• 음성과학
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• 제12권3호
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• pp.125-136
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• 2005

Automatic speech recognition is much more difficult in real world. Speech recognition according to SIR (Signal to Interface Ratio) is difficult in situations in which noise of surrounding environment and multi-speaker exists. Therefore, study on main speaker's voice extractions a very important field in speech signal processing in binaural sound. In this paper, we used directional filter and harmonic filter among other existing methods to extract the main speaker's information in binaural sound. The main speaker's voice was extracted using directional filter, and other remaining speaker's information was removed using harmonic filter through main speaker's pitch detection. As a result, voice of the main speaker was enhanced.

• 한국농업기계학회:학술대회논문집
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• 한국농업기계학회 1998년도 하계 학술대회 논문집
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• pp.270-275
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• 1998

It was assumed that the voice of cattle is generated with the virtual white noise through the digital filter called the linear prediction filter, and filter parameters (prediction coefficients) were estimated by the maximum entropy method (MEM) , using the sound signal of the animal . The feature planes were defined by the pairs of two parameters selected appropriately from these parameters. The cattle voices were divided into three levels, that is the high, medium and low levels according to their total power equivalent to the variances of the sound signal . It was found that the straight lines could be used for recognizing tow cow and one calf for high level voices. For high and medium level voices, however, it was difficult or impossible to recognize individual cattle on the parameters planes.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2006년도 하계 학술대회 논문집
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• pp.9-12
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• 2006

높은 컴퓨터 보급률에 따른 인터넷의 대중화로 인하여 새로운 소식을 원하는 사람들은 기존의 정해진 시각에 전달되는 지면 신문보다 인터넷을 통해 새로운 소식을 접하는 경향이 높아지면서, 국내의 각 언론사들은 RSS(RDF Site Summary)문서를 제공하기 시작하였다. 차세대 웹인 시맨틱 웹의 여러 가지 규격 및 기술 중에서도 그 유용함과 편리성을 인정받아 우리 생활에 가장 먼저 적용되고 있는 RSS는 컨텐츠 배급을 위해 나온 XML형태의 규격 중 하나로서 웹사이트에서 사용자가 원하는 정보의 갱신된 내용을 신속하게 사용자에게 전달하는 자동 정보 수집 기술이다. 본 논문에서는 특정 언론사에서 제공하는 RSS문서에 음성인식 및 합성기술을 기반으로 동작하는 다른 XML형태의 규격인 음성 확장성 생성 언어(VoiceXML)를 접목하여 휴대전화 및 유선전화로 새로운 뉴스를 접할 수 있는 서비스를 제안하고 구현하였다. 실험 결과, 시간과 장소에 구애받지 않고 신뢰성 있는 언론사의 새로운 뉴스를 실시간으로 전달받을 수 있음을 확인하였다.

• 음성과학
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• 제10권4호
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• pp.107-115
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• 2003

In this paper we tried to classify the pathological voice signal with severe noise component based on two different parameters, the spectral slope and the ratio of energies in the harmonic and noise components (HNR), The spectral slope is obtained by using a curve fitting method and the HNR is computed in cepstrum quefrency domain. Speech data from normal peoples and patients are collected, diagnosed and divided into three different classes (normal, relatively less noisy and severely noisy data), The mean values and the standard deviations of the spectral slope and the HNR are computed and compared with in the three kinds of data to characterize and classify the severely noisy pathological voice signals from others.

• 전기학회논문지
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• 제59권2호
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• pp.445-451
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• 2010

This paper presents a threshold adaptation based voice query transcription scheme for music information retrieval. The proposed scheme analyzes monophonic voice signal and generates its transcription for diverse music retrieval applications. For accurate transcription, we propose several advanced features including (i) Energetic Feature eXtractor (EFX) for onset, peak, and transient area detection; (ii) Modified Windowed Average Energy (MWAE) for defining multiple small but coherent windows with local threshold values as offset detector; and finally (iii) Circular Average Magnitude Difference Function (CAMDF) for accurate acquisition of fundamental frequency (F0) of each frame. In order to evaluate the performance of our proposed scheme, we implemented a prototype music transcription system called AMT2 (Automatic Music Transcriber version 2) and carried out various experiments. In the experiment, we used QBSH corpus [1], adapted in MIREX 2006 contest data set. Experimental result shows that our proposed scheme can improve the transcription performance.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2005년도 학술대회 논문집 정보 및 제어부문
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• pp.201-203
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• 2005

The best method to study language is to talking with a native speaker. A voice recognition technology can be used to develope a language study machine. SD(Speaker dependant) and SI(speaker independant) voice recognition method is used for the language study machine. MP3 Player. FM Radio. Alarm clock functions are added to enhance the value of the product. The machine is designed with a DSP(Digital Signal Processing) chip for voice recognition. MP3 encoder/decoder chip. FM tumer and SD flash memory card. This paper deals with the application of SD ad SD voice recognition. flash memory file system. PC download function using USB ports, English conversation text function by the use of SD flash memory. LCD display control. MP3 encoding and decoding, etc. The study contents are saved in SD flash memory. This machine can be helpful from child to adult by changing the SD flash memory.

