• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.39-44
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• 2012

In this paper, we studied a high speed Trellis Coded Modulation (TCM) system and 8PSK-TCM system, and compare 8PSK-TCM system with QPSK system of BER. We know that 8PSK-TCM had 3 db improvement of BER than QPSK. And we demonstrated the superiority of 8PSK-TCM by simulating with Matlab. Input signal was 2 bits long, and 8PSK-TCM used 3bits long. Results of this simulation said that 8PSK-TCM had 3 db improvement of BER than QPSK.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.427-430
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• 2005

A multi-code biorthogonal code keying (MBCK) system consists of multiple waveform coding blocks, and the sum of output codewords is transmitted. Drawback of MBCK is that it requires amplifier with high linearity because its output symbol is multi-level. MBCK with constant amplitude precoding block (CA-MBCK) has been proposed, which guarantees sum of orthogonal codes to have constant amplitude. The precoding block in CA-MBCK is a redundant waveform coder whose input bits are generated by processing the information bits. Redundant bits of constant amplitude coded CA-MBCK are not only used to make constant amplitude signal but also used to improve the BER performance at the receiver. In this paper, we proposed a transmission scheme which combines CA-MBCK with $Q^2PSK$ modulation to improve bandwidth efficiency of CA-MBCK and also uses chip interleaving to maintain a constant amplitude feature of CA-MBCK. bandwidth efficiency of a proposed transmission scheme is increased fourfold. And the BER performance of the scheme is same as that of CA-MBCK.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.8
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• pp.1684-1690
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• 2003

In this paper, we proposed a fuzzy controller for the evaluation of the quantization Parameters in the H.263 coder to optimize the subjective quality of each coded frame, keeping the transmission rate constant. We adopted the Mamdani method for fuzzification and the centroid method for defuzzification. The energy and entropy are correlated to features of the HVS in spatial domain, while motion vectors are used to estimate the temporal characteristics of the signal. And then, the fuzzy inputs adapted the variance and the entropy in spatial domain, and the motion vector in temporal domain. We induced the fuzzy membership function and decided the fuzzy relevance to be compatible in visual characteristics. And then, we designed FAM banks. The fuzzy technology has been applied to a practical video compression. This results is obtained an effective rate control technique, an optimum bit allocation and a high subjective quality using fuzzy quantization.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.348-356
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• 2005

In this paper, we propose a fast bitrate reduction transcoding method to convert a bitstream coded by H.264 into a lower bitrate H.264 bitstream. Block mode informations and motion vectors generated by H.264 decoder are used for probability-based block mode determination in the proposed transcoding method. And the motion vector reuse and motion vector refinement process are applied in the proposed transcoding. In the experiment results, the proposed methods achieves approximately 40 times improvement in computation complexity compared with the cascaded pixel domain transcoding, while the PSNR(Peak Signal to Noise Ratio) is degraded with only $0.1\~0.3$ dB.

• Journal of Broadcast Engineering
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• v.8 no.2
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• pp.153-162
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• 2003

This paper describes a structure of adaptive equalizer to improve reception performance of ATSC digital television (DTV) for 8-VSB receivers. There are many strong and dynamic echoes affecting reliable reception of DTV signal. Conventional DFE based least mean square (LMS) algorithm is readily implemented and has good Performance. There are still problems to be solved, however, in handling strong echoes and indoor reception. In this paper, structure of adaptive equalizer to mitigate these Problems in strong multipath interference conditions and indoor reception environment is first presented. Methods to reduce error propagation effects on DFE and initialization scheme of filter coefficients for fast convergence are then introduced. Computer simulation results prove that an adaptive equalizer with proposed design methods can combat with Brazil Ensemble and the Threshold of Visibility(TOV) is improved.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.3A
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• pp.352-361
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• 2000

The performance of a multi-carrier CDMA system is analyzed considering frequency selective multipath fading and Near/Far effects. The number of multicarrier, multiuser, and arms of RAKE receiver, and the decay ratio of frequency selective multipath fading are used as a parameter for the performance analysis. More over, the distribution and the strength of multiuser interference are also considered. To evaluate the Near/Far effects in a multi-carrier CDMA system, three distribution models are assumed. In the first model, interference to carrier Ratio, I/C, ranges from -4 dB to 4dB, and at each 2 dB interval, 20 % of multiuser is assumed to be uniformly distributed. In the second one, I/C ranges from -2 dB to 2 dB, and 33.3% of multiuser is assumed to be equally dispersed at each 2dB interval. The third model is 0 dB of I/C, that is, with perfect power control, multiuser are assumed to be evenly located. In this paper, multi-carrier CDMA system adoption RAKE receiver is proposed to mitigate the frequency selective multipath fading. From the results, the third model(i.e. perfect power control) shows the best performance, and the narrower range of I/C causes the less effects to the desired signal, which reads to the better performance.

