• Title/Summary/Keyword: Finite word length

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A Study on the Scaling in Wave Digital Filter (웨이브 디지털 필터의 스케일링에 관한 연구)

  • 권희훈;김명기
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.12 no.1
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    • pp.27-35
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    • 1987
  • Digital filter suffer from roundoff noise and adder overflows due to finite word length effects. Scaling is an attempt to internal signal levels such that all signals are as large as possible, yet without the occurrence of overflows. Scaling requirements are implemented by the use of transformer. This paper proposes a procedure for scaling wave digital filters to avoid overflow problems and at the same time maximizing the output signal-to-noise ratio. Results indicate that the scaled networks have an improved signal to noise ratio over th unscaled filters under the condition that there be no overflow occuring.

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Generalised Non Error-Accumulative Quantisation Algorithm with feedback loop

  • Koh, Kyoung-Chul;Choi, Byoung-Wook
    • 제어로봇시스템학회:학술대회논문집
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    • 2004.08a
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    • pp.1269-1274
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    • 2004
  • This paper presents a new quantisation algorithm which has the closed-loop form and guarantees the boundness of accumulative error. This algorithm is particularly useful for mobile robot navigation that is usually implemented on embedded systems. If wheel commands of the mobile robot are given by velocity or positional increment at every control instant and quantised due to finite word length of controller's CPU, the quantisation error gets accumulated to causes large position error. Such an error accumulative characteristic is fatal for non wheeled mobile robots or autonomous vehicles with non-holonomic constraint. To solve this problem, we propose a non-error accumulative quantisation algorithm with closed-loop form. We also show it can be extend to a generalized form corresponding to the n-th order accumulation. The boundness of the accumulative quantisation error is investigated by a series of computer simulation. The proposed method is particularly effective to precise navigation control the autonomous mobile robots.

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Architecture of 2-D DCT processor adopting accuracy comensator (정확도 보상기를 적용한 2차원 이산 코사인 변환 프로세서의 구조)

  • 김견수;장순화;김재호;손경식
    • Journal of the Korean Institute of Telematics and Electronics A
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    • v.33A no.10
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    • pp.168-176
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    • 1996
  • This paper presetns a 2-D DCT architecture adopting accurac y compensator for reducing the hardware complexity and increasing processing speed in VL\ulcornerSI implementation. In the application fields such as moving pictures experts group (MPEG) and joint photographic experts group (JPEG), 2-D DCT processor must be implemented precisely enough to meet the accuracy specifications of the ITU-T H.261. Almost all of 2-D DCT processors have been implemented using many multiplications and accumulations of matrices and vectors. The number of multiplications and accumulations seriously influence on comlexity and speed of 20D DCT processor. In 2-D DCT with fixed-point calculations, the computation bit width must be sufficiently large for the above accuracy specifications. It makes the reduction of hardware complexity hard. This paper proposes the accuracy compensator which compensates the accuracy of the finite word length calculation. 2-D DCT processor with the proposed accuracy compensator shows fairly reduced hardware complexity and improved processing speed.

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A Quantization Algorithm without Accumulative Error

  • Koh, Kyoung-Chul;Cho, Hyun-Suck
    • 제어로봇시스템학회:학술대회논문집
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    • 1999.10a
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    • pp.313-316
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    • 1999
  • In this paper, a quantization algorithm by which the accumulative error can be prevented is presented. In digital control systems, the quantization cannot be avoided because of the finite word length of digital computers. The error due to quantization of the computed values may be tolerable in case of directly using them. In case of using the accumulated values, the error between sum of the original values and that of the quantized values becomes larger as the number of the values to be summed increases. Such an increasing accumulative error is critical for the control of precise NC machines, robots and autonomous vehicles. To solve this problem, a quantization algorithm without the accumulative error is presented. Basically, the algorithm is based on the feedback loop by which the accumulationive of the quantization error can be prevented. The error boundness of the proposed algorithm is proven and a computer simulation is performed to show the validity of the algorithm.

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A Study on the Implementation of Connected-Digit Recognition System and Changes of its Performance (연결 숫자음 인식 시스템의 구현과 성능 변화)

  • Yun Young-Sun;Park Yoon-Sang;Chae Yi-Geun
    • MALSORI
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    • no.45
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    • pp.47-61
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    • 2003
  • In this paper, we consider the implementation of connected digit recognition system and the several approaches to improve its performance. To implement efficiently the fixed or variable length digit recognition system, finite state network (FSN) is required. We merge the word network algorithm that implements the FSN with one pass dynamic programming search algorithm that is used for general speech recognition system for fast search. To find the efficient modeling of digit recognition system, we perform some experiments along the various conditions to affect the performance and summarize the results.

