• 대한전기학회:학술대회논문집
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• 대한전기학회 2005년도 학술대회 논문집 정보 및 제어부문
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• pp.551-553
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• 2005

When a hearing aid' s user is listening through telephone or cellular phone, he/she usually suffers from severe electrical magnetic interference noise. It is because hearing aids amplify voice signal as well as background noise. A telecoil, an induction coil, is a possible solution for the problem. Because a telecoil has the characteristic of high pass filter, it has some problem of resulting increased high frequency noise. For solving this problem, we can use a capacitor connected with the telecoil in parallel. According to capacitance, receiving signal quality may change. In this paper, proper capacitor values for the best sound quality are investigated by experimental work.

• 전자공학회논문지B
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• 제30B권2호
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• pp.51-58
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• 1993

This paper describes the development of a speaker-dependent isolated word recognizer as applied to voice dialing in a car noise environment. for this purpose, several methods to improve performance under such condition are evaluated using database collected in a small car moving at 100km/h The main features of the recognizer are as follow: The endpoint detection error can be reduced by using the magnitude of the signal which is inverse filtered by the AR model of the background noise, and it can be compensated by using variants of the DTW algorithm. To remove the noise, an autocorrelation subtraction method is used with the constraint that residual energy obtainable by linear predictive analysis should be positive. By using the noise rubust distance measure, distortion of the feature vector is minimized. The speech recognizer is implemented using the Motorola DSP56001(24-bit general purpose digital signal processor). The recognition database is composed of 50 Korean names spoken by 3 male speakers. The recognition error rate of the system is reduced to 4.3% using a single reference pattern for each word and 1.5% using 2 reference patterns for each word.

• 한국인쇄학회지
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• 제23권2호
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• pp.15-23
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• 2005

This analog communication system overcomes a limit of the digital communication that used an electric power line, and shows a strong characteristic by decrease of impedance along a load or surge-voltage and the same noise. this system is to detect an sound signal added to light through an photo-sensor and a filter circuit. And a signal detected in this way is transmitted to sound through a speaker of an earphone again.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2000년도 하계학술대회 논문집 D
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• pp.3028-3030
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• 2000

In general. a speech coding is dedicated to the compression performance or the speech quality. But. the speech coding in this paper is focused on the performance of flexible transmission to the, network speed. For this. the subbanding coding is needed. which is used the wavelet packet concept in the signal analysis. The extraction of each frequency-band is difficult to general signal analysis methods, after coding each band, the reconstruction of these is also a difficult problem. But. with the wavelet packet concept(perfect reconstruction) and its fast computation algorithm. the extraction of each band and the reconstruction are more natural. Also, this paper describes a direct solution of the voice transmission on network and implement this algorithm at the TCP/IP network environment of PC.

• 한국군사과학기술학회지
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• 제13권3호
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• pp.388-396
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• 2010

Generally, flight information is transmitted by voice signal over legacy UHF radio in ground to air communication system. In this paper, we have implemented the transceiver of TDL(tactical data link) which transmits tactical information, such as flight information, using digital signal. For transmitting digital information over radio path, we have designed data modem that is processing CPFSK modulation, and TDMA(Time Division Multiple Access) network for Synchronization among multi user(platform). By simulating aeronautical propagation modeling with the environment of Korea terrain, it is predicted the maximum performance of communication range of the transceiver. As result of the transceiver's aviational boarding test, it is proved that the transceiver of TDL over legacy UHF radio transmits and receives the tactical information in TDMA network within communication range of 160km.

• International Journal of Advanced Culture Technology
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• 제7권2호
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• pp.218-226
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• 2019

In modern society, people live in an environment with artificial or natural noise. Especially, the sound that corresponds to the artificial noise makes the noise itself and affects each other because many people live and work in the city. Sounds are generated by the activities and causes of various people, such as construction sites, aircraft, production machinery, or road traffic. These sounds are essential elements in human life and are recognized and judged by human auditory organs. Noise is a sound that you do not want to hear by subjective evaluation, and it is a loud sound that gives hearing damage or a sound that causes physical and mental harm. In this study, we introduce the method of stimulating the human hearing by finger vibration and explain the advantages of the proposed method in various kinds of a noise environment. And how to improve the sound quality to improve efficiency. In this paper, we propose a method to prevent the loss of hearing loss and the transmission of sound information based on proper signal to noise ratio when using portable IT equipment in various noise environments.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제16권3호
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• pp.1076-1094
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• 2022

