• Journal of Communications and Networks
• /
• v.15 no.3
• /
• pp.266-275
• /
• 2013

The exact closed-form expressions for outage probability and bit error rate of spectrum sharing-based multi-hop decode-and-forward (DF) relay networks in non-identical Rayleigh fading channels are derived. We also provide the approximate closed-form expression for the system ergodic capacity. Utilizing these tractable analytical formulas, we can study the impact of key network parameters on the performance of cognitive multi-hop relay networks under interference constraints. Using a linear network model, we derive an optimum relay position scheme by numerically solving an optimization problem of balancing average signal-to-noise ratio (SNR) of each hop. The numerical results show that the optimal scheme leads to SNR performance gains of more than 1 dB. All the analytical expressions are verified by Monte-Carlo simulations confirming the advantage of multihop DF relaying networks in cognitive environments.

• The Transactions of the Korea Information Processing Society
• /
• v.2 no.3
• /
• pp.374-381
• /
• 1995

For designing radio communication systems, radio-channel effects must be considered in order to obtain the desired communication quality, transmitting power, transmission speed and bit error rate. In radio channel, one of major factors that degrade communication quality is multipath fading between transmitting and receiving points. Therefore, a channel model which can describe fading effects correctly is requested. This paper deals with the composition of the channel simulator model that describes multipath fading effects and delay times which occur in the channel. For the performance analysis of the proposed model, trandmitting signal is used in the simulation. From simulation results, it can be shown that probability density function. level crossing rates and average fading-duration time distribution of the faded receive signal are very similar with theoretic values.

• ETRI Journal
• /
• v.39 no.5
• /
• pp.707-717
• /
• 2017

This paper presents a novel generalized quadrature spatial modulation (GQSM) transmission scheme using antenna grouping. The proposed GQSM scheme combines QSM and conventional spatial multiplexing (SMux) techniques in order to improve the spectral efficiency (SE) of the system. Analytical and simulation results show that the proposed transmission scheme has minimal losses in terms of the average bit error probability along with the advantage of an increased SE compared with previous SM and QSM schemes. For the case studies, this advantage represents a reduction of up to 81% in terms of the number of required transmit antennas compared with QSM. In addition, a detection architecture based on the ordered successive interference cancellation scheme and the QR decomposition is presented. The proposed QRD-M adaptive algorithm showed a near-maximum-likelihood performance with a complexity reduction of approximately 90%.

• Proceedings of the Korea Contents Association Conference
• /
• 2008.05a
• /
• pp.69-73
• /
• 2008

In this paper, analytical methods for calculating the average probability of bit error of time hopping pulse position modulation ultra wideband (TH-PPM UWB) system are given. For the multi-user DS-PAM UWB system, the bipolar pulse amplitude modulation is used in order to achieve better performance. As we know, more attention is paid to the TH-PPM UWB systems recently. In this paper, we first introduce the accurate BER calculation methods of the multi-user TH-PPM UWB systems and then give the performance analysis over the ideal AWGN channel and a correlation receiver. Furthermore, we also introduce their applications in image transmission and data transmission and give the simulation results. The analytical method yields simple and exact formulas relating the performance to the system parameters.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.8 no.1
• /
• pp.42-51
• /
• 1997

The satellite channel with a line-of-sight signal component is modeled by a shadowed Rician fading channel. We adopt a direct-sequence / code division multiple access (DS / CDMA), which has the advantage to suppress the multipath effect and increase the user capacity. The performance which is evaluated by bit error probability is subjected to the influence of branch number, multi-user number, and spreading code-length. As the result of the analysis, performance advance is achieved with multi-user number decreasing, number of brnaches increasing, and spreading code-length increasing as chip duration is constant. To use both of diversity combining scheme and channel coding is more efficient for performance improvement than the case using diversity combining scheme only. The use of BCH coding and convolutional coding shows better consequence than Hamming coding. Totally, the performance degradation for heavy shadowing is much larger than that for light and average shadowing as heavy shadowing decreases LOS signal.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.22 no.3
• /
• pp.282-291
• /
• 2011

Adaptive modulation(AM) is an important technique to increase the system efficiency, in which transmitter selects the most suitable modulation mode adaptively according to channel state in the temporary and spatially varying communication environment. Fixed modulation on channels with varying signal-to-noise ratio(SNR) is that the bit-errorrate(BER) probability performance is changing with the channel quality. An adaptive modulation scheme can be designed to have a BER which is constant for all channel SNRs. The correct as well as fast and simple SNR estimation is required essentially for this adaptive modulation. In order to operate adaptive modulation system effectively, in this paper, we analyze the effect of SNR estimation performance to it through the average BER and data throughput. Applying SNR estimation based on auto-correlation of decision feedback signal and others to adaptive modulation system, we also confirm performance degradation or improvement of its which is decided by SNR estimation error at each transition point of modulation level. Since SNR estimation based on auto-correlation of decision feedback signal shows stable estimation performance for various quadrature amplitude modulation(QAM) comparatively, this can be reduced degradation than others at each transition point of modulation level.

• Journal of Intelligence and Information Systems
• /
• v.24 no.2
• /
• pp.21-35
• /
• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.