비교해야 하는데 실제 환경에서는 화자들이 동일한 내용의 발성을 하지 않으므로 다른 발성내용에 의한 정보가 포함되어 검출 성능을 저하시킨다. 그러므로 각 화자의 개별특성만 강조되도록 발성내용에 포함된 음성학적 정보의 영향을 제거하는 방법을 통해 검출 성능을 향상시켰다.

• Speech Sciences
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• v.10 no.2
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• pp.85-95
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• 2003

In speech signal processing, it is very important to detect the pitch exactly in speech recognition, synthesis and analysis. If we exactly pitch detect in speech signal, in the analysis, we can use the pitch to obtain properly the vocal tract parameter. It can be used to easily change or to maintain the naturalness and intelligibility of quality in speech synthesis and to eliminate the personality for speaker-independence in speech recognition. In this paper, we proposed a new pitch detection algorithm. First, positive center clipping is process by using the incline of speech in order to emphasize pitch period with a glottal component of removed vocal tract characteristic in time domain. And rough formant envelope is computed through peak-fitting spectrum of original speech signal infrequence domain. Using the roughed formant envelope, obtain the smoothed formant envelope through calculate the linear interpolation. As well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. Inverse fast fourier transform (IFFT) compute this flattened harmonics. After all, we obtain Residual signal which is removed vocal tract element. The performance was compared with LPC and Cepstrum, ACF. Owing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.4
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• pp.178-183
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• 2006

As a lot of researches on the speech signal processing are performed due to the recent rapid development of the information-communication technology. the pitch period is used as an important element to various speech signal application fields such as the speech recognition. speaker identification. speech analysis. or speech synthesis. A variety of algorithms for the time and the frequency domains related with such pitch period detection have been suggested. One of the pitch detection algorithms for the time domain. AMDF (average magnitude difference function) uses distance between two valley points as the calculated pitch period. However, it has a problem that the algorithm becomes complex in selecting the valley points for the pitch period detection. Therefore, in this paper we proposed the modified AMDF(M-AMDF) algorithm which recognizes the entire minimum valley points as the pitch period of the speech signal by using the rotation transform of AMDF. In addition, a threshold is set to the beginning portion of speech so that it can be used as the selection criteria for the pitch period. Moreover the proposed algorithm is compared with the conventional ones by means of the simulation, and presents better properties than others.

• The Journal of the Acoustical Society of Korea
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• v.6 no.3
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• pp.60-67
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• 1987

In this paper, a phoneme classification method for Korean speech recognition has been proposed and its performance has been studied. The phoneme classification has been done based on the phonemic features extracted by the prewindowed recursive least-square (PRLS) algorithm that is a kind of adaptive filter algorithms. Applying the PRLS algorithm to input speech signal, precise detection of phoneme boundaries has been made, Reference patterns of Korean phonemes have been generated by the ordinery vector quantization (VQ) of feature vectors obtained manualy from prototype regions of each phoneme. In order to obtain the performance of the proposed phoneme classification method, the method has been tested using spoken names of seven Korean cities which have eleven different consonants and eight different vowels. In the speaker-dependent phoneme classification, the accuracy is about $85\%$ considering simple phonemic rules of Korean language, while the accuracy of the speaker-independent case is far less than that of the speaker-dependent case.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.23 no.6
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• pp.1157-1167
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• 2013

The development of electronic devices has recently led to many problems such as personal information rape and leakage of business information. Conventional loud-speakers have been generally used to output devices. It can be, however, operated as a micro-phone which was abused as a means for eavesdropping since the speaker and microphone have basically the equivalent structure. Most importantly, the general peoples are not aware of the approaching danger about using speaker as microphone. And, traditional eavesdropping detection equipment does not check the attack. In this paper, we demonstrate that there is a serious danger and vulnerability in using loud-speakers since they can be used as eavesdropping devices.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.3
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• pp.170-175
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• 2009

In this paper, we propose real time automatic speaker verification (ASV) system to protect personal information on personal digital assistant (PDA) device. Recently, the capacity of PDA has extended and been popular, especially for mobile environment such as mobile commerce (M-commerce). However, there still exist lots of difficulties for practical application of ASV utility to PDA device because it requires too much computational complexity. To solve this problem, we apply the method to relieve the computational burden by performing the preprocessing such as spectral subtraction and speech detection during the speech utterance. Also by applying the hidden Markov model (HMM) optimal state alignment and the sequential probability ratio test (SPRT), we can get much faster processing results. The whole system implementation is simple and compact enough to fit well with PDA device's limited memory and low CPU speed.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.50-55
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• 2001

This paper presents a korean isolated words recognition algorithm which used new endpoint detection method, auditory model, 2D-DCT and new distance measure. Advantages of the proposed algorithm are simple hardware construction and fast recognition time than conventional algorithms. For comparison with conventional algorithm, we used DTW method. At result, we got similar recognition rate for speaker dependent korean isolated words and better it for speaker independent korean isolated words. And recognition time of proposed algorithm was 200 times faster than DTW algorithm. Proposed algorithm had a good result in noise environments too.