• Journal of Internet Computing and Services
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• v.9 no.1
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• pp.21-32
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• 2008

In this paper, we propose new modulation schemes using the conventional CACB (Constant Amplitude Coded Multicode Biorthogonal) modulation with constant amplitude property. Also the proposed modulation schemes supports high transmission data rate by increasing the spectral efficiency, In order to obtain the high spectral efficiency, the $Q^2$AM (Quadrature-Quadrature Amplitude Modulation) and CA-$Q^2$AM (Constant Amplitude-$Q^2$AM) are used. We explain the simplest combining modulation scheme of CACB and $Q^2$AM (i.e., CACB-$Q^2$AM). However, this modulation scheme cannot support the constant amplitude property. Hence the first CACB-CA-$Q^2$AM (or CACB-CA-$Q^2$AM I) modulation scheme is proposed for the constant amplitude property. In the modulation scheme, the redundant constant amplitude encoding (spectral efficiency decrease) is required. Therefore, the second CACB-CA-$Q^2$AM (or CACB-CA-$Q^2$AM II) modulation scheme is proposed retaining the constant amplitude and the spectral efficiency. Computer simulations show that the proposed CACB-CA-$Q^2$AM II is the efficient modulation scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.532-538
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• 2003

In this paper we have investigated a novel approach applying low-density parity-check coding to a COFDM-CDMA system, which operates in a multi-path fading mobile channel. Developed as a linear-block channel coder, the LDPC code is known for a superior signal reception capability in AWGN and/or flat fading channels with respect to increased encoding rates, however, its performance degrades when the communication channel becomes multi-path fading. For a typical multi-path fading mobile channel with a SNR of 16㏈ or lower. in order to obtain a BER lower than 1 out of 10000, the LDPC code with encoding rates below 1:3 requires not only the inherent parity check information but also the piloting information for refreshing front-end equalizer taps of COFDM-CDMA, periodically. For instance, while the 1:3-rate LDPC coded transmission symbol is consisted of data bits and parity-check bits in 1 to 3 proportion, on the other hand, in the proposed method the same rate LDPC transmission symbol contains data bits, parity check bits, and pilot bits in 1 to 2 to 1 proportion, respectively. The included pilot bits are effective not only for channel estimation and channel equalization but for symbol decoding by assisting the parity-check bits, hence, improving SNR vs BER performance over the conventional 1:3-rate LDPC code. The proposed system performance has been verified using computer simulations in multi-path, Rayleigh fading channels, and the results show us that the proposed method out-performs the general LDPC channel coding methods in terms of SNR vs BER measurements.

• Journal of KIISE:Computing Practices and Letters
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• v.10 no.4
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• pp.309-318
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• 2004

This paper proposes the H.264-based selective fine granular scalable (FGS) coding scheme that selectively uses the temporal prediction data in the enhancement layer. The base layer of the proposed scheme is basically coded by the H.264 (MPEG-4 Part 10 AVC) visual coding scheme that is the state-of-art in codig efficiency. The enhancement layer is basically coded by the same bitplane-based algorithm of the MPEG-4 (Part 2) fine granular scalable coding scheme. In this paper, we introduce a new algorithm that uses the temproal prediction mechanism inside the enhancement layer and the effective selection mechanism to decide whether the temporally-predicted data would be sent to the decoder or not. Whenever applying the temporal prediction inside the enhancement layer, the temporal redundancies may be effectively reduced, however the drift problem would be severly occurred especially at the low bitrate transmission, due to the mismatch bewteen the encoder's and decoder's reference frame images. Proposed algorithm selectively uses the temporal-prediction data inside the enhancement layer only in case those data could siginificantly reduce the temporal redundancies, to minimize the drift error and thus to improve the overall coding efficiency. Simulation results, based on several test image sequences, show that the proposed scheme has 1∼3 dB higher coding efficiency than the H.264-based FGS coding scheme, even 3∼5 dB higher coding efficiency than the MPEG-4 FGS international standard.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.21-29
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• 2006

In this Paper a split bandwidth wideband speech coder and its highband coding method are Proposed. The coder uses a split-band approach. where the wideband input speech signal is split into two equal frequency bands from 0-4kHz and 4-8kHz. The lowband and the highband are coded respectively by the 11.8kb/s G.729 Annex E and the proposed coding method. After the LPC analysis, the highband is divided by two modes according to the properties of signals. In stationary mode. the highband signals are compressed by the mixture excitation model; CELP algorithm and W (Matching Pursuit) algorithm. The others are coded by the only CELP algorithm. We compare the performance of the new wideband speech coder with that of G.722 48kbps SB-ADPCM and G.722.2 12.85kbps in a subjective method. The simulation results show that the Performance of the proposed wideband speech coder has better than that of 48kbps G.722 and no better than that of 12.85kbps G.722.2.