Technology for emotion recognition is an essential part of human personality analysis. To define human personality characteristics, the existing method used the survey method. However, there are many cases where communication cannot make without considering emotions. Hence, emotional recognition technology is an essential element for communication but has also been adopted in many other fields. A person's emotions are revealed in various ways, typically including facial, speech, and biometric responses. Therefore, various methods can recognize emotions, e.g., images, voice signals, and physiological signals. Physiological signals are measured with biological sensors and analyzed to identify emotions. This study employed two sensor types. First, the existing method, the binary arousal-valence method, was subdivided into four levels to classify emotions in more detail. Then, based on the current techniques classified as High/Low, the model was further subdivided into multi-levels. Finally, signal characteristics were extracted using a 1-D Convolution Neural Network (CNN) and classified sixteen feelings. Although CNN was used to learn images in 2D, sensor data in 1D was used as the input in this paper. Finally, the proposed emotional recognition system was evaluated by measuring actual sensors.

• 한국전자통신학회논문지
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• 제9권4호
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• pp.491-498
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• 2014

무선 접속을 통한 인터넷 서비스가 보편화된 최근 사람이 서비스를 이용하는 과정에서 실제로 느끼는 품질인 사용자 체감 품질(QoE; Quality of Experience)의 중요성이 더욱 강조되고 있는데 사용자 체감 품질은 서비스 품질 (QoS; Quality of Service)와 같이 객관적인 수치화가 불가능하다는 특징이 있다. 유선과는 달리 무선 접속을 통해서 제공되는 IP 서비스는 매우 많은 요인에 의해서 사용자 체감 품질이 영향을 받을 수 있다. 따라서 본 논문에서는 대표적인 무선 접속 서비스인 HSPA (High Speed Packet Access)를 통해서 음성 서비스가 제공될 때 측정 가능한 품질지표를 선정하고 이들에 대한 실측값을 통계적으로 분석하여 서비스 품질과 사용자 체감 품질 지표와의 상관관계를 밝히고자 한다. 분석 결과 RSSI (Received Signal Strength Indicator)와 전송 지연의 상관관계가 매우 높고 그에 이어 전송 지연과 MOS (Mean Opinion Score)와 매우 높은 상관관계를 가짐을 알 수 있었다.

• 한국음향학회지
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• 제22권4호
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• pp.276-285
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• 2003

본 논문에서는 고성능 DSP (Digital Signal Processor)를 사용하여 R2MFC/DTMF (R2 Multi Frequency Combinations/Dual Tone Multiple Frequency) 톤 송수신기를 실시간 구현하여 대용량 VoIP (Voice over Internet Protocol) 게이트웨이 시스템에 적용한다. 수신기는 Goertzel 필터를, 송출기는 고조파 공명 필터를 이용한다. DMA (Direct Memory Access)와 McBSP(Multi Channel Buffered Serial Port)를 사용한 효과적인 PCM 입출력, HPI (Host Port Interface)를 통한 MPU (Main Processing Unit)와의 메시지 통신 등 Texas Instruments TMS320C62x DSP를 이용한 다채널 실시간 구현 기법에 관하여 상세히 기술한다. 실험 결과, 구현된 R2MFC/DTMF 송수신기는 ITU-T(International Telecommunication Union-Telecommunication) 조건을 만족하며, 최적화 된 코드는 250 ㎒ C62x에서 780 채널을 수용할 수 있는 계산량을 보였다.

• 한국음향학회지
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• 제22권4호
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• pp.286-292
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• 2003

음성 구간 추출이란 입력된 음성신호를 음성 구간과 묵음, 또는 잡음구간으로 구분하는 과정이다. 잡음이 섞여있는 음성신호의 무성자음 신호는 잡음신호와 매우 유사하다. 따라서 음성 구간을 추출하거나 잡음을 제거 또는 감소시킬 때 무성자음에 특별히 주의하지 않으면 무성자음을 손상시키거나 잘못된 잡음 추정으로 이어질 수 있다. 본 논문에서는 잡음 환경에서 연속음성신호의 음성 구간을 정확하게 추출하기 위해 잡음과 무성자음사이의 경계를 명시적으로 검출함으로써 무성자음의 구간을 추출하는 방법을 제안한다. 제안하는 추출방법은 Hirsch가 잡음 추정을 위해 사용한 히스토그램 방법과 연속된 프레임 사이의 주파수 성분의 유사성을 나타내는 파라미터들을 이용하였다. 제안한 방법의 성능을 평가하기 위해 음성신호에 SNR이 각각 10㏈와 15㏈인 7가지의 잡음을 첨가하여 무성자음신호의 추출 실험을 수행하였다.