• Title/Summary/Keyword: 신호 천이

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Detection of Underwater Transient Signals Using Noise Suppression Module of EVRC Speech Codec (EVRC 음성부호화기의 잡음억제단을 이용한 수중 천이신호 검출)

  • Kim, Tae-Hwan;Bae, Keun-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.6
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    • pp.301-305
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    • 2007
  • In this paper, we propose a simple algorithm for detecting underwater transient signals on the fact that the frequency range of underwater transient signals is similar to audio frequency. For this, we use a preprocessing module of EVRC speech codec that is the standard speech codec of the mobile communications. If a signal is entered into EVRC noise suppression module, we can get some parameters such as the update flag, the energy of each channel, the noise suppressed signal, the energy of input signal, the energy of background noise, and the energy of enhanced signal. Therefore the energy of the enhanced signal that is normalized with the energy of the background noise is compared with the pre-defined detection threshold, and then we can detect the transient signal. And the detection threshold is updated using the previous value in the noisy period. The experimental result shows that the proposed algorithm has $0{\sim}4% error rate in the AWGN or the colored noise environment.

Signal transient simulation of multi-coupledm frequency-variant transmission lines (주파수 종속 다중 전송선의 신호 천이 특성)

  • Cho, Young-Il;Eo, Yung-Seon
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.43 no.12 s.354
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    • pp.89-101
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    • 2006
  • Frequency-variant transmission line parameters are determined. Then the signal transient characterizations of frequency-dependent multi-coupled lines are investigated. With the proposed method, an accurate signal integrity degradation such as signal ringing (overshoot, undershoot) and crosstalk noises relevant to the switching patterns of signals, rising / falling time(tr, tf) and line lengths is investigated. It is shown that there may be approximately 26% discrepancy of signal transients and 260% difference of crosstalk noises between the constant RLC model and frequency-variant RLC model in on-chip global interconnects while those of package lines are 11% and 70%, respectively.

Angle Difference Based State Transition Modeling Technique for the Classification of Signal Pattern from the Sensor Array (센서 어레이의 신호패턴 분류를 위한 각도 변이 기반 상태 천이 모델링 기법)

  • Kim, A-Ram;Lee, Seung-Jae;Kim, Sung-Kyung;Park, Soo-Hyun;Kim, Chang-Hwa
    • Journal of the Korea Society for Simulation
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    • v.15 no.3
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    • pp.49-60
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    • 2006
  • We propose a method to use a state transition model so that the sensing object can be distinguished through classification of signal patterns sensed by a sensor array. Focusing on the design of the model that is able to distinguish the sensed object more exactly, we present an idea in which the modeling elements, 'states' and 'transitions' are defined as each same-sized angle intervals into which the angle interval $(-\frac{\pi}{2},\frac{\pi}{2})$ is divided and the angle differences between adjacent signal values on sampling signal value sequence value sequence sensed from the sensor array in the uniform time interval, respectively. In addition we show the usefulness of our model through experiments.

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Noise Shaping effects of Signal-blurred DPCM (신호 몽롱화 예측부호화의 잡음 천이 특성)

  • 황재정;이문호
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.16 no.10
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    • pp.964-971
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    • 1991
  • In DPCM coder which has flat notse distributions, reconstruction noises might be shaped or transferred to hign frequency portion using the relevant noise reduction loop under the constraints of image compression. Noise shaping filter directly in loop can be used, but we shows the complete characteristics of signal blurred DPCM coder in view of data compression and noise processing. Equi weighting in bluming procedure of prefilter is introduced and the blurred signal is restored by debiurning postfilter. Noise shaping tegion on two dimensional frequency domain depends upon the size of masking filter. In spite of its noise increments by the signal blurring, the algonthm is effective for the visually relevant coding which has low pass freqency properties.

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Feature Vector Extraction Method for Transient Sonar Signals Using PR-QMF Wavelet Transform (PR-QMF Wavelet Transform을 이용한 천이 수중 신호의 특징벡타 추출 기법)

  • Jung, Yong-Min;Choi, Jong-Ho;Cho, Yong-Soo;Oh, Won-Tcheon
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.1
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    • pp.87-92
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    • 1996
  • Transient signals in underwater show several characterisrics, that is, short duration, strong nonstationarity, various types of transient sources, which make it difficult to analyze and classify transient signals. In this paper, the feature vector extraction method for transient SOMAR signals is discussed by applying digital signal processing methods to the analysis of transient signals. A feature vector extraction methods using wavelet transform, which enable us to obtain better recognition rate than automatic classification using the classical method, are proposed. It is confirmed by simulation that the proposed method using wavelet transform performs better than the classical method even with smaller number of feature vectors. Especially, the feature vector extraction method using PR-QMF wavelet transform with the Daubechies coefficients is shown to perform well in noisy environment with easy implementation.

