• Journal of information and communication convergence engineering
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• v.7 no.4
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• pp.507-515
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• 2009

In this paper, we propose an image signal processor (ISP) for an ultra small HD-grade video sensor with 3A (AWB, AE, and AF) in camera phones that can process 720P/30fps videos. In order to enhance the video quality of the systems, it is necessary to achieve the high performance of the 3A. The proposed AWB algorithm multiplies the adjusted coefficients of color gains to the captured data of white objects. The proposed AE method adopts the index step moving based on the difference between an averaged Y luminance and a target luminance, together with IIR filters with variable time responses. The proposed AF technique controls the focus curve to find the lens position that maximizes the integrated high frequency components in luminance values by using highpass filters. Finally, we compare the image quality captured from our system to the quality of a commercial HD camcorder in order to evaluate the performance of the proposed ISP. The proposed ISP system is also fabricated with 0.18um CMOS flash memory process.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.1
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• pp.349-354
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• 2006

For robust INS/GPS navigation system, an efficient multi-filter fusion technique is proposed. In the filtering for nonlinear systems, the representative filter - EKF, and the alternative filters - RHKF filter, SPKF, etc. have individual advantages and weak points. The key concept of the multi-filter fusion is the mergence of the strong points of the filters. This paper fuses the IIR type filter - EKF and the FIR type filter - RHKF filter using the adaptive strategy. The result of the fusion has several advantages over the EKF, and the RHKF filter. The advantages include the robustness to the system uncertainty, temporary unknown bias, and so on. The multi-filter fusion technique is applied to the tightly coupled INS/GPS navigation system and the performance is verified by simulation.

• International Journal of Control, Automation, and Systems
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• v.6 no.6
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• pp.794-799
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• 2008

This paper presents a user identification method at H.264 streaming using watermarking with fingerprints. The watermark can efficiently reduce the potential danger of forgery or alteration. Especially a biometric watermark has convenient, economical advantages. The fingerprint watermark can also improve reliability of verification using automated fingerprint identification systems. These algorithms, however, are not robust against common video compression. To overcome this problem, we analyze H.264 compression pattern and extract watermark after restoring damaged watermark using various filters. The proposed algorithm consists of enhancement of a fingerprint image, watermark insertion using discrete wavelet transform and extraction after restoring. The proposed algorithm can achieve robust watermark extraction against H.264 compressed videos.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.93-96
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• 2000

This paper has been proposed an audio DAC structure composed of FIRs and IIR filters as digital interpolation filter to integrate the off-chip analog low-pass filter on-a-chip. The passband ripple(< 0.41${\times}$fs), passband attenuation(at 0.41${\times}$fs) and stopband attenuation(> 0.59${\times}$fs) of the Δ$\Sigma$ modulator output using the proposed digital interpolation filter had ${\pm}$ 0.001 ［㏈］, -0.0025［㏈］ and -75［㏈］, respectively. Also the inband group delay was 30.07/fs［s］ and the error of group delay was 0.1672%. Also, the attenuation of stopband has been increased -20［㏈］ approximately at 65［㎑］, out-of-band. Therefore the RC products of analog low-pass filter on chip have been decreased compared with the conventional digital interpolation filter structure.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.447-450
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• 2005

Multi-channel artificial reverberation algorithm to control perceived direction and distance is described in this paper. In conventional algorithms using IIR filters, reverberation time is the only parameter to be controlled. Moreover, since the convolution-based conventional algorithms apply only same impulse responses, but not considering sound localization, it was not realistic enough. The new algorithm proposed in this paper utilizes early reflections segmented according to the azimuth from which direct sound comes and controls perceived direction by panning the direct sound, and controls perceived distance by adjusting Energy Decay Curve (EDC) of reverberation and gain of the direct sound. In addition, the algorithm enhances Listener Envelopment(LEV) to make late reverberation incoherent among channels.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.2
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• pp.306-317
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• 1997

This ppaer presents a bit-rate conversion system for efficient communications between 3 channel CVSD systems with different bit-rates. The proposed conversion system is implemented in the digital domain and specially, the conversion problem between 32 Kbps and 16 Kbps CVSD systems is studied. The conventional conversion system implemented in the analog domain allows signals to be easily degraded by external noises. To overcome this problem, a digital CVSD bit-rate conversion system robust to external noises is developed. the new systemdecodes CVSD bit sequences and converts sampling rates of decoded signals, then encodes signals at target bit-rates. Since linear phase property does not matter in this application, instead of FIR filters a IIR filter is employed to reduce the system complexity. Therefore, a 3 channel digital CVSD bit-rate conversion system was successfully real-time implemented using a general purpose DSP. In addition, conversion problems with unkown time constants were experimented and good experimental results were obtained.

• 제어로봇시스템학회:학술대회논문집
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• 1990.10b
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• pp.1222-1227
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• 1990

The problem of identification of continuous systems is considered when both the discrete input and output measurements are contaminated by white noises. Using a predesigned digital low-pass filter, a discrete-time estimation model is constructed easily without direct approximations of system signal derivatives from sampled data. If the pass-band of the filter is designed so that it includes the main frequencies of both the system input and output signals in some range, the noise effects are sufficiently reduced, accurate estimates can be obtained by least squares(LS) algorithm in the presence of low measurement noises. Two classes of filters(infinite impulse response(IIR) filter and finite impulse response(FIR) filter) are employed. The former requires less computational burden and memory than the latter while the latter is suitable for the bias compensated least squares(BCLS) method, which compensates the bias of the LS estimate by the estimates of the input-output noise variances and thus yields unbiased estimates in the presence of high noises.

• Proceedings of the KOSOMBE Conference
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• v.1998 no.11
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• pp.68-69
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• 1998

Discriminating the clutter signal from Doppler signal is the main function of CDP(color Doppler processor). Up to now, a general method of eliminating clutter signal is using IIR high pass filter. There are many filters that were introduced in other paper. In this paper, we propose the new method of filtering clutter signal. To the new method, we adopt an appropriate filter that can eliminate clutter filter most effectively.

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

• Journal of the Korean Institute of Telematics and Electronics A
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• v.29A no.3
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• pp.96-105
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• 1992

The use of fixed-point digital signal processors, such as the TMS 320C25, requires scaling of data at each arithmetic step to prevent overflows while keeping the accuracy. A software which automatizes this process is developed for TMS 320C25. The programmers use a model of a hypothetical floating-point digital signal processor and a floating-point format for data representation. However, the program and data are automatically translated to a fixed-point version by this software. Thus, the execution speed is not sacrificed. A fixed-point variable has a unique binary-point location, which is dependent on the range of the variable. The range is estimated from the floating-point simulation. The number of shifts needed for arithmetic or data transfer step is determined by the binary-points of the variables associated with the operation. A fixed-point code generator is also developed by using the proposed automatic scaling software. This code generator produces floating-point assembly programs from the specifiations of FIR, IIR, and adaptive transversal filters, then floating-point programs are transformed to fixed-point versions by the automatic scaling software.