• 한국음향학회지
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• 제33권5호
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• pp.335-340
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• 2014

본 논문에서는 충격성 잡음(impulsive noise) 환경 하에서 고유치 분포가 큰 입력 신호를 다루기 위해 제안된 부호 부밴드 적응 필터(Sign Subband Adaptive Filter, SSAF)의 성능을 향상시키기 위한 새로운 SSAF를 제안하였다. 기존에 제안된 SSAF는 각각의 부밴드 입력 신호를 모든 부밴드 입력 신호의 $l_2-norm$으로 정규화하기 때문에 밴드의 수를 증가시켜도 수렴속도가 향상되지 않는다. 본 논문에서는 부밴드 입력 신호를 각각의 부밴드 입력 신호의 $l_2-norm$으로 정규화하며 밴드의 수를 증가시킴에 따라 수렴속도가 증가하는 새로운 부호 부밴드 적응필터를 제안하였다. 시스템 식별 환경에서 두 알고리즘의 성능을 비교하는 컴퓨터 모의 실험을 수행하여 제안된 알고리즘의 수렴속도가 더 빠름을 보였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.102-105
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• 2000

This paper propose new sound localization algorithm that calculates TDOA(Time Difference Of Arrival) between the two received signals via two microphone array, The proposed Subband CPSP is a development of Previous CPSP method using subband approach. It first split the received microphone signals into three frequency bands and then calculates subband CPSP with corresponding SNR weights. This type of algorithm, Subband CPSP, can provide more accurate TDOA estimation results because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, computer simulation was conducted and it was compared with previous CPSP method. From the both simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than accuracy for TDOA estimation.

• The Journal of the Acoustical Society of Korea
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• 제20권4E호
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• pp.52-60
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• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1997년도 한국자동제어학술회의논문집; 한국전력공사 서울연수원; 17-18 Oct. 1997
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• pp.1730-1733
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• 1997

In this paper, a realization of a subband coding with TMS320C80 is studied. TMS320C80 is a multi-media processor specially designed for an image process. A main topic of this paper, as mentioned above, is an application of TMS320C80 to subband coding. Subband coding is the coding that devides full image to several subbands and encodes each subband with different coding methods. As using that methods, good image compression can be obtained. First above all, goal of this paper deals with TMS320C80 in coding still image and useds it in expending it's application to 3-D video coding.

• 전자공학회논문지S
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• 제35S권2호
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• pp.97-105
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• 1998

Subband/DCT coding has been introduced in order to transmit images of various resultions using one given image-codec, for nowadays there are various grades of quality in visual communication services. However, subband/DCT results in the increawse of multiplication number and memory size. In order to resolve this problem, we propose block-based subband/DCT coding in this paper. In block-based subband/DCT, the number of multiplications is not only reduced because we combine subband decomposistion with DCT, but the size of memory is also reduced because images can be parallel-processed block by block. We show that the number of multiplications is reduced, by analyzing the property ofblock-based subband/DCT matrix mathematically, and examine the performance of proposed coder, which adopts JPEG as backhand-coder after block-based subband/DCT.

• 한국음향학회지
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• 제20권1호
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• pp.36-42
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• 2001

본 연구는 폐쇄된 사무 공간상에서 2개의 마이크로폰을 이용하여 임의의 위치에서 발생한 음성 및 음향의 방향성 (방향각)을 추적하는 새로운 알고리듬을 제안한다. 본 논문에서 제안한 Subband CPSP (Cross Power Spectrum Phase) 알고리듬은 기존의 CPSP 알고리듬을 개선한 것으로서, 마이크로폰에 수신된 2개의 입력 신호에 대해 서브밴드 필터 뱅크를 이 용하여 대역 분할하고 각 서브밴드 대역에서 구해지는 대역별 CPSP 결과의 평균값을 제공한다. 이러한 주파수 대역 분할방식은 잡음의 영향을 각 대역으로 한정 분산시켜 사무 공간내 잡음의 영향을 각 대역으로 한정하여 음원의 방향각 계산시 발생하는 오차를 최소화할 수 있는 보다 강인하고 정확한 음원 추적 시스템을 가능하게 한다. 제안된 알고리듬의 성능을 입증하기 위해 기존의 CPSP 와 Subband CPSP 알고리듬의 실시간 음원 추적 실험을 수행하였으며, 실험 결과 제안된 Subband CPSP가 CPSP에 비해 평균 5％ 이상의 성능 향상을 가져옴을 확인할 수 있었다.

• 전자공학회논문지B
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• 제31B권4호
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• pp.88-96
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• 1994

In this paper, we propose a subband pyramid image coding scheme that uses ECVQ (ntropy Constrained Vector Quantizer). In subband pyramid image coding, each subband can be encoded with a coder matched to the statistics of that particular subband, and available versions of the original image at different resolution are easily obtained. ECVQ, aiming at the minimization of the distortion for a fixed entropy of the quantizer output, is well combined with the subband pyramid image coding which yields high compression ratio and good image quality. The optimum bit allocation to each subbands corresponds to the points where the individual distortion rate curves are of particular slope, weighted to the number of samples in that subband, in designing ECVQ.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 하계종합학술대회논문집
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• pp.677-680
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• 1998

In this paper, we proposed a lwo bit rate subband coding with adaptive vector quantization using the correlation between motion vector and block energy in subband. In this method, the difference between the input signal and the motion compensated interframe prediction signal is decomposed into several narrow bands using quadrature mirror filter (QMF) structure. The subband signals are then quantized by adaptive vector quantizers. In the codebook generating process, each classified region closer to the block value in the same region after the classification of region by the magnitude of motion vector and the variance values of subband block. Because codebook is genrated considering energy distribution of each region classified by motion vector and variance of subband block, this technique gives a very good visual quality at low bit rate coding.

• 한국통신학회논문지
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• 제21권5호
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• pp.1096-1108
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• 1996

In this paper, we propose a temporal adaptive tranform 3D SBC coder with motion compensation, exploiting redundancy in the temporal domain. We propose a temporal adaptivity measure, by which the R-D optimal temporal transform can be chaosen. The base temporal subband frame is coded using H.261-like MC-DCT coder, while the higher temporal subband frames are coded using the 2D adaptive wavelet packet bases, considering the various energy distribution which results from the temporal variation. In encoding the subbands, we employ adaptive scanning methods, uniform step-size quantization with VLC, and coded/not-coded flag reduction technique using the quadtree structure. From the simulation results, the proposed adaptive 3D subband coder shows about 0.29~3.14 dB gain over the H.261 and the fixed 3D subband coder techniques.