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Feature Extraction and Classification of Underwater Transient Signal using MFCC and Wavelet Packet Based on Entropy (MFCC과 엔트로피 기반의 웨이블릿 패킷 기법을 이용한 수중 천이신호의 특징추출 및 식별)

  • Jung, Jae-Gun;Park, Jeong-Hyun;Kim, Dong-Wook;Hwang, Chan-Sik
    • Proceedings of the KAIS Fall Conference
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    • 2009.05a
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    • pp.781-784
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    • 2009
  • 본 논문에서는 실제 수중 환경에서 선박 또는 잠수함으로부터 발생하는 인위적인 천이신호와 돌고래, 새우 등의 해양 생물로부터 발생하는 천이신호들을 식별하기 위한 특징벡터 추출 기법을 제안하였다. MFCC와 엔트로피 기반의 웨이블릿 패킷 기법을 이용하여 특징을 추출하고, 이 두 특징들을 동시에 적용하여 수중 천이신호를 식별하고자 한다. 기존의 방법인 MFCC와 웨이블릿 패킷 기법과 이 두 방법을 동시에 적용했을 때의 식별률을 비교하였고, 전방향 신경회로망(feed-forward neural network)을 그 특징벡터의 성능을 평가하기 위한 식별기로 사용하였다.

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A Study on the Conversion Algorithm and its Implementation Technique Between NAS and CEPT Channel Associated Signalling (이종 채널 결합 신호 방식간 변환을 위한 알고리즘 및 실현 기법에 관한 연구)

  • 김춘희;염흥열;김재근;김재명
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.14 no.4
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    • pp.412-422
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    • 1989
  • The need to conversion between NAS and CEPT networks has led to the development of PCM standards conversion techniques, especially signalling conversion techniques. This paper contains a brief description and general overview of channel associated signalling(CAS) of 24-channel system and of 30-channel system. The line signalling conversion algorithm between two systems and the transition detection algorithm of call status is proposed. And this paper presents the effective inplementation techniques of these algorithm.

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Blind Equalization Using a Decision-Directed Algorithm for Partial Response Signals (부분응답신호에서 'Stop-and-Go' 알고리듬을 이용한 블라인드 적응 등화)

  • 강민구;이영조;윤영우;홍대식;강창언
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.19 no.4
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    • pp.597-604
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    • 1994
  • In this paper, the "Stop-and-GO" algorithm is applied to the PRS(Partial Response Signal) type signaling. Stop-and-GO blind equalizer has a property of stopping the adaptation of its tab coefficients by means of a simples flag telling the equalizer whether the current output error with respect to decided symbol is sufficiently reliable to be used. PRS has the rule os level transition, which makes it possible that the level of currently received symbol is in the adjacent levels of the previously received symbol. New nonlinear estimators for PRS, based on the rule of level transition, is proposed. The computer simulation results show the improvement in performance achievable with proposed nonlinear estimators.stimators.

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Sinusoidal Modeling of Polyphonic Audio Signals Using Dynamic Segmentation Method (동적 세그멘테이션을 이용한 폴리포닉 오디오 신호의 정현파 모델링)

  • 장호근;박주성
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.4
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    • pp.58-68
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    • 2000
  • This paper proposes a sinusoidal modeling of polyphonic audio signals. Sinusoidal modeling which has been applied well to speech and monophonic signals cannot be applied directly to polyphonic signals because a window size for sinusoidal analysis cannot be determined over the entire signal. In addition, for high quality synthesized signal transient parts like attacks should be preserved which determines timbre of musical instrument. In this paper, a multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in sinusoidal modeling a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. The improved dynamic segmentation is proposed which shows better performance about transients and reduced computation. For various polyphonic audio signals the result of simulation shows the suggested sinusoidal modeling can model polyphonic audio signals without loss of perceptual quality.

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Improvement of Detection Performance of a Tag response signal in ISO/IEC 18000-6 Type-B Readers (ISO/IEC 18000-6 Type-B RFID 리더의 태그 응답신호 검출 성능 향상)

  • Choi, Woo-Seok;Suh, Ki-Hwan;Kim, Hyoung-Nam
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.1C
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    • pp.59-66
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    • 2007
  • This paper proposes a windowing method to reduce noise effect and bit synchronization error caused by tolerance of a tag-response signal for ISO/IEC 18000-6 type-B readers. A tag response signal has very weak power because of a back-scattering modulation scheme and thus may be sensitive to noise. In addition, bit tolerance admitted in a tag response signal requires robust timing synchronization because it affects readers' detection performance. To reduce the two undesirable effects in a tag signal, we acquire bit transition position by using variable windows from over-sampled data, and average whole data in one bit duration. With a hardware system adopting the proposed method, we tested and verified its performance.