• 대한임베디드공학회논문지
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• 제10권1호
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• pp.41-48
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• 2015

To deal with the emergency of the solitary aged people, we have developed a collapse-sensing phone, in which a collapse sensor, a GPS receiving chipset and a CDMA sending chipset are included. The general cellular phone is somewhat expensive communication device using sound and characters, but the collapse-sensing phone is a cheaper and popular version. If the collapse sensor recognizes a certain of collapse of the aged people, CDMA sending chipset will send the location of the phone which is received from satellite by GPS receiving chipset. In this paper, a collapse recognition algorithm which is developed by using much experimental data, will be introduced to explain how to recognize the real collapse from fast sitting or immediate standing after collapse. Once a true collapse is ecognized, the phone-ID and the coordinate will be sent to the server of administrative office via CDMA network. And the position of emergency will be displayed on the GIS with the rescue center.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1997년도 추계학술대회 논문집 학회본부
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• pp.594-596
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• 1997

This paper presents the speaker-dependent Korean isolated digit recognition system using advanced competitive learning. Since competitive learning algorithms are easy and simple to implement, they are used in various fields. The proposed recognition algorithm consists of three procedures: comparing winning number of codebook vectors, selecting the representative vector out of codebook vectors, and generating a new codebook with the representative vectors. In this paper, we use a sound blaster 16 for obtaining speech data. Speech data are sampled by 16 bits and 11 kHz sampling rate.

• 음성과학
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• 제10권1호
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• pp.27-39
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• 2003

This study attempts to verify the reliability of automatically generated segment labels as compared to those obtained by conventional labelling by hand. First of all, an autolabeller is constructed using the standard HMM speech recognition technique. For evaluation, we compare the automatically generated labels with manually annotated labels for the same speech data. The comparison is performed by calculating the temporal difference between an autolabel boundary and its corresponding hand label boundary. When the mismatched duration between two labels falls within 10 msec, we consider the autolabel as correct. The results suggest that overall 78% of autolabels are correctly obtained. It is found that the boundary of obstruents is better aligned than that of sonorants and vowels. In case of stop sound classes, strong stops in manner-of-articulation wise and velar stops in place-of-articulation wise show better performance in boundary alignment. The result suggests that more phone-specific consideration is necessary to improve autosegmentation performance.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2001년도 하계학술대회 논문집 D
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• pp.2610-2612
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• 2001

This study examines the recognition of the user's sound command based on speech recognition and natural language processing, and develops the natural language interface agent which can analyze the recognized command. The natural language interface agent consists of speech recognizer and semantic interpreter. Speech recognizer understands speech command and transforms the command into character strings. Semantic interpreter analyzes the character strings and creates the commands and questions to be transferred into the application program. We also consider the problems, related to the speech recognizer and the semantic interpreter, such as the ambiguity of natural language and the ambiguity and the errors from speech recognizer. This kind of natural language interface agent can be applied to the telephony environment involving all kind of communication media such as telephone, fax, e-mail, and so on.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2003년도 하계학술대회 논문집
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• pp.18-21
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• 2003

In general it has been considered to be the difficult problem that we divide continuous speech into short interval with having identical phoneme quality. In this paper we used Gaussian Mixture Model (GMM) related to probability density to divide speech into phonemes, an initial, medial, and final sound. From them we peformed continuous speech recognition. Decision boundary of phonemes is determined by algorithm with maximum frequency in a short interval. Recognition process is performed by Continuous Hidden Markov Model(CHMM), and we compared it with another phoneme divided by eye-measurement. For the experiments result we confirmed that the method we presented is relatively superior in auto-segmentation in korean speech.

• International Journal of Advanced Culture Technology
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• 제9권4호
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• pp.320-328
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• 2021

As the number of visually impaired people steadily increases, interest in independent walking is also increasing. However, there are various inconveniences in the independent walking of the visually impaired at present, reducing the quality of life of the visually impaired. The white cane, which is an existing walking aid for the visually impaired, has difficulty in recognizing upper obstacles and obstacles outside the effective distance. In addition, it is inconvenient to cross the street because the sound signal to help the visually impaired cross the crosswalk is lacking or damaged. These factors make it difficult for the visually impaired to walk independently. Therefore, we propose the design of an embedded system that provides traffic light recognition through object recognition technology, voice guidance using TTS, and upper obstacle recognition through ultrasonic sensors so that blind people can realize safe and high-quality independent walking.

