• Smart Media Journal
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• v.1 no.3
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• pp.36-42
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• 2012

Recently, the smart applications, such as smart phone and smart TV, become a hot issue in IT consumer markets. In particular, the smart TV provides 3D video services, hence efficient coding methods for 3D video data are required. Three-dimensional (3D) video involves stereoscopic or multi-view images to provide depth experience through 3D display systems. Binocular cues are perceived by rendering proper viewpoint images obtained at slightly different view angles. Since the number of viewpoints of the multi-view video is limited, 3D display devices should generate arbitrary viewpoint images using available adjacent view images. In this paper, after we explain a view synthesis method briefly, we propose a new algorithm to compensate view synthesis errors around object boundaries. We describe a 3D warping technique exploiting the depth map for viewpoint shifting and a hole filling method using multi-view images. Then, we propose an algorithm to remove boundary noises that are generated due to mismatches of object edges in the color and depth images. The proposed method reduces annoying boundary noises near object edges by replacing erroneous textures with alternative textures from the other reference image. Using the proposed method, we can generate perceptually inproved images for 3D video systems.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.34-41
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• 2000

This paper proposes a new double-talk detection algorithm which is based on the fuzzy rules, in the adaptive echo canceller of telecommunication system. In this method, the two inputs of the fuzzy inference for detecting double-talk condition are used. One is the cross-correlation coefficient between the error signal and the primary signal which is the summation of the real echo signal and the near-end signal. The other one is the cross-correlation coefficient between the estimation error signal and the primary signal. The fuzzy controller makes a fuzzification for two inputs by the membership functions of trapezoid does the max-min composition using if-then rules. The composed result is defuzzificated by the center gravity method. And by defuzzificated values, the double-talt the echo path variance, and the echo path variance during the double-talk are detected. It is confirmed by computer simulation that this fuzzy double-talk detector is able to estimate the double talk and the echo path variation condition, and even track echo path variation more accurately than the conventional algorithm during the double-talk period.

• Journal of Broadcast Engineering
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• v.5 no.1
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• pp.123-129
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• 2000

This paper addresses a new double detection algorithm which is based on the fuzzy control in the adaptive echo canceller of communication system. In this method, the two input of the fuzzy inference for detecting double talk condition are used. The one is the cross-correlation coefficient between the error signal and the primary signal which is the summed signal of the real echo signal and the near-end signal. The other is the cross-correlation coefficient between the estimation error signal and the primary signal. The fuzzy controller made a fuzzification for two inputs by the membership functions of trapezoid and them became the composition using inference rules. The composed result is defuzzificated by the center gravity method. The output is compared with two threshold values to detect double talk and echo path variation effectively. It is confirmed by computer simulation that this fuzzy double talk detector is able to track echo path variation accurately.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.10a
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• pp.673-676
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• 2008

In wireless communications such as Bluetooth, Baseband should be able to minimize the DC bias of the data which passed the modem interface of either transmitter or receiver for the reliability of the circuit and the integrity of the data. The transmitter scrambles the data to send randomly to the error correction block and the receiver recovers the randomly spread data as they have been. To design the whitening block, it is important to select the prime polynomial for the filtering. In this paper, we designed a optimal whitening block using the prime polynomial $g(D)=D^7+D^4+1$ for hardware and area efficiency. The proposed hardware whitening block was described and verified using Verilog HDL and later to be automatically synthesized. The synthesized whitening block operated at 40Mhz normal clock speed of the target baseband microcontroller.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.11 no.7
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• pp.1186-1193
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• 2000

