• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4C
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• pp.200-206
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• 2005

Adaptive transmission techniques can improve the performance of wireless communication system by adaptively changing the transmission parameter such as modulation, code-rate, and power according to the channel state. For orthogonal frequency division multiplexing (OFDM) system, the adaptive transmission techniques can be applied to each subcarrier unit. In this paper, we consider the adaptive code-rate OFDM system in which optimal code-rate is applied to each subcarrier according to the subchannel state. Performance analysis show that $3\sim6$dB gain of SNR or up to $30\sim50\%$ increase of data rate are achieved in the condition of bit error rate $10^{-6}$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.2
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• pp.121-131
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• 1992

In this paper, three methods are studied to enhance the performance of SBC ( Sub-Band Coding )schemes for voice signal at the medium bit rate between 12 kbps and If; kbps, and adaptive postfilteritng using human auditory characteristics Is (Bone at the decoder out put. First, GQMF(Generalized Quadrature Mirror Filter ) Is used instead of QME'((Quadrature MirrorFiltcr ) to have better performance. Second, by adaptive bit allocation to each sub-band, speech quality is enhanced and valuable rate ceding If possible. Third, corriparlson study oS thr: coder performance using APCM(Adaptive Pulse Code ModulatioTi) and ADPCM( Adaptive Differentiai Pulse Code Modulatiori) , Indicates that SB AfCM performance better than the other. Adaptive postfiltering at the decoder output enhances the quality of the coded speech. The two proposed postfiltering methods decrease the noise sufficiently at the expense of the low computational load.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.59-61
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• 1995

An adaptive code rate change scheme in DS-SSMA systems is proposed. In the proposed scheme, the error correcting code rate is changed according to the channel state, the effective number of users. The channel state is estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.3B
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• pp.251-258
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• 2002

In this paper, an adaptive convolutional coding method in multicarrier direct sequence code division multiple access (DS/CDMA) systems is considered. In order to accommodate a number of coding rates easily and make the encoder and decoder structure simple, we use the rate compatible punctured convolutional (RCPC) code. To achieve maximum data throughput, an adaptive rate system based on the channel state information (the SINR estimate) is proposed. We show that the proposed adaptive rate convolution coded multicarrier DS/CDMA systems can enhance spectral efficiency and provide frequency diversity.

• Journal of Communications and Networks
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• v.7 no.4
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• pp.462-470
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• 2005

We propose a multi-rate and multi-BEP transmission scheme using adaptive overlapping pulse-position modulator (OPPM) and optical power controller in optical code division multiple access (CDMA) networks. The proposed system achieves the multi-rate and multi-BEP transmission by accommodating users with different values of OPPM parameter and transmitted power in the same network. The proposed scheme has advantages that the system is not required to change the code length and number of weight depending on the required bit rate of a user and the difference of bit rates does not have so much effect on the bit error probabilities (BEPs). Moreover, the difference of transmitted powers does not cause the change of bit rate. We analyze the BEPs of the four multimedia service classes corresponding to the com­binations of high/low-rates and low/high-BEPs and show that the proposed scheme can easily achieve distinct differentiation of the service classes with the simple system configuration.

• Journal of Electrical Engineering and information Science
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• v.1 no.1
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• pp.23-28
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• 1996

In this series of two papers, adaptive code rate change schemes in DS-SSMA systems are proposed. In the proposed schemes the error correcting code rate is changed according to the channel states. Two channel states having significant effects on the bit error probability are considered: one is the effective number of users considered in Part 1, and the other is the fading environment considered in Part 2. These channel states are estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound. Simulation results show that we can transmit maximum amount of information if we change the code rate based on the channel states.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.258-262
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• 2005

An adaptive beamforming system based on code filtering and differential correlation approaches is proposed. The differential correlation method was originally proposed for time delay estimation of direct sequence code division multiple access (DS-CDMA) systems under near-far ratio conditions and the code filtering correlation algorithm, on the other hand, was proposed for array response estimation in DS-CDMA systems under perfect power control. In this paper, by combining differential correlation concept with the code filtering beamforming technology, an accurate estimate of the beam forming weights and an enhanced performance of DS-CDMA systems under sever near-far ratio conditions is achieved. The system performance in terms of beam pattern and bit-error-rate (HER) shows that the proposed adaptive beamformer outperforms the conventional code filtering correlation technique.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.157-158
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• 2008

This paper examines an adaptive coding scheme for flat fading channels to maximize the average code rate of a coded system. The proposed adaptation technique is employed by using the required free distance of a rate compatible code depending on the channel realization. First, the system will calculate the required free distance based on the instantaneous channel gain. Based on this channel gain we will select a set of convolution code to optimize the code rate with a certain performance requirement. Simulation results show that our proposal can achieve a higher code rate.

• Journal of Electrical Engineering and information Science
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• v.1 no.1
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• pp.17-22
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• 1996

Adaptive code rate change schemes in DS-SSMA systems are proposed. In the proposed schemes, the error correcting code rate is changed according to the channel states. Two channel states having significant effects on the bit error probability are considered: one is the effective number of users, and the other is the fading environment. These channel states are estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.