• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2005.11a
• /
• pp.159-162
• /
• 2005

Recently, due to rapid developments of wireless communication and digital TV, modern society needs to process of aquisition, storage and transmission of much information. So the importance of signal processing is increasing and various digital filters are used in the two-dimensional signal such as image. And kinds of these digital filters are IIR(infinite impulse response) filter and FIR(finite impulse response) filter. And FIR filter which has the phase linearity, the easiness of creation and stability is applied to many fields. In design of this FIR filter, flatness property is a important factor in pass-band and stop-band. In this paper, we designed a 2-D Circular FIR filter using the Bernstein polynomial, it is presented flatness property in pass-band and stop-band. And we simulated the designed filter with noisy test image and compared the results with existing methods.

• ETRI Journal
• /
• v.35 no.4
• /
• pp.697-705
• /
• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.

• The Transactions of The Korean Institute of Electrical Engineers
• /
• v.61 no.10
• /
• pp.1518-1523
• /
• 2012

In this study, the novel digital filtering algorithms are implemented to suppress the noisy characteristics embedded in ambulatory electrocardiogram signals by an android smartphone platform. With this aim, Graphical User Interface (GUI) is designed and implemented by utilizing multithread-Java programming to realize Finite Impulse Response and Infinite Impulse Response filter. With simulating our implemented digital filters built in an android smartphone, we can find the fact that we can efficiently suppresses the noisy characteristics due to baseline wandering and 60 Hz powerline source fluctuations especially in electrocardiograms.

• Journal of Institute of Control, Robotics and Systems
• /
• v.21 no.9
• /
• pp.836-842
• /
• 2015

In this paper, the target size and model based target size estimator (TMBE) algorithm is presented for iimaging infrared (IIR) seeker. At the imaging seeker, target size information is important factor for accurate tracking. The model based target size estimator filter (MBEF) algorithm was proposed to estimate target size at imaging infrared seeker. But, the model based target size estimator filter algorithm need to know relative distance from the target. In order to overcome the problem, we propose target size and model based target size estimator filter (TMBEF) algorithm which based on the target size. The performance of proposed algorithm is tested at target intercept scenario. The experiment results show that the proposed algorithm has the accurate target size estimating performance.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.24 no.2
• /
• pp.190-195
• /
• 1987

This approach to the transmultiplexer is for the 12-channel TDM/FDM translation system with the polyphase network and the FDCT. For the reduction of the conversion time the 14-point FDCT algorithm is used and the polyphase network which translate the protorype filter into the channel filtrs required in each channel is designed. The prototype filters is designed by the IIR/FIR hybrid filter. The number of numerator terms of the hybrid filter is very large compaired to the denomiator terms. Because of symmetrical properties for numerator terms, required multiplication rate is 0.11396x10**6M/sec.ch. and reduced to 25%-45% of the rate required in the other papers. The proposed system is simulated with the computer and by the results it is proved that the proposed conversion method is valid.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2001.11b
• /
• pp.1104-1108
• /
• 2001

It is a difficult and important task to measure the reverberation time of an acoustic room with a short reverberation time. This paper presents a new technique to measure the reverberation time of an acoustic room with low value of BT60. The digital signal processing technique used to do this is the wavelet filter which is very flexible to design the 1/n octave band filter and has no delay problem compared with the conventional IIR digital filter. This method is successfully applied to the measurement of the reverberation time at low frequency band of famous concert halls in Korea.

• Journal of Korea Multimedia Society
• /
• v.17 no.8
• /
• pp.923-936
• /
• 2014

Images can be contaminated with different types of noise, for different reasons. The neighborhood averaging and smoothing by image averaging are the classical image processing techniques for noise removal. The classical spatial filtering refers to the aggregate of pixels composing an image and operating directly on these pixels. To reduce or remove effectively noise in image sequences, it usually needs to use noise reduction filter based on space or time domain such as method of spatial or temporal filter. However, the method of spatial filter can generally cause that signals of objects as the target are also blurred. In this paper, we propose temporal filter using the piece-wise quadratic function model and enhancement algorithm of image quality for the performance improvement of target detection and tracking by heterogeneous noise reduction. Image tracking simulation that utilizes real IIR(Imaging Infra-Red) images is employed to evaluate the performance of the proposed image processing algorithm.

• Journal of the Korean Institute of Navigation
• /
• v.24 no.5
• /
• pp.443-450
• /
• 2000

Before some experiments were carried out with analog bandpass filter which used for filtering the noise included in sound source signal. And this filter was constituted by condenser, register and operational amplifier. Hut these elements made the phase characteristics to differentiate in each sensing channel and cause a little of measurement error. We made new measurement system that was substituted digital filter for the analog filter in order to develop the optimal system which could find the time delay between each sensors with high accuracy. This paper describes the new system's constitution and the function of each parts. Specially three digital filters were designed and applied to the digital signal processing Part. And a series of experiments were carried out with the source＇s distance 9.53meters and the random bearing interval within the limits of $0^{\circ}$ ~ $180^{\circ}$. As a result, we have recognized that the accuracy of measurements were differentiated by the methods what kind of digital filter were adopted. And we have confirmed the facts that IIR LPF was suitable for sound source's bearing measurement and FIR LPF reduced the range measurement error.

• Journal of Advanced Navigation Technology
• /
• v.13 no.3
• /
• pp.337-343
• /
• 2009

In this paper, we proposed a burst mode symbol timing recovery unit that is applicable to the VDL Mode-2 using D8PSK modulation. A method that IIR loop filter is used to minimize symbol timing error is hard to apply to burst mode because its convergence time is long. That is, the fast convergence property is important. In this paper, the proposed method takes one sample which has maximum symbol power after the initial synchronization has been achieved by using preambles. The main principle of operation is that the unit moves one sample clock to advance or retard according to symbol power. We verify that the proposed method is operated well in ${\pm}100$ ppm or greater through the test results between Australia ADS Corp. transmitter and the designed receiver.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
• /
• v.18 no.1
• /
• pp.154-160
• /
• 2004

In this paper, implementation of active noise curtains using multiple channel adaptive filters is presented. The same numbers of single channel LMS algorithms as control loudspeakers is used instead of a multi-channel LMS algorithm to reduce the computational burden of adaptive filter algorithms. In general, a multi-channel LMS algorithm is usually used in active noise control system. but this algorithm has much more computational complexity. The single channel control techniques have less amount of DSP calculation, compared to multiple channel control techniques. A stabilizing procedure for adaptive IIR filters is also proposed to improve the stability of recursive LMS algorithms. Both experimental results of two control techniques using TMS320VC33 digital signal processor show the similar noise reduction, but the single channel control techniques are more efficient in practical active noise curtain applications