• International journal of advanced smart convergence
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• v.7 no.4
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• pp.92-100
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• 2018

This paper presents a link adaptation scheme for adaptive full duplex (AFD) systems. The signal modulation levels and communication link patterns are adaptively selected according to the changing channel conditions. The link pattern selection process consists of two successive steps such as a transmit-receive antenna pair selection based on maximum sum rate or minimum maximum symbol error rate, and an adaptive modulation based on maximum minimum norm. In AFD systems, the antennas of both nodes are jointly determined with modulation levels depending on the channel conditions. An adaptive algorithm with relatively low complexity is also proposed to select the link parameters. Simulation results show that the proposed AFD system offers significant bit error rate (BER) performance improvement compared with conventional full duplex systems with perfect or imperfect self-interference cancellation under the same fixed sum rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Transactions of the Korean Society of Mechanical Engineers
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• v.18 no.10
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• pp.2523-2532
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• 1994

In this paper, active noise control is performed in a duct system using the periodic pulse train which corresponds to the periodic component of noise source as a reference signal. Control algorithm applied in this study is possible to eliminate the acoustic feedback which occurs in the conventional filtered-x and filtered-u LMS algorithm by using electrical reference signal and has the fast adaptation speed with low filter orders by using synchronous sampling method is discussed via computer simulations and experiments of case studies such as frequency modulation, amplitude modulation and frequency differency between source signal and reference signal.

• Journal of the Korean Institute of Telematics and Electronics D
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• v.34D no.11
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• pp.1-8
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• 1997

In this paper, we propose a new method for the improvement of linearizing and adaptive convergence charateristics of the feedforward linear power amplifier. In this circuit, errors at the signal cancellation stage can be compensated at the error cancellation stage and the overall linearizaton and adaptation characteristics of the linear amplifier are improved. The broadband characteristics and linearizing capability are improved without increasing the complexity of circuits and the signal processing structure.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.8C
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• pp.801-805
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• 2006

A normalization approach to coefficient adaptation in the escalator(ESC) filter structure that conventionally employs least mean square(LMS) algorithm is introduced. Using Taylor's expansion of the local error signal, a normalized form of the ESC-LMS algorithm is derived. Compared with the computational complexity of the conventional ESC-LMS algorithm employs input power estimation for time-varying convergence coefficient using a single-pole low-pass filter, the computational complexity of the proposed method can be reduced by 50% without performance degradation.

• Proceedings of the Korean Biophysical Society Conference
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• 1999.06a
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• pp.20-20
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• 1999

Bacterial chemotaxis receptors are some of the simplest and most studied transmembrane receptors. Their simple signaling pathway has elements relevant for understanding the mechanisms for signal recognition, transduction through the membrane, relays among the molecules in the pathway, and adaptation to a persistent signal.(omitted)

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.46-49
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• 2003

The use of confidence measures for word/utterance verification has become art essential component of any speech input application. Confidence measures have applications to a number of problems such as rejection of incorrect hypotheses, speaker adaptation, or adaptive modification of the hypothesis score during search in continuous speech recognition. In this paper, we present a new utterance verification method using vowel string. Using subword HMMs of VCCV unit, we create anti-models which include vowel string in hypothesis words. The experiment results show that the utterance verification rate of the proposed method is about 79.5%.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.82-89
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• 2004

This paper describes how to reduce the effect of an occupation threshold by that the transform of mixture components of HMM parameters is controlled in hierarchical tree structure to prevent from over-adaptation. To reduce correlations between data elements and to remove elements with less variance, we employ PCA (principal component analysis) and ICA (independent component analysis) that would give as good a representation as possible, and decline the effect of over-adaptation. When we set lower occupation threshold and increase the number of transformation function, ordinary WLLR adaptation algorithm represents lower recognition rate than SI models, whereas the proposed MLLR adaptation algorithm represents the improvement of over 2% for the word recognition rate as compared to performance of SI models.

• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.520-529
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• 2011

In this paper, we propose the extension of the MPEG-2 Transport Stream (TS) header for efficient adaptation of multi-layer coded video such as scalable video coding (SVC) and multiview video coding (MVC) in the HTTP streaming. First of all, the limit of the existing TS in terms of flexible adaptation of multi-layer video is investigated, and the signaling by extending TS header is proposed to provide efficient adaptation in a TS level. The proposed extension utilizes the private data field in the adaptation field of TS header to signal scalability and/or view information, which enable us to support diverse adaptation that suits underlying constraints of client capabilities, network conditions and user preferences. In short, the extension enables adaptation of scalable video with full scalability as well as view selection of multiview video in a TS level while keeping backward compatibility with the existing TS syntax/semantics. The performance of the proposed extension is compared with the existing adaptation using PID (packet ID) in terms of efficiency and complexity of adaptation. Furthermore, the increase of TS overhead caused by proposed extension is analyzed and an extension scheme to minimized the overhead is proposed.

• 제어로봇시스템학회:학술대회논문집
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• 2002.10a
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• pp.95.6-95
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• 2002

1. Introduction 2. Basic Concepts 2.1 Signal Model 2.2. Least-Mean-Square Adaptation Algorithm 3. Minimum Mean-Square Error 4. Bayesian Beamformer References