• Journal of Sensor Science and Technology
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• v.15 no.4
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• pp.223-230
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• 2006

An FIR filter for digital voice signal processing has been designed and implemented using a microcontroller in implantable middle ear hearing devices (IMEHDs). The designed digital voice signal processing filter which has fast and accurate filtering operation and controllable filter characteristics has been implemented using a hardware multiplier and a direct memory access (DMA) in the low power microcontroller, MSP430F169. It has been confirmed that each of the implemented 6-orders Remez FIR filters with 1 channel and 2 channels can be applied to the voice signal processing module of IMEHDs based on the evaluation results of the filtering performance experiment.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.2
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• pp.145-151
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• 2010

This paper proposes a reverse filtering system of sound field obtaining a state of sound field transmitted from two sounds, using an adaptive filter and neural network. The proposed system uses the reverse filtering method with calculating and renewing a coefficient of a filter, using least mean square. Based on training the neural network, experiments confirm that the proposed system is effective for a simple waveform with non-linear distortion, by using neural network and adaptive filtering method.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.695-698
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• 2009

본 논문에서는 음원으로부터 나온 음과 동일한 음을 들을 수 있는 시스템을 구축하는 것을 목적으로 하여 이 두 개의 음으로부터 전달되어온 음장의 상태를 구하여 이 역 필터를 구성하는 방법을 연구한다. 본 논문에서는 최소 2승 평균법(Least Mean Square, LMS)을 사용하여 FIR 필터(Finite Impulse Response)의 계수를 계산하여 이를 갱신함으로써 역 필터법을 구축하는 방법을 사용한다. 또한 이 방법과는 별도로 LMS법의 부분을 신경회로망에 대처하는 알고리즘을 제안하였다. 시뮬레이션 실험으로부터 상당히 간단한 파형에 비선형인 왜곡이 있는 것을 본 논문에서 제안한 신경회로망에 의한 학습 가능한 것을 확인하였다.

• Journal of KSNVE
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• v.4 no.2
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• pp.187-198
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• 1994

This study is to describe the effects of the duct termination conditions conditions upon the active noise attenuation system. The adaptive filtering algorithm using FIR filter is implemented for duct noise attenuation. To avoid the instability caused by the acoustic feedback, two methods are considered. One is to use a compensating FIR filter. The other is to utilize uni-directional detecting microphone and uni-directional control speaker. Experimental results show that the reflections of sound from duct terminations greatly reduce the performance of ANC system. The directionality of detecting microphone and control speaker is a major factor to decide ANC performance. When there are some reflections from both duct terminations, the noise attenuation using finite FIR filter is not enough to model the duct plant. Especially, the reflection from the upstream termination reduces the noise attenuation in the frequencies related to the distance between control speaker and upstream termination. The performance of the noise attenuation is found to be largely enhanced by using uni-directional coupler, both on detecting microphone and control speaker, even if the duct system has an arbitrary termination conditions.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.11
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• pp.2249-2254
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• 1994

An observation which is useful for hardware implementation of programmable FIR filters with powers-of two coefficients (2PFIR filters) is made. Specifically, it is shown that the exponents of filter coefficients representable by the canonical signed digit(CSD) code with M ternary digits can be chosen from some subsets of {0, 1, $\cdots$, M-1}. This observation naturally leads to 2PFIR filters with shorter shifters whose length is strictly less than M and, as a consequence, leads to an efficient hardware structure fo programmable 2PFIR filtering. In addition, we present some experimental results indicating that the shifters of 2PFIR filters can be shortened further in some cases.

• The Journal of the Acoustical Society of Korea
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• v.11 no.2
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• pp.59-67
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• 1992

Recently, the problem which actively control the unwanted noise propagated from the technical structure by the generated secondary sound has become considerable topic from the environmental preservation point of view. In most of these studies, active noise control deals with a plane wave propagation at low frequency using adaptive filtering techniques. On the other hand, in real acoustic systems are mostly short due to the limitation of geometric configuration. In this case, the acoustic properties such as reflections and resonances inside the acoustic system should be considered. In this paper, the acoustic modeling method for short length duct was introduced using the transfer matrix method, and the active noise control problem was investigated with \implementation of FIR filter for the transfer function of control system derived from this modeling method. The identification methods for the acoustic model of actual control system was proposed by numerical computation technique based on the estimation of optimal FIR filter coefficients. The acceptable attenuation on the real acoustic system and stability of the controller are predicted in this computational simulation.

• Journal of Satellite, Information and Communications
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• v.5 no.1
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• pp.32-37
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• 2010

The algorithm of multiple channel signal processing requires the flexibility of variable frequency band, efficient allocation of transmission power, and flexible frequency band reallocation to satisfy various service types which requires different transmission rates and frequency band. There are three methods including per-channel approach, multiple tree approach, and block approach performing frequency band reallocation method by channelization and dechannelization in the multiple-channel signal. This paper proposes an improved per-channel approach for converting the frequency band of multiple carrier signals efficiently. The proposed algorithm performs decimation and interpolation using CIC(cascaded integrator comb filter), half-band filter, and FIR filter. In addition, it performs filtering of each sub-channel, and reallocates channel band through FIR low-pass filter in the multiple-channel signal. The computer simulation result shows that the perfect reconstruction of output signal and the flexible frequency band reallocation is performed efficiently by the proposed algorithm.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.28B no.5
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• pp.343-356
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• 1991

An efficient real-time implementation of digital filtering algorithms using a multiprocessor system in a ring network is investigated. The development time and cost for implementing a high speed signal processing system can be considerably reduced because algorithm are implemented in software using commercially available digital signal processors. This method is based on a parallel block processing approach, where a continuously supplied input data is divided into blocks, and the blocks are processed concurrently by being assigned to each processor in the system. This approach not only requires a simple interconnection network but also reduces the number of communications among the processors very much. The data dependency of the blocks to be processed concurrently brings on dependency problems between the processors in the system. A systematic scheduling method has been developed by using a processors which can be used efficiently, the methods for solving dependency problems between the processors are investigated. Implementation procedures and results for FIR, recursive (IIR), and adaptive filtering algorithms are illustrated.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.8
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• pp.1226-1237
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• 1993

For data transmission over strictly band-limited non-ideal channels, different types of filters with arbitrary responses are needed. In this paper. we proposed two efficient techniques for the design of such FIR filters whose response is specified in either the time or the frequency domain. In particular when a fractionally-spaced structure is used for the transceiver, these filters can be efficiently designed by making use of characteristics of oversampling. By using a minimum mean-squared error criterion, we design a fractionally-spaced FIR filter whose frequency response can be controlled without affecting the output error. With proper specification of the shape of the additive noise signals, for example, the design results in a receiver filter that can perform compromise equalization as well as phase splitting filtering for QAM demodulation. The second method ad-dresses the design of an FIR filter whose desired response can be arbitrarily specified in the frequency domain. For optimum design, we use an iterative optimization technique based on a weighted least mean square algorithm. A new adaptation algorithm for updating the weighting function is proposed for fast and stable convergence. It is shown that these two independent methods can be efficiently combined together for more complex applications.