• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.1C
• /
• pp.81-89
• /
• 2003

For improving the performance of noise suppression in tactical communication environments, an enhanced MELP vocoder is suggested, in which an acoustic noise suppressor is integrated into the front end of the MELP algorithm, and an FEC code into the channel side of the MELP algorithm. The acoustic noise suppressor is the modified IS-127 EVRC noise suppressor which is adapted for the MELP vocoder. As for FEC, the turbo code, which consists of rate-113 encoding and BCJR-MAP decoding algorithm, is utilized. In acoustic noise environments, the lower the SNR becomes, the more the effects of noise suppression is increased. Moreover, The suggested system has greater noise suppression effects in stationary noise than in non-stationary noise, and shows its superiority by 0.24 in MOS test to the original MELP vocoder. When the interleave size is one MELP frame, BER 10-6 is accomplished at channel bit SNR 4.2 ㏈. The iteration of decoding at 3 times is suboptimal in its complexity vs. performance. Synthetic quality is realized as more than MOS 2.5 at channel bit SNR 2 ㏈ in subjective voice quality test, when the interleave size is one MELP frame and the iteration of decoding is more than 3 times.

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.44 no.10
• /
• pp.67-74
• /
• 2007

Turbo code, a kind of channel coding technique, has ben used in the field of digital mobile communication system if the number of iterations is increased in the several channel environments, my further iteration results in very little improvement, and requires much delay and computation in proportion to the number of iterations. In this paper, it proposes an efficient stopping rules for the iteration process in turbo decoding. By using absolute mean values for the LLR difference values between the first and second decoder in the present decoding process, the proposed algorithm can largely reduce the average number of iterations without BER performance degradation in all SNR regions. As a result of simulation, the average number of iterations of proposed algorithm is reduced by about $18.25%{\sim}20.58%$ compared to SDR algerian in the lower SNR region, and is reduced by about $22.96%{\sim}28.74%$ compared to method using variance values of extrinsic information in the upper SNR region.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.20 no.9
• /
• pp.835-842
• /
• 2009

DVB-SSP is a new broadcasting system for hybrid satellite communications, which supports mobile handheld systems and fixed terrestrial systems. However, a critical factor must be considered in upper layer decoding which including erasure Reed-Solomon error correction combined with cyclic redundancy check. If there is only one bit error in an IP packet, the entire IP packet is considered as unreliable bytes, even if it contains correct bytes. IF, for example, there is one real byte error, in an If packet of 512 bytes, 511 correct bytes are erased from the frame. Therefore, this paper proposed two kinds of upper layer decoding methods; LLR-based decoding and hybrid decoding. By means of simulation we show that the performance of the proposed decoding algorithm is superior to that of the conventional one.

• Proceedings of the IEEK Conference
• /
• 2000.06d
• /
• pp.178-181
• /
• 2000

In this paper, we present the implementation of filterbank for MPEG-2 Advanced Audio Coding (AAC) decoder with VHDL. The filterbank of AAC employs a technique called time-domain aliasing cancellation (TDAC). In order to make the algorithm more efficiently, we decompose and reorganize the filterbank algorithm lot the high speed decoding process and lower computational cost. And we make this filterbank algorithm to be used with other modules of AAC decoder in parallel processing.

• Journal of KIISE:Software and Applications
• /
• v.32 no.4
• /
• pp.268-276
• /
• 2005

This paper deals with optimum communication spanning tree(OCST) problem. Generally, OCST problem is known as NP-hard problem and recently, it is reveled as MAX SNP hard by Papadimitriou and Yannakakis. Nevertheless, many researchers have used polynomial approximation algorithm for solving this problem. This paper uses evolutionary algorithm. Especially, when an evolutionary algorithm is applied to tree network problem such as the OCST problem, representation and genetic operator should be considered simultaneously because they affect greatly the performance of algorithm. So, we introduce a new representation method to improve the weakness of previous representation which is proposed for solving the degree constrained minimum spanning tree problem. And we also propose a new decoding method to generate a reliable tree using the proposed representation. And then, for finding a suitable genetic operator which works well on the proposed representation, we tested three kinds of genetic operators using the information of network or the genetic information of parents. Consequently, we could confirm that the proposed method gives better results than the previous methods.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.39 no.6
• /
• pp.696-708
• /
• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.51 no.5
• /
• pp.11-19
• /
• 2014

In this paper, a low complexity parallel sphere decoder algorithm is proposed for high-order MIMO system. It reduces the computational complexity compared to the fixed-complexity sphere decoder (FSD) algorithm by static tree-pruning and dynamic tree-pruning using scalable node operators, and offers near-maximum likelihood decoding performance. Moreover, it also offers hardware-friendly node operation algorithm through fixing the variable computational complexity caused by the sequential nature of the conventional SD algorithm. A Monte Carlo simulation shows our proposed algorithm decreases the average number of expanded nodes by 55% with only 6.3% increase of the normalized decoding time compared to a full parallelized FSD algorithm for high-order MIMO communication system with 16 QAM modulation.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.20 no.1
• /
• pp.58-64
• /
• 2016

In this paper presents Robust k-means clustering-based high-speed bar code decoding method to blur and lighting. for fast operation speed and robust decoding to blur, proposed method uses adaptive local threshold binarization methods that calculate threshold value by dividing blur region and a non-blurred region. Also, in order to prevent decoding fail from the noise, decoder based on k-means clustering algorithm is implemented using area data summed pixel width line of the same number of element. Results of simulation using samples taken at various worst case environment, the average success rate of proposed method is 98.47%. it showed the highest decoding success rate among the three comparison programs.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.37 no.7A
• /
• pp.512-521
• /
• 2012

In this paper, we propose feasible combining rules for a decoding scheme of product codes to apply finite soft decision. Since the decoding scheme of product codes are based on complex tanh calculation with infinite soft decision, it requires high decoding complexity and is hard to practically implement. Thus, simple methods to construct look-up tables for finite soft decision are derived by analyzing the operations of the scheme. Moreover, we focus on using convolutional codes, which is popular for easy application of finite soft decision, as the horizontal codes of product codes so that the proposed decoding scheme can be properly implemented. Numerical results show that the performance of the product codes with convolutional codes using 4-bit soft decision approaches to that of same codes using infinite soft decision.

• Journal of the Institute of Convergence Signal Processing
• /
• v.2 no.2
• /
• pp.62-67
• /
• 2001

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need a large time delay over the reception process. Moreover, Turbo Code has been known as the robust ending method with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications such as IMT-2000. Therefore, in this paper, we proposed of the method to improve the conventional performance with the parallel concatenated 4-New Turbo Decoder using MAP a1gorithm in spite of complexity increasement. In the real-time video and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and fading channels.