• 한국음향학회지
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• 제36권6호
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• pp.401-406
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• 2017

본 논문에서는 양서류 울음소리를 통한 종 인식 시스템 개발을 위해, 음향 신호 분석에서 활용되는 주요 알고리즘의 인식 성능을 평가했다. 먼저, 멸종위기 종을 포함하여 총 9 종의 양서류를 선정하여, 각 종별 울음소리를 야생에서 녹음하여 실험 데이터를 구축했다. 성능평가를 위해, MFCC(Mel Frequency Cepstral Coefficient), RCGCC(Robust Compressive Gammachirp filterbank Cepstral Coefficient), SPCC(Subspace Projection Cepstral Coefficient)의 세 특징벡터와 GMM(Gaussian Mixture Model), SVM(Support Vector Machine), DBN-DNN(Deep Belief Network - Deep Neural Network)의 세 인식기가 고려됐다. 추가적으로, 화자 인식에 널리 사용되는 i-vector를 이용한 인식 실험도 수행했다. 인식 실험 결과, SPCC-SVM의 경우 98.81 %로 가장 높은 인식률을 확인 할 수 있었으며, 다른 알고리즘에서도 90 %에 가까운 인식률을 확인했다.

• 전자공학회논문지SC
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• 제42권3호
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• pp.33-40
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• 2005

심해저 자원의 확보를 위해서는 먼저 해양환경을 탐사하고 관측해야하며 이를 위해서 계측장비를 해저에 위치시키고 탐사가 끝난 후 회수하는 방법을 사용하는데 이 경우 계절변화에 따른 염분의 농도 및 온도 변화로 다양한 형태의 외란성 노이즈가 발생하여 제어 안정성과 수중통신에 대한 신뢰성이 떨어지고 있다. 그래서 기존의 제어방법은 수신된 제어정보를 하드웨어적인 방법으로 식별하고 기준 정보와 비교하며 이 과정을 수차례 반복하여 획득한 데이터를 제어정보로 활용하고 있다. 이는 제어의 신뢰성이 중요시 되는 시스템에서는 제어안정성 및 효율성이 떨어지고 있다. 따라서 본 논문에서는 해양환경 변화에 대한 제어안정성 및 동작신뢰성을 향상시킨 수중 탐측장비 회수용 원격 이탈제어 시스템을 제안하고자 한다. 그리고 제어알고리즘 및 원격 이탈제어 시스템의 적합성을 실험을 통하여 확인하였다.

• 한국멀티미디어학회논문지
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• 제21권4호
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• pp.450-463
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• 2018

Sound based machine fault diagnosis is the process consisting of detecting automatically the damages that affect the machines by analyzing the sounds they produce during their operating time. The collected sounds being inevitably corrupted by random disturbance, the most important part of the diagnosis consists of discovering the hidden elements inside the data that can reveal the faulty patterns. This paper presents a novel feature extraction methodology that combines various digital signal processing and pattern recognition methods for the analysis of the sounds produced by the drills. Using the Fourier analysis, the magnitude spectrum of the sounds are extracted, converted into two-dimensional vectors and uniformly normalized in such a way that they can be represented as 8-bit grayscale images. Histogram equalization is then performed over the obtained images in order to adjust their very poor contrast. The obtained contrast enhanced images will be used as the features of our diagnosis system. Finally, principal component analysis is performed over the image features for reducing their dimensions and a nonlinear classifier is adopted to produce the final response. Unlike the conventional features, the results demonstrate that the proposed feature extraction method manages to capture the hidden health patterns of the sound.

• 한국음향학회지
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• 제37권1호
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• pp.60-65
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• 2018

본 논문에서는 양서류 울음소리 구별을 CNN(Convolutional Neural Network)에 적용하기 위한 방법으로 공분산 행렬과 모듈로그램(modulogram)을 제안한다. 먼저, 멸종 위기 종을 포함한 양서류 9종의 울음소리를 자연 환경에서 추출하여 데이터베이스를 구축했다. 구축된 데이터를 CNN에 적용하기 위해서는 길이가 다른 음향신호를 정형화하는 과정이 필요하다. 음향신호를 정형화하기 위해서 분포에 대한 정보를 나타내는 공분산 행렬과 시간에 대한 변화를 내포하는 모듈로그램을 추출하여, CNN의 입력으로 사용했다. CNN은 convolutional layer와 fully-connected layer의 수를 변경해 가며 실험하였다. 추가적으로, CNN의 성능을 비교하기 위해 기존에 음향 신호 분석에서 쓰이는 알고리즘과 비교해보았다. 그 결과, convolutional layer가 fully-connected layer보다 성능에 큰 영향을 끼치는 것을 확인했다. 또한 CNN을 사용하였을 때 99.07 % 인식률로, 기존에 음향분석에 쓰이는 알고리즘 보다 높은 성능을 보인 것을 확인했다.