• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2005.11a
• /
• pp.250-255
• /
• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the frequency domain algorithm is prefer than the existent time domain. we analyzed the Wavelet algorithm, short-length fast running FIR algorithm, fast-short-length fast running FIR algorithm and proposed algorithm.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.7 no.4
• /
• pp.625-634
• /
• 2006

In this paper, an efficient and fast algorithm to reduce calculation amount of FIR(Finite Impulse Responses) filtering is proposed. Proposed algorithm enables arbitrary size of parallel processing, and their structures are also easily derived. Furthermore, it is shown that the number of multiplication/sample is remarkably reduced. For theoretical improvement, numbers of sub filters are compared with those of conventional algorithm. In addition to the theoretical improvement, it is shown that number of element for hardwired implementation are reduced comparison to those of the conventional algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.6
• /
• pp.187-194
• /
• 2005

In this paper, an efficient and fast algerian to reduce calculation amount of FIR(Finite Impulse Responses) filtering is proposed. Proposed algorithm enables arbitrary size of parallel processing, and their structures are also easily derived. Furthermore, it is shown that the number of multiplication/sample is reduced, and number of instructions using MAC(Multiplication and Accumulation) processor are also reduced. For theoretical improvement numbers of sub filters are compared with those of conventional algorithm. In addition to the theoretical improvement, it is shown that number of element for hardwired implementation are reduced comparison to those of the conventional algorithm.

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

• Proceedings of the Korea Society for Simulation Conference
• /
• 2001.10a
• /
• pp.313-317
• /
• 2001

The approach to predict time series without neglecting the fluctuation in a short period is tried by using a digital FIR filter and a neural network. The differential waveform of the Nikkei average closing price is filtered by the FIR band-pass filter of 101 length. It is filtered into the five frequency bands of 0-1Hz, 1-2Hz, 2-3Hz, 3-4Hz and 4-5Hz by setting the sampling frequency 10Hz. The each filtered waveform is learned and forecasted by the neural network. The neural network of the back propagation method is adopted in the learning the waveform. By inputting the data of 20 days in the past, the prediction of 10 days ahead is carried out. After learning the time series of each frequency band by the neural network, the predicted data far each frequency band are obtained. The predicted waveforms of each frequency band are synthesized to obtain a final forecast. The waveform can be forecasted well as a whole.

• Proceedings of the IEEK Conference
• /
• 2006.06a
• /
• pp.263-264
• /
• 2006

In this paper, an efficient and fast algorithm to reduce calculation amount of FIR(Finite Impulse Responses) filtering is proposed. Proposed algorithm enables arbitrary size of parallel processing, and their structures are also easily derived. Furthermore, it is shown that the number of multiplication/sample is reduced, and number of instructions using MAC(Multiplication and Accumulation) processor are also reduced. For theoretical improvement, numbers of sub filters are compared with those of conventional algorithm. In addition to the theoretical improvement, it is shown that number of element for hardwired implementation are reduced comparison to those of the conventional algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.10 no.3
• /
• pp.11-17
• /
• 2010

Multi-standard, multi-band, multi-service system to ensure a flexible interface between the SDR (Software Defined Radio) technology for the implementation of the Stability and Low-Power, Low-Calcualrion DDC (Digital Down Conversion) technology is essential. DDC technology consists of a digital channel filter. This is a typical digital filter because of the limited fisheries are vulnerable to overflow and rounding errors are drawbacks. In this paper, we overcome this disadvantage, we propose the structure of the DDC. The way WDF (Wave Digital Filter) Structural rounding error due to the structural resistance to noise. Therefore, This is the useful structure when the filter coefficients's word length is short. In addition, since IIR filters based on FIR filters based on the amount of computation is reduced because fewer than filter's tap. The proposed structure is used in DDC that CIC (Cascaded Integrator Comb) filter, WDF, IFOP (Interpolated Fourth-Order Polynomials) were analyzed with respect to, the results were confirmed by computer simulation.

• Journal of the Korean Institute of Telematics and Electronics D
• /
• v.35D no.10
• /
• pp.39-50
• /
• 1998

Wide-band sharp-transition filters are widely used in applications such as wireless CODEC design or medical systems. Since these filters suffer from large sensitivity and roundoff noise, large word-length is required for the VLSI implementation, which increases the hardware size and the power consumption of the chip. In this paper, a low-power implementation technique for digital filters with wide-band sharp-transition characteristics is proposed using CPL (Complementary Pass-Transistor Logic), LWDF (Lattice Wave Digital Filter) and a modified DIFIR (Decomposed & Interpolated FIR) algorithm. To reduce the short-circuit current component in CPL circuits due to threshold voltage reduction through the pass transistor, three different approaches can be used: cross-coupled PMOS latch, PMOS body biasing and weak PMOS latch. Of the three, the cross-coupled PMOS latch approach is the most realistic solution when the noise margin as well as the energy-delay product is considered. To optimize CPL transistor size with insight, the empirical formulas for the delay and energy consumption in the basic structure of CPL circuits were derived from the simulation results. In addition, the filter coefficients are encoded using CSD (Canonic Signed Digit) format and optimized by a coefficient quantization program. The hardware cost is minimized further by a modified DIFIR algorithm. Simulation result shows that the proposed method can achieve about 38% reductions in power consumption compared with the conventional method.