• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.2
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• pp.27-37
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• 2022

In this paper, we presents a Multiple User Multiple Input Multiple Output (MU-MIMO) test-bed system for Time Division Duplex (TDD) Long Term Evolution-Advanced (LTE-A). Using two parameters, the condition number of the channel matrix and the path gain, the MU-MIMO system could switch pre-coder to maintain target Bit Error Rate (BER) level. This paper also introduces a calibration procedure for compensating error of Radio Frequency (RF) paths of the antennas and RF transceivers. From experimental measurements, dynamic pre-coding scheme could maintain target BER, set to 10^-3, with the pre-coder set configured with Zero Forcing (ZF), Tomlinson Harashima Pre-coding (THP), Lattice Reduction (LR). The simplest pre-coder ZF is adopted in stable channel, and when path gain become less than 0.25, LR is adopted. Lastly, when condition number of channel matrix become larger than 7, THP is adopted.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.605-608
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• 1999

A conventional image coder, such as JPEG, requires not only DCT and quantization but also additional pre-filtering under noisy environment. Since the pre-filtering removes camera noise and improves coding efficiency dramatically, its efficient implementation has been an important issue. Based on well-known noise removal techniques in image processing fields, this paper introduces an efficient scheme by adapting a noise removal procedure to block-based image coders. By using two-dimensional DCT factorization, the proposed image coder has only a modified DCT and a VLC, and performs pre-filtering and quantization simultaneously in the modified DCT operation.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.443-451
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• 2003

Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5C
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• pp.295-301
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• 2011

In [1], it has been shown that the energy spreading transform (EST) based iterative equalizer (IE) could enhance its performance by improving the reliability of the decision feedback symbols without the help of the complicated channel decoder. In the matched filter (MF) based IE proposed in [1], however, its feedforward filter (FFF) has been designed in the frequency domain while its feedback filter (FBF) in the time domain. So its complexity increases proportional to the channel memory length. To solve this problem, in this paper, both FFF and FBF are designed in the frequency domain. This enables the proposed frequency domain IE (FD-IE) to achieve the lower complexity over the conventional method in the highly dispersive channel. In addition, simulation results demonstrate that the BER performance of the proposed method is the same as the conventional.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.10
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• pp.11-20
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• 2014

In this paper, a novel transmission scheme is proposed to improve the data rates of spatial modulation (SM) which has low complexity and improves the spectral efficiency in correlated channel environments. The conventional SM scheme utilizes partial multiple antennas to transmit signal constellations and additional bits using antenna combinations. Therefore the channel capacity of SM is less than that of the conventional multiple input-multiple output (MIMO) scheme which uses all the available antennas. In this paper, an SM transmission scheme is proposed to improve the channel capacity using a tight frame pre-coder. The improvement in channel capacity of the SM scheme will be shown using computer simulations.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1E
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• pp.11-16
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• 2008

In this paper, a pitch cycle waveform (PCW) is extracted, quantized, and interpolated in a time domain to synthesize high-quality speech at low bit rates. The pre-alignment technique is proposed for the accurate and efficient PCW extraction, which predicts the current PCW position from the previous PCW position assuming that pitch periods evolve slowly. Since the pitch periods are different frame by frame, the original PCW is converted into the fixed-dimension PCW using the dimension-conversion method, and subsequently quantized by code-excited linear predictive (CELP) coding. The excitation signal for the linear predictive coding (LPC) synthesis filter is generated using the time-domain interpolation and interlink of the quantized PCW's. The coder operates at 4.2 kbit/s and 3.2 kbit/s depending on the pitch period. Informal listening test demonstrates the effectiveness of the proposed coding scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1430-1437
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• 1996

A low delay code excited linear predictive speech coder(LD-CELP) at bit rates under 8kbps is considered. We try to improve the perfomance of speech coder with frame type dependent modification of synthesis filter. We first classify frames into 3 groups: voiced, unvoiced and onset. For voicedand unvoiced frame, the spectral envelope of the synthesis filter is adapted to the phonetic characteristics. For transition frame from unvoiced to voiced, the synthesis filter which has been interpolated with the bias filter is used. The proposed vocoder produced more clear sound with similar delay level than other pre-existing LD-CELP vocoders.

• The Journal of Engineering Research
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• v.2 no.1
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• pp.57-62
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• 1997

Two psychoacoustic models used in MPEG-1 employ different masking patterns, different masking indexes, and different computational procedures. As a result, Model 1 is inferior to Model 2 due to its worst case approach in computing the SMR even though it determines tonality and masking levels accurately. In this study, we investigate the performances of psychoacoustic models when we modify the MPEG-1 audio coder for variable bitrates. Simulation results show that Model 2 has a gain of 30 kbps in the dual channel mode and 20 kbps in the joint stereo mode. It is generally known that the joint stereo mode has a gain in bitrate compare to the dual channel mode. For signals with frequent attacks, this gain becomes larger in Model 1 than in Model 2. This is due to the fact that Model 1 uses the worst case approach in computing the SMR to reduce pre-echo

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2004.11a
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• pp.39-42
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• 2004

본 논문은 H.264 동영상 표준 부호화 방식의 성능 향상을 위한 잔여 신호 전처리 필터에 대해 제안한다. 영상의 화질에 대한 최종적인 판단은 인간의 시각에 의하므로 HVS(Human Visual System)을 영상 압축에 적용한 수 있다(1,2,3). 잔여 신호 전처리 필터는 원 영상으로부터 중요하지 않은 부분을 제거하여 영상의 화질을 주어진 비트율로 최대화 시켜 부호화 율을 향상시키는데 목적을 둔다(4). 부호화 방식의 형태에 따라 잔여신호(DFD: Displaced Frame Difference)에 잔여 신호 전처리 필터 알고리즘을 적용하여 노이즈를 제거하고 전송 비트율을 감소시킬 수 있다 제안된 방식의 성능을 실험 결과로부터 확인할 수 있다

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9C
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• pp.877-886
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• 2005

Various frame recovery algorithms have been suggested to overcome the communication quality degradation problem due to Internet-typical impairments on Voice over IP(VoIP) communications. In this paper, we propose a new receiver-based recovery method which is able to enhance recovered speech quality with almost free computational cost and without an additional increment of delay and bandwidth consumption. Most conventional recovery algorithms try to recover the lost or erroneous speech frames by reconstructing missing coefficients or speech signal during speech decoding process. Thus they eventually need to modify the decoder software. The proposed frame recovery algorithm tries to reconstruct the missing frame itself, and does not require the computational burden of modifying the decoder. In the proposed scheme, the Vector Quantization(VQ) codebook indices of the erased frame are directly estimated by referring the pre-computed VQ Codebook Index Interpolation Tables(VCIIT) using the VQ indices from the adjacent(previous and next) frames. We applied the proposed scheme to the ITU-T G.723.1 speech coder and found that it improved reconstructed speech quality and outperforms conventional G.723.1 loss recovery algorithm. Moreover, the suggested simple scheme can be easily applicable to practical VoIP systems because it requires a very small amount of additional computational cost and memory space.