• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.9 no.10
• /
• pp.3983-4001
• /
• 2015

Exact average bit error probability (BEP) expressions for mobile-relay-based mobile-to-mobile (M2M) cooperative networks with incremental decode-and-forward (IDF) relaying over N-Nakagami fading channels are derived in this paper. The average BEP performance under different conditions is evaluated numerically to confirm the accuracy of the analysis. Results are presented which show that the fading coefficient, the number of cascaded components, the relative geometrical gain, and the power allocation parameter have a significant influence on the average BEP performance.

• International Journal of Internet, Broadcasting and Communication
• /
• v.14 no.1
• /
• pp.78-84
• /
• 2022

In this paper, to improve further the bit-error rate (BER) performance of the stronger channel user in non-uniform source non-orthogonal multiple access (NOMA) with symmetric superposition coding (SSC), we propose a smart bit-to-symbol (BTS) mapping of SSC. First, the analytical expression for the total allocated power of the proposed BTS mapping scheme is derived, and then we show that the BER of the proposed BTS mapping scheme improves further, compared to that of the existing BTS mapping scheme. Moreover, based on the simulations, the signal-to-noise (SNR) gain of the proposed BTS mapping scheme over the existing BTS mapping scheme is calculated. In result, the proposed BTS mapping could be a candidate scheme for non-uniform source SSC NOMA with the SNR gain.

• International journal of advanced smart convergence
• /
• v.11 no.1
• /
• pp.36-41
• /
• 2022

Recently, to improve the performance of the strongest channel gain user in non-orthogonal multiple access (NOMA) with a non-uniform source and symmetric superposition coding (SSC), a novel bit-to-symbol (BTS) mapping have been proposed. However, only the performance of the user with the stronger channel gain was analyzed. Thus, we compare the bit-error rate (BER) of the new BTS scheme with that of uniform sources, especially for the user with weakest channel gain. First, we show that the performance of the novel BTS scheme for the user with weakest channel gain also improves, compared to that of the uniform sources. Furthermore, the signal-to-noise (SNR) gain of the new BTS scheme over the uniform sourcesis calculated. As a consequence, the novel BTS scheme would improve the performance of the user with weakest channel gain as well as that with the stronger channel gain for SSC NOMA with a non-uniform source.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.16 no.2
• /
• pp.162-171
• /
• 1991

This paper presents an algorithm for compressing image data by separating the image into two parts. I.e. edge image containing high-frequency components and non-edge image containing low-frequency components of image. The edge image is extracted by using 8 level compass gradient masks and the non-edge image is obtained by removing the edge image from the original image. The edge image is coded by Huffman run-length code and the non edge image is transformed first by DCT and the transformed images is coded next by a quantized bit allocation table. For an example image. GIRL. the proposed algorithm shows bit rate of 0.52 bpp with PSNR of 36dB.

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

• Electronics and Telecommunications Trends
• /
• v.5 no.4
• /
• pp.42-59
• /
• 1990

세계적인 추세에 근거하여, 디지틀 이동 통신용 음성 부호화 방식 표준안 선정을 위해 평가 대상 방식으로 DSBC(Dynamic bit allocation SubBand Coding), RPE-LTP(Regular Pulse Excited Long Term Prediction),CELP(Code Excited Linear Prediction) 등을 선정한 바 있다.본 논문에서는 이들 방식중 13 kbps RPE-LTP의 실현 및 성능평가에 관하여 다루었다. 먼저 음질에 중요한 영향을 미치는 분석/합성부호화에 근거한 파라미터 양자화 방법 그리고 채널 코딩과의 연계를 위한 비트 중요도 해석 등을 언급하였다. 끝으로 시뮬레이션 결과를 나타내었다.

• Proceedings of the IEEK Conference
• /
• 1999.06a
• /
• pp.712-715
• /
• 1999

In this paper we propose high quality audio coding algorithm using psychoacoustic modelling and the adaptive wavelet Packet decomposition. The bit allocation scheme exploits the remnants of temporal correlations that exist in the wavelet packet coefficients by SPIHT. The proposed algorithm achieve almost transparent coding of monophonic compact disk(CD) quality signals at about 44 kbps.

• Proceedings of the IEEK Conference
• /
• 2006.06a
• /
• pp.1033-1034
• /
• 2006

The paper demonstrates OFDM with LDPC and adaptive modulation applied to Multiple-Input Multiple-Output (MIMO) system. An optimization algorithm to obtain a bit and power allocation for each subcarrier assuming instantaneous channel knowledge is used. The experimental results are shown the potential of our proposed system.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.26 no.9A
• /
• pp.1574-1582
• /
• 2001

본 논문에서는 무선 ATM 망에서 VBR 트래픽에 대한 동적인 대역할당 기법을 제안하였다. 제안된 기법에서는 동적 파라미터를 전송하기 위하여 인 밴드 방식을 사용하였으며, 버퍼의 길이와 변이를 고려하여 이동국에서 2 state bit를 기지국에서 전송하도록 하였다. 기지국에서 수신된 비트의 상태에 따라서 이동국에 데이터 슬롯을 할당한다. 제안한 알고리즘은 VBR 트래픽의 QoS를 보장하고, 시스템의 효율을 높일 수 있도록 하였고, 컴퓨터 시뮬레이션을 통하여 제안된 알고리즘의 성능을 평가하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.41 no.1
• /
• pp.47-56
• /
• 2004

This paper describes a new bit allocation algorithm that can achieve a constant bit rate when coding multiple video objects, while improving rate-distortion (R-D) performance over the VM5 method for MPEG-4 object-based video coding. In particular, we propose two models to estimate the rate-distortion characteristics of coded objects as well as skipped objects. Based on the proposed models, we present several R-D coding modes with spatio-temporal trade-offs to improve coding efficiency. The proposed algorithm is performed at the object level for object-based video coding. Simulation results demonstrate moderate improvement at low as well as high bit rates. The proposed algorithm can produce the actual coded bits very close to the target bits over a wide range of bit rates. Consequently, the proposed algerian has not experienced any buffer overflow or underflow over the bit rates between 32 kbps and 256 kbps.