• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10A
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• pp.950-957
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• 2007

In this paper, a recursive intercarrier interference (ICI) cancellation technique for cooperative space-time block coded orthogonal frequency division multiplexing (STBC-OFDM) with different carrier frequency offsets (CFOs) is proposed. Also, different CFOs existing in a cooperative STBC-OFDM system is shown to produce phase difference in the received signal, causing noise enhancement effect in the STBC decoding process. The performance degradation caused by the noise enhancement effect can be effectively reduced by a proposed ML-based recursive ICI cancellation technique. It is shown by computer simulation that the proposed recursive ICI cancellation technique is effective in reducing ICI, especially for STBC-OFDM systems with a large FFT size and large CFOs.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.12
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• pp.3118-3134
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• 2013

This paper deliberates for a multimedia transmission scheme combining multiple description coding (MDC) and network coding (NC). Our goal is to take advantage from the property of MDC to provide quantized and compressed independent and identically distributed (iid) descriptions and also from the benefit of network coding, which uses network resources efficiently to recover lost data in the network. Recently, p-cycle NC has been introduced to recover and protect any lost or distorted descriptions at the receiver part exactly without need of retransmission. So far, MDC have not been explored using this type of NC. Compressed and coded descriptions are transmitted through the network where p-cycle NC is applied. P-cycle based algorithm is proposed for single and multiple descriptions lost. Results show that in the fixed bit rate, the PSNR (Peak Signal to Noise Ratio) of our reconstructed image and also subjective evaluation is improved significantly compared to previous work which is averaging method joint with MDC in order to conceal lost descriptions.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.24 no.1
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• pp.1-6
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• 1988

For the purpose of making the detection range of fish detection system more longer and computerizing the system a parametric sound source, a timer and a digitizing circuit for the Apple II computer have been studied. The parametric sound of 5 KHz generated by passing AND gate two signals from carrier signal generator of 200KHz with modulator of 5KHz. This parametric acoustic source of 5KHz difference frequency had more higher directional resolution of 10 degrees than single frequency sound of 200KHz. Peripheral interface adaptor MC 6821 was adopted for interfacing to the Apple II personal computer. The timer consisted of six decade binary coded decimal counters (74 LS 190), and the digitizing circuit consisted of a sample and hold (LF 398) and an A/D converter(ADC 0808). The timer with 10KHz clock pulse had the measuring time from 0.1msec to 100sec. This time measuring range was satisfactory for the aim of the fish finding acoustic system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2397-2404
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• 2012

Underwater acoustic communication has multipath error because of reflection by sea-level and sea-bottom. The multipath of underwater channel causes signal distortion and error floor. In order to improve the performance, it is necessary to employ an iterative coding scheme. Among the iterative coding scheme, turbo codes and LDPC codes are dominant channel coding schemes in recent. This paper concluded that turbo coding scheme is optimal for underwater communications system in aspect to performance, coded word length, and equalizer combining. Also, decision directed phase recovery was used for correcting phase offset induced by multipath. Based on these algorithms, we confirmed the performance in the environment of oceanic experimentation.

• Economic and Environmental Geology
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• v.21 no.2
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• pp.185-191
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• 1988

A two channel seismic recorder suitable for long-term observation of microearthquakes is developed. The direct analogue recording on cassette tape is adopted in the recorder whose circuits of amplifier and mortor units of an audio cassette recorder are modified. The recorder provides contineous record of 10 days with DC 12V battery (100AH) and with standard cassette tape of 60 minute use. The binary coded time signals of date, hour, and minute are generated once a minute by the timing system and absolute time input using radio to measure the time drift is also possible. For the seismic signal processing, the analogue signals from audio cassette player pass A/D converter and digitized data are stored in personal computer. Then visual records can be obtained using computer graphic mode. Basic programs "ADCONVO" and "DRAWO" to accomplish A/D conversions, the creation of data files and visualization of signals were written. Some sample signals reproduced from the recorded tape are presented.

• Journal of Institute of Control, Robotics and Systems
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• v.20 no.6
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• pp.612-618
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• 2014

A pattern classifier-based approach for recognizing internal states of human participants in interactions is presented along with its experimental results. The approach includes a step for collecting video recordings of human-human interactions or humanrobot interactions and subsequently analyzing the videos based on human coded annotations. The annotation includes social signals directly observed in the video recordings and the internal states of human participants indirectly inferred from those observed social signals. Then, a pattern classifier is trained using the annotation data, and tested. In our experiments on human-robot interaction, 7 video recordings were collected and annotated with 20 social signals and 7 internal states. Several experiments were performed to obtain an 84.83% recall rate for interaction engagement, 93% for concentration intention, and 81% for task comprehension level using a C4.5 based decision tree classifier.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.217-223
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• 2005

In this paper, we propose a predictive block constrained trellis coded quantization (BC-TCQ) to quantize cepstral coefficients for the distributed speech recognition. For Prediction of the cepstral coefficients. the 1st order auto-regressive (AR) predictor is used. To quantize the prediction error signal effectively. we use a BC-TCQ. The performance is compared to the split vector quantizers used in the ETSI standard, demonstrating reduction in the cepstral distance and computational complexity.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.3 no.1
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• pp.52-57
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• 2003

In this paper, we propose a Fuzzy Classifier System(FCS) to find a set of fuzzy rules which can carry out the edge detection. The classifier system of Holland can evaluate the usefulness of rules represented by classifiers with repeated learning. FCS makes the classifier system be able to carry out the mapping from continuous inputs to outputs. It is the FCS that applies the method of machine learning to the concept of fuzzy logic. It is that the antecedent and consequent of classifier is same as a fuzzy rule. In this paper, the FCS is the Michigan style. A single fuzzy if-then rule is coded as an individual. The average gray levels which each group of neighbor pixels has are represented into fuzzy set. Then a pixel is decided whether it is edge pixel or not using fuzzy if-then rules. Depending on the average of gray levels, a number of fuzzy rules can be activated, and each rules makes the output. These outputs are aggregated and defuzzified to take new gray value of the pixel. To evaluate this edge detection, we will compare the new gray level of a pixel with gray level obtained by the other edge detection method such as Sobel edge detection. This comparison provides a reinforcement signal for FCS which is reinforcement learning. Also the FCS employs the Genetic Algorithms to make new rules and modify rules when performance of the system needs to be improved.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1567-1570
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• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.1
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• pp.51-59
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• 2004

We design multilevel coding(MLC) with a semi bit-interleaved coded modulation(BICM) scheme based on low density parity check(LDPC) codes. Different from traditional designs, we joint the MLC and BICM together by using the Gray mapping, which can transmit the multimedia data over several equivalent channels with different code rates. To get a good performance from signal-to-noise ratio(SNR) very close to the capacity of the additive white Gaussian noise(AWGN) channel, random regular LDPC code and a simple semi-algebra LDPC(SA-LDPC) code are discussed in MLC with parallel independent decoding(PID). Finally, the numerical results demonstrate that the proposed scheme could achieve both power and bandwidth efficiency for multimedia communication system.