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Sliding-DFT based multi-channel phase measurement FPGA system (Sliding-DFT를 이용한 다채널 위상 측정 FPGA 시스템)

  • Eo, Jin-Woo;Chang, Tae-Gyu
    • Journal of IKEEE
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    • v.8 no.1 s.14
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    • pp.128-135
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    • 2004
  • This paper proposes a phase measurement algorithm which is based on the recursive implementation of sliding-DFT. The algorithm is designed to have a robust behavior against the erroneous factors of frequency drift, additive noise, and twiddle factor approximation. The size of phase error caused by the finite wordlength implementation of DFT twiddle factors is shown significantly lower than that of magnitude error. The drastic reduction of the phase error is achieved by the exploitation of the quadruplet symmetry characteristics of the approximated twiddle factors in the complex plane. Four channel power-line phase measurement system is also designed and implemented based on the time-multiplexed sharing architecture of the proposed algorithm. The operation of the developed system is also verified by the experiment performed under the test environment implemented with the multi-channel function generator and the on-line interfaced host processor system. The proposed algorithm's features of phase measurement accuracy and its robustness against the finite wordlength effects can provide a significant impact especially for the ASIC or microprocessor based embedded system applications where the enhanced processing speed and implementation simplicity are crucial design considerations.

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A New Noise Reduction Method Based on Linear Prediction

  • Kawamura, Arata;Fujii, Kensaku;Itho, Yoshio;Fukui, Yutaka
    • Proceedings of the IEEK Conference
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    • 2000.07a
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    • pp.260-263
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    • 2000
  • A technique that uses linear prediction to achieve noise reduction in a voice signal which has been mixed with an ambient noise (Signal to Noise (S-N) ratio = about 0dB) is proposed. This noise reduction method which is based on the linear prediction estimates the voice spectrum while ignoring the spectrum of the noise. The performance of the noise reduction method is first examined using the transversal linear predictor filter. However, with this method there is deterioration in the tone quality of the predicted voice due to the low level of the S-N ratio. An additional processing circuit is then proposed so as to adjust the noise reduction circuit with an aim of improving the problem of tone deterioration. Next, we consider a practical application where the effects of round on errors arising from fixed-point computation has to be minimized. This minimization is achieved by using the lattice predictor filter which in comparison to the transversal type, is Down to be less sensitive to the round-off error associated with finite word length operations. Finally, we consider a practical application where noise reduction is necessary. In this noise reduction method, both the voice spectrum and the actual noise spectrum are estimated. Noise reduction is achieved by using the linear predictor filter which includes the control of the predictor filter coefficient’s update.

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On the Finite-world-length Effects in fast DCT Algorithms (고속DCT변환 방식의 정수형 연산에 관한 연구)

  • 전준현;고종석;김성대;김재균
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.12 no.4
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    • pp.309-324
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    • 1987
  • In recent years has been an increasing interest with respect to using the discrete cosine transform(DCT) of which performance is found close to that of the Karhumen-Loeve transform, known to be optimal in the area of digital image processing for tha purpose of the image data compression. Among most of reported algorithms aimed at lowering the coputation complexity. Chen's algorithm is is found to be most popular, Recently, Lee proposed a now algorithm of which the computational complexity is lower than that of Chen's. but its performance is significantly degraded by FWL(Finite-Word-Lenght) effects as a result of employinga a fixed-poing arithmetic. In this paper performance evaluation of these two algorithms and error analysis of FWL effect are described. Also a scaling technique which we call Up & Down-scaling is proposed to allevaiate a performance degradation due to fixed-point arithmetic. When the 16x16point 2DCT is applied on image data and a 16-bit fixed-point arithmetic is employed, both the analysis and simulation show that is colse to that of Chen's.

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Memory Organization for a Fuzzy Controller.

