• The Journal of the Acoustical Society of Korea
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• v.29 no.3E
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• pp.131-139
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• 2010

This paper proposes a novel spectral hole substitution technique for low bit-rate audio coding. The spectral holes frequently occurring in relatively weak energy bands due to zero bit quantization result in severe quality degradation, especially for harmonic signals such as speech vowels. The enhanced aacPlus (EAAC) audio codec artificially adjusts the minimum signal-to-mask ratio (SMR) to reduce the number of spectral holes, but it still produces noisy sound. The proposed method selectively predicts the spectral shapes of hole bands using either intra-band correlation, i.e. harmonically related coefficients nearby or inter-band correlation, i.e. previous frames. For the bands that have low prediction gain, only the energy term is quantized and spectral shapes are replaced by pseudo random values in the decoding stage. To minimize perceptual distortion caused by spectral mismatching, the criterion of the just noticeable level difference (JNLD) and spectral similarity between original and predicted shapes are adopted for quantizing the energy term. Simulation results show that the proposed method implemented into the EAAC baseline coder significantly improves speech quality at low bit-rates while keeping equivalent quality for mixed and music contents.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.11
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• pp.102-113
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• 1997

In this paepr, a new image segmentation method reducing efficiently contour information causing bottleneck problem at segmentatio-based very low bit rate codingis proposed, while preserving objective and subjective quality. It consists of 4-level hierarchical image segmentation based on mathematical morphology and 1-leve region merging structure using contast of two adjacent regions. For two adjacent region pairs at the fourth level included in each region of the thid level, contrast is calculated. Among the pairs of two adjacent regions with less value than threshold, two adjacent regions having the minimum contrast are merged first. After region merging, texture of the merged region is updated. The procedure is performed recursively for all the adjacent region pairs at the fourth level included in each region of the third level. Compared with the previous method, the objective and subjective image qualities are similar. But it reduces 46.65% texture information on the average by decreasing total region number to be tansmitted. Specially, it shows reduction of the 23.95% contour information of the average. Thus, it can improve efficiently the bottleneck problem at segementation-based very low bit rate coding.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.17 no.8
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• pp.2157-2169
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• 2023

In order to handle the problem that puncture patterns will change the position distribution of original information bits and frozen bits in polar codes, which affects performance of puncture polar codes further, a minimum row weight and polar spectrum based puncture polar codes construction algorithm (called PA-MRWP) is proposed in this paper. The algorithm calculates row weight of generator matrix and sorts the row weight in ascending order first. Next, the positions with the minimum row weight are selected as initial puncture positions. If the rows with the same row weight cannot all be punctured, polar spectrum based auxiliary puncture scheme is used. In sub-channels with the same row weight, rows corresponding to the polarized sub-channels with higher reliability are selected as puncture positions to construct puncture vector, and the reliability is calculated based on polar spectrum. It is actually a two-step selection strategy, the proposed minimum row weight puncture (MRWP) algorithm is used for primary selection and polar spectrum based auxiliary puncture is used for adjustment. Simulation results show that, compared with worst quality puncture (WQP) algorithm, the proposed PA-MRWP algorithm and Gaussian approximation-aided minimum row weight puncture (GA-MRWP) algorithm provide gains of about 0.46 dB and 0.29 dB at bit error rate (BER) of 10^-4, respectively when code length N=400, code rate R=1/2. In addition, the proposed puncture algorithms improve the BER performance significantly with respect to quasi-uniform puncture (QUP) algorithm.

• International Journal of Naval Architecture and Ocean Engineering
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• v.13 no.1
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• pp.617-627
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• 2021

Underwater Acoustic Channels (UAC) have inherent sparse characteristics. The traditional adaptive equalization techniques do not utilize this feature to improve the performance. In this paper we consider the Variable Adaptive Subgradient Projection (V-ASPM) method to derive a new sparse equalization algorithm based on the Minimum Symbol Error Rate (MSER) criterion. Compared with the original MSER algorithm, our proposed scheme adds sparse matrix to the iterative formula, which can assign independent step-sizes to the equalizer taps. How to obtain such proper sparse matrix is also analyzed. On this basis, the selection scheme of the sparse matrix is obtained by combining the variable step-sizes and equalizer sparsity measure. We call the new algorithm Sparse-Control Proportional-MSER (SC-PMSER) equalizer. Finally, the proposed SC-PMSER equalizer is embedded into a turbo receiver, which perform turbo decoding, Digital Phase-Locked Loop (DPLL), time-reversal receiving and multi-reception diversity. Simulation and real-field experimental results show that the proposed algorithm has better performance in convergence speed and Bit Error Rate (BER).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.12A
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• pp.941-949
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• 2011