• 융합신호처리학회논문지
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• 제9권2호
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• pp.121-128
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• 2008

정확한 음성 구간 검출은 음성 인식 및 음성 코딩 그리고 음성 통신 시스템 등과 같은 음성 어플리케이션의 성능에 큰 영향을 미친다. 본 논문에서는 실제 운전하고 있는 상태에서 다양한 차량 노이즈 환경의 음성 구간 검출 방법을 제안한다. 기존의 음성 구간 검출은 시간 에너지, 주파수 에너지, 영 교차율, spectral entropy 등 다양한 방법을 사용하였으며 잡음 환경에서 급격하게 성능이 저하되는 단점이 있었다. 본 논문에서는 기존의 spectral entropy를 기반으로 하여 MFB(Mel-frequency Filter Banks) spectral entropy, 기울기 FFT(Fast Fourier Transform) spectral entropy, 기울기 MFB spectral entropy를 이용한 음성 구간 검출 방법을 제안한다. MFB는 멜 스케일과 FFT를 곱한 것으로 멜 스케일은 인간이 소리를 인지할 때 주파수에 대해 비선형적인 스케일이며 음성의 특징을 잘 반영한다. 제안한 MFB spectral entropy 방법은 다양한 차량 잡음 환경에서 음성 및 비음성 분별 능력을 향상시킬 수 있으며 실험 결과 93.21%의 음성 구간 검출율을 나타내었다. 이는 기존의 spectral entropy 방법과 비교할 때 MFB를 이용한 음성 구간 검출 방법이 3.2%의 검출율이 향상되었다.

• International Journal of Railway
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• 제3권2호
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• pp.39-45
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• 2010

The RF-CBTC (Radio Frequency-Communication Based Train Control) System is a communication system in railroad systems. The communication method of RF-CBTC system is the wireless between the wayside device and on-board device. The wayside device collects its location and speed from each train and transmits the distance from the forwarding train to the speed-limit position to it. The on-board device controlling device controls the speed optimum for the train. In the case of the RF-CBTC system used in Korea, transmission frequency is 2.4 [GHz]. It is the range of ISM(Industrial Scientific and Medical equipment) band and transmission of voice and data is performed by CDMA (Code Division Multiple Access) method. So noises are made in the AWGN (Additive White Gaussian Noise) and fading environment. Currently, the SNR (Signal to Noise Ratio) is about 20 [dB], so due to bit errors made by noises, transmission of reliable information to the train is not easy. Also, in the case that two tracks are put to a single direction, it is needed that two trains transmit reliable voice and data to a wayside device. But, by noises, it is not easy that just a train transmits reliable information. In this paper, we estimated the BER (Bit Error Rate) related to the SNR of voice and data transmission in the environment such as AWGN and fading from the RF-CBTC system using the CDMA method. Also, we supposed the SNR which is required to meet the BER standard for voice and data transmission. By increasing the processing gain that is a ratio of chip transmission to voice and data transmission, we made possible voice and data transmission from maximally two trains to a wayside device, and demonstrated it by using Matlab program.

• 음성과학
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• 제12권4호
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• pp.71-76
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• 2005

There have been numerous studies on the enhancement of noisy speech signal. In this paper, we propose a new speech enhancement method, that is, a filtering of a dissonant frequency combined with noise suppression algorithm. The simulation results indicate that the proposed method provides a significant gain in voice clarity. Therefore if the proposed enhancement scheme is used as a pre-filter, the perceptual clarity of husky voice is greatly enhanced.

• 한국산업융합학회 논문집
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• 제22권2호
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• pp.137-146
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• 2019

This study proposed a new technique to control of mobile robot based on voice command for (Human-Robot Cooperative operation in manufacturing precess). High performance voice recognition and control system was designed In this paper for smart factory. robust voice recognition is essential for a robot to communicate with people. One of the main problems with voice recognition robots is that robots inevitably effects real environment including with noises. The noise is captured with strong power by the microphones, because the noise sources are closed to the microphones. The signal-to-noise ratio of input voice becomes quite low. However, it is possible to estimate the noise by using information on the robot's own motions and postures, because a type of motion/gesture produces almost the same pattern of noise every time it is performed. In this paper, we describe an robust voice recognition system which can robustly recognize voice by adults and students in noisy environments. It is illustrated by experiments the voice recognition performance of mobile robot placed in a real noisy environment.