  • Jee, K.D.S.;Poluzzi, R.;Russo, B.
    • Proceedings of the Korean Institute of Intelligent Systems Conference
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    • 1993.06a
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    • pp.1041-1043
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    • 1993
  • Fuzzy logic based Control Theory has gained much interest in the industrial world, thanks to its ability to formalize and solve in a very natural way many problems that are very difficult to quantify at an analytical level. This paper shows a solution for treating membership function inside hardware circuits. The proposed hardware structure optimizes the memoried size by using particular form of the vectorial representation. The process of memorizing fuzzy sets, i.e. their membership function, has always been one of the more problematic issues for the hardware implementation, due to the quite large memory space that is needed. To simplify such an implementation, it is commonly [1,2,8,9,10,11] used to limit the membership functions either to those having triangular or trapezoidal shape, or pre-definite shape. These kinds of functions are able to cover a large spectrum of applications with a limited usage of memory, since they can be memorized by specifying very few parameters ( ight, base, critical points, etc.). This however results in a loss of computational power due to computation on the medium points. A solution to this problem is obtained by discretizing the universe of discourse U, i.e. by fixing a finite number of points and memorizing the value of the membership functions on such points [3,10,14,15]. Such a solution provides a satisfying computational speed, a very high precision of definitions and gives the users the opportunity to choose membership functions of any shape. However, a significant memory waste can as well be registered. It is indeed possible that for each of the given fuzzy sets many elements of the universe of discourse have a membership value equal to zero. It has also been noticed that almost in all cases common points among fuzzy sets, i.e. points with non null membership values are very few. More specifically, in many applications, for each element u of U, there exists at most three fuzzy sets for which the membership value is ot null [3,5,6,7,12,13]. Our proposal is based on such hypotheses. Moreover, we use a technique that even though it does not restrict the shapes of membership functions, it reduces strongly the computational time for the membership values and optimizes the function memorization. In figure 1 it is represented a term set whose characteristics are common for fuzzy controllers and to which we will refer in the following. The above term set has a universe of discourse with 128 elements (so to have a good resolution), 8 fuzzy sets that describe the term set, 32 levels of discretization for the membership values. Clearly, the number of bits necessary for the given specifications are 5 for 32 truth levels, 3 for 8 membership functions and 7 for 128 levels of resolution. The memory depth is given by the dimension of the universe of the discourse (128 in our case) and it will be represented by the memory rows. The length of a world of memory is defined by: Length = nem (dm(m)+dm(fm) Where: fm is the maximum number of non null values in every element of the universe of the discourse, dm(m) is the dimension of the values of the membership function m, dm(fm) is the dimension of the word to represent the index of the highest membership function. In our case then Length=24. The memory dimension is therefore 128*24 bits. If we had chosen to memorize all values of the membership functions we would have needed to memorize on each memory row the membership value of each element. Fuzzy sets word dimension is 8*5 bits. Therefore, the dimension of the memory would have been 128*40 bits. Coherently with our hypothesis, in fig. 1 each element of universe of the discourse has a non null membership value on at most three fuzzy sets. Focusing on the elements 32,64,96 of the universe of discourse, they will be memorized as follows: The computation of the rule weights is done by comparing those bits that represent the index of the membership function, with the word of the program memor . The output bus of the Program Memory (μCOD), is given as input a comparator (Combinatory Net). If the index is equal to the bus value then one of the non null weight derives from the rule and it is produced as output, otherwise the output is zero (fig. 2). It is clear, that the memory dimension of the antecedent is in this way reduced since only non null values are memorized. Moreover, the time performance of the system is equivalent to the performance of a system using vectorial memorization of all weights. The dimensioning of the word is influenced by some parameters of the input variable. The most important parameter is the maximum number membership functions (nfm) having a non null value in each element of the universe of discourse. From our study in the field of fuzzy system, we see that typically nfm 3 and there are at most 16 membership function. At any rate, such a value can be increased up to the physical dimensional limit of the antecedent memory. A less important role n the optimization process of the word dimension is played by the number of membership functions defined for each linguistic term. The table below shows the request word dimension as a function of such parameters and compares our proposed method with the method of vectorial memorization[10]. Summing up, the characteristics of our method are: Users are not restricted to membership functions with specific shapes. The number of the fuzzy sets and the resolution of the vertical axis have a very small influence in increasing memory space. Weight computations are done by combinatorial network and therefore the time performance of the system is equivalent to the one of the vectorial method. The number of non null membership values on any element of the universe of discourse is limited. Such a constraint is usually non very restrictive since many controllers obtain a good precision with only three non null weights. The method here briefly described has been adopted by our group in the design of an optimized version of the coprocessor described in [10].

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The Design of Optimal Filters in Vector-Quantized Subband Codecs (벡터양자화된 부대역 코덱에서 최적필터의 구현)

  • 지인호
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.1
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    • pp.97-102
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    • 2000
  • Subband coding is to divide the signal frequency band into a set of uncorrelated frequency bands by filtering and then to encode each of these subbands using a bit allocation rationale matched to the signal energy in that subband. The actual coding of the subband signal can be done using waveform encoding techniques such as PCM, DPCM and vector quantizer(VQ) in order to obtain higher data compression. Most researchers have focused on the error in the quantizer, but not on the overall reconstruction error and its dependence on the filter bank. This paper provides a thorough analysis of subband codecs and further development of optimum filter bank design using vector quantizer. We compute the mean squared reconstruction error(MSE) which depends on N the number of entries in each code book, k the length of each code word, and on the filter bank coefficients. We form this MSE measure in terms of the equivalent quantization model and find the optimum FIR filter coefficients for each channel in the M-band structure for a given bit rate, given filter length, and given input signal correlation model. Specific design examples are worked out for 4-tap filter in 2-band paraunitary filter bank structure. These optimum paraunitary filter coefficients are obtained by using Monte Carlo simulation. We expect that the results of this work could be contributed to study on the optimum design of subband codecs using vector quantizer.

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