In a cooperative communication system with a source node and multiple relays equipping single antenna and a destination node equipping multiple antennas, the selective cooperative spatial multiplexing scheme can obtain spatial multiplexing gain and additional selection diversity gain. But it can degrade a bit error rate performance because some received symbols forwarded from particular relays may be lost by attenuation due to path-loss. We propose a relay and transmission mode selection scheme which selects minimum number of multiple relays having the channel capacity larger than a given data rate and transmission mode which switches spatial multiplexing and spatial diversity mode in cooperation phase to enhance the bit error rate performance. The proposed scheme achieves 1.5~2dB gain at the low SNR range compared with the conventional scheme by obtaining additional spatial diversity gain.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.7
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• pp.2338-2353
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• 2015

In this paper, the optimal power allocation algorithm that minimizes the aggregate bit error rate (BER) of the secondary user (SU) in a downlink orthogonal frequency division multiplexing (OFDM) based cognitive radio (CR) system, while subjecting to the interference power constraint and the transmit power constraint, is investigated under the assumption that the instantaneous channel state information (CSI) of the interference links between the secondary transmitter and the primary receiver, and between the primary transmitter and the secondary receiver is perfectly known. Besides, a suboptimal algorithm with less complexity is also proposed. In order to deal with more practical situations, we further assume that only the channel distribution information (CDI) of the interference links is available and propose heuristic power allocation algorithms based on bisection search method to minimize the aggregate BER under the interference outage constraint and the transmit power constraint. Simulation results are presented to verify the effectiveness of the proposed algorithms.

• Electrical & Electronic Materials
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• v.9 no.3
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• pp.246-250
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• 1996

In optical soliton transmission systems with nonlinear dispersive materials, which is utilized for ultra-long and high bit rate transmission, it is shown that the value of initial time difference between adjacent solitons is analyzed for optimum bit rate. The method is inducted by uncorrelation condition with minimum interaction forces in initial covariance coefficient between adjacent solitons. When the initial time difference is 6 times of soliton pulse width by the results, it is shown that the reliability is maintained with more than 90% within transmission length of soliton period. multiplied by 93.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.5
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• pp.912-919
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• 1994

A new error control line code based on an extended Hamming code is proposed and its performance is analyzed in this paper. The proposed code is capable of single error correction and double error detection since its minimum Hamming distance is 4. In addition, the error detection capability can be oncreased due to the redundancy bit used for line coding. As a result, the proposed code shows lower code rate, but better spectral characteristics in low frequency region and lower residual bit error rate than the conventional error correction line code using Hamming (7, 4) code.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.63-67
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• 1997

The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.

• Proceedings of the KSRS Conference
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• 2007.10a
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• pp.537-539
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• 2007

Telemetry processing system requires minimum bit transition level in data streams to maintain a bit synchronization while receiving telemetry signal. PN code has a capability of providing the bit transition and is widely used in the packet communication of CCSDS. CCSDS PN code that generator polynomial is $h(x)=x^{8}+x^{7}+x^{5}+x^{3}+1$, and the random bit sequence that is generated from this polynomial is repeated with the cycle of 255 bits. As the resolution of satellite image increases, the size and transmission rate of data increases. To process of huge and bulky size of satellite image, the speed of CCSDS PN Processing is very important. This paper introduces the way of improving the CCSDS PN Processing speed through processing 128 bits at one time using the feature of cyclic structure that repeats after first 255 bytes by grouping the random bit sequence with 1 byte and Intel Streaming SIMD Extensions 2. And this paper includes the comparison data of processing speed between SSE2-applied implementation and not-applied implementation, in addition, the measured value of speed improvement.