In this paper, the system performance with the selection diversity, which is applied to the FH/MFSK system in Nakagami fading channel, is analyzed. The deletion probability is derived from the received signal to noise ratio(SNR) after selection combining and the parameters such as the number of users(M), SNR, Nakagami fading figure(m), and the number of diversity branches(D) is used for the performance analysis of the FH/MFSK system. Assuming that m set 1, it is observed that the bit error rate(BER) is 1.0$\times$$10^{-3}$ and 1.0$\times$$10^{-4}$ at D =1(no diversity) and D=2, respectively, and then is decreased by 10 times. Assuming that m set 2, it is also shown that the BER has a constant value although D is increased. In the case of D=2, the system capacity is more 75% and 20% than that considering no diversity at SNR=15 dB and 25 dB, respectively.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.69-75
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• 2010

In this paper, the pitch is detected after the elimination of formant ingredients by flattening the spectrum in frequency domain. In order to remove impact of formant and transition frequency in the signal spectrum, formant envelop is made by linear interpolation with any points each sub-band and the spectrum of speech signal is compensated by the reverse of the envelop interpolated linearly after we divide frequency band into several segment based on LSP and detect the points. The experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum, Lifter methods. The method reduced the gross error rate 1.30% than the LPC method which appeared a good performance except the proposed method. Also, the proposed method showed low error rate in noise environment.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.1
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• pp.24-36
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• 2003

The digital mobile communication will be developed till getting multimedia service in anyone, any where, any time. Theses requiring items are going to be come true via IMT-2000 system. Transmitting signal bandwidth of IMT-2000 system is 3 times as large as IS-95 system. That is mean peak to average of signal is higher than IS-95A system. So we have to design it carefully not to effect in adjacent channel. HPA(High Power Amplifier) located in the end point of system is operated in 1-㏈ compression point(Pl㏈), then it generates 3rd and 5th inter modulation signals. Theses signals affect at adjacent channel and RF signal is distorted by compressed signal which is operated near by Pl㏈ point. Then the most important design factor is how we make HPA having high linearity. Feedback, Pre-distorter and Feed-forward methods are presented to solve theses problems. Feed-forward of these methods is having excellent improving capacity, but composed with complex structure. Generally, Linearity and Efficiency in power amplifier operate in the contrary, then it is difficult for us to find optimal operating point. In this paper we applied algorithm which searches optimal point of linear characteristics, which is key in Power Amplifier, using minimum current point of error amplifier in 1st loop. And we made 2nd loop compose with new structure. We confirmed fabricated LPA is operated by having high linearity and minimum current condition with ACPR of -26 ㏈m max. @ 30㎑ BW in 3.515㎒ and ACLR of 48 ㏈c max@${\pm}$㎒ from 1W to 40W.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.11
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• pp.2465-2473
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• 2011

This paper describes a circuit-level optimization of DFU(decoding function unit) for LDPC decoder which is used in wireless communication systems including WiMAX and WLAN. A new design of DFU based on sign-magnitude arithmetic instead of two's complement arithmetic is proposed, resulting in 18% reduction of gate count for 96 DFUs array used in mobile WiMAX LDPC decoder. A multi-mode LDPC decoder for mobile WiMAX standard is designed using the proposed DFU. The LDPC decoder synthesized using a 0.18-${\mu}m$ CMOS cell library with 50 MHz clock has 268,870 gates and 71,424 bits RAM, and it is verified by FPGA implementation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.12
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• pp.2762-2770
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• 2012

In this paper, we propose a novel approach to effectively compress the sub-images transformed from the picked-up elemental images in integral imaging, in which motion vectors of the object in each sub-image are fast and accurately estimated and compensated by combined use of MSE(mean square error)-based TSS(tree-step search) and FS(full search) schemes. This is, the possible object areas in each sub-image are searched by using the fast TSS algorithm in advance, then the these selected object areas are fully searched with the accurate FS algorithm. Furthermore, the sub-images in which all object's motion vectors are compensated, are transformed into the residual images by using the difference image method and finally compressed with the MPEG-4 algorithm. Experimental results reveal that the proposed method shows 214% improvement in the compression time per each image frame compared to that of the conventional method while keeping the same compression ratio with the conventional method. These successful results confirm the feasibility of the proposed method in the practical application.