• Journal of Broadcast Engineering
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• v.14 no.2
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• pp.178-188
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• 2009

In this paper, we propose the method which provides efficient sub-channel allocation for handoff and new call supporting multi-media service in AMC-based OFDMA system. Firstly, we apply the multi-band method which provides different AMC method according to the location of user terminals. Also, in OFDMA system environment that a base station has a lot of sub-channels, we adopt the sub-channel allocation scheme that provides a higher priority to handoff call and real-time service about handoff and new calls with multi-meida service. The simulation results show that the proposed scheme plays a role in increasing the number of new and handoff calls meeting the required blocking rate.

• ETRI Journal
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• v.23 no.4
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• pp.177-189
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• 2001

Multicarrier DS-CDMA is an effective approach to combat fading and various kinds of interference. In this paper, we present an overlapped multicarrier DS-CDMA system, wherein each of the rate 1/M convolutionally-encoded symbols is also repetition coded and transmitted using overlapped multicarriers. However, since the frequency spectrums of successive carriers are allowed to overlap, the transmission bandwidth is more efficiently utilized. The effect of the overlapping percentage between successive carriers of a multicarrier DS-CDMA system on the performance is investigated to determine the overlapping percentage showing the best performance. We suggest two methods for sub-band overlapping variation. One is to allow variation of sub-band overlapping percentage when the total number of subcarriers is fixed. The other is to increase the number of sub-bands (the number of repetitions R) with fixed sub-band bandwidth. Given a total number of subcarriers MR, we show that the BER variation is highly dependent on the roll-off factor ${\beta}$ of a raised-cosine chip wave-shaping filter irrespective of convolutional encoding rate 1/M and repetition coding rate 1/R. We also analyze the possibility of reduction in total multi-user interference by considering the variation of both the roll-off factor ($0<{\beta}{\leq}1$) and the sub-band overlapping factor ($0<{\lambda}{\leq}2$), and show that the proposed system may outperform the multicarrier DS-CDMA system in [3].

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.216-219
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• 2003

In this paper, we present performance analysis of an IEEE P802.15.3a high rate wireless personal area network transceiver. This physical layer standard uses QOSK as its sub-channel modulation scheme and orthogonal frequency domain modulation (OFDM) for sub-bands. OFDM is used for each sub-band so that multi-path effects are absorbed by equalizer and guard, and fading can be approximately modeled as additive white Gaussian noise. In multi-band ultra-wideband system, DAC quantization error is important noise source since high resolution conversion cannot be used due to high power consumption. Simulation result shows that, to get 640-Mbps throughput, at least 5-bits precision is necessary to maintain bit-error rate under 10$\^$-2/, which can be lowered, with channel coding, to 10$\^$-6/ that is the bit-error rate required by IEEE 802.15 upper protocol layer, in 4-meter LOS fading channel.

• ETRI Journal
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• v.39 no.3
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• pp.398-405
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• 2017

In this paper, an artificial stereo extension method that creates stereophonic sound from a mono sound source is proposed. The proposed method first trains deep neural networks (DNNs) that model the nonlinear relationship between the dominant and residual signals of the stereo channel. In the training stage, the band-wise log spectral magnitude and unwrapped phase of both the dominant and residual signals are utilized to model the nonlinearities of each sub-band through deep architecture. From that point, stereo extension is conducted by estimating the residual signal that corresponds to the input mono channel signal with the trained DNN model in a sub-band domain. The performance of the proposed method was evaluated using a log spectral distortion (LSD) measure and multiple stimuli with a hidden reference and anchor (MUSHRA) test. The results showed that the proposed method provided a lower LSD and higher MUSHRA score than conventional methods that use hidden Markov models and DNN with full-band processing.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.4
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• pp.429-438
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• 1986

In this paper we have designed and implemented a real-time, full-duplex SBC (sub-band coding) codec at 16kbps using a high speed digital signal processor, NEC 7720. The SBC codec employs a QMF(quadrature mirror filter) filter bank based on the tree structures of two-band analysis-synthesis pairs to partition speech signal into 4 octabe bands. Computer simulation has been done to investigate the effect of fixed-point computation of the NEC 7720. Three different performance measures, the conventional signal-to-noise ratio, the informal listening test, and an LPC(linear predictive coding)distance measure, have been used in this simulation. The necessary parameters have been optimized through the simulation. The developed hardware and software have been tested in real-time operation using a hardware emulator.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.556-562
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• 2013

This paper is aimed to improve performance of Band-Selection speech/audio Coder reconstucted band spectrum that is not sent by the comfort noise. To improve the performance, we use the Spectral Band Replication(SBR) technique instead of substitution of Comfort noise. To synthesize SBR signal, the SBR algorithm is referenced in selected signals and the spectrum synthesized by SBR is injected to non-selected band. Each sub-band spectrum has been energy-weighted by real audio signal. We propose the enhanced the Band-Selection Coder that utilizes synthesized SBR signal from selected signal instead of comfort noise.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.62-68
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• 2010

This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.

• Journal of Korea Society of Digital Industry and Information Management
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• v.7 no.4
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• pp.119-131
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• 2011

In this paper, we explore the application of 2-D dual-tree discrete wavelet transform (DDWT), which is a directional and redundant transform, for image coding. DDWT main property is a more computationally efficient approach to shift invariance. Also, the DDWT gives much better directional selectivity when filtering multidimensional signals. The dual-tree DWT of a signal is implemented using two critically-sampled DWTs in parallel on the same data. The transform is 2-times expansive because for an N-point signal it gives 2N DWT coefficients. If the filters are designed is a specific way, then the sub-band signals of the upper DWT can be interpreted as the real part of a complex wavelet transform, and sub-band signals of the lower DWT can be interpreted as the imaginary part. The quincunx lattice is a sampling method in image processing. It treats the different directions more homogeneously than the separable two dimensional schemes. Quincunx lattice yields a non separable 2D-wavelet transform, which is also symmetric in both horizontal and vertical direction. And non-separable wavelet transformation can generate sub-images of multiple degrees rotated versions. Therefore, non-separable image processing using DDWT services good performance.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.4
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• pp.491-501
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• 1990

Recently, four speech coding techniques, namely, SBC-APCM(sub-band coding adaptive PCM), RPE-LPC(regualr pulse excitation linear predictive codec), MPE-LTP(multi-pulse excited long-term prediction) and CELP (code-excited linear prediction) are proposed for digital mobile radio applications. However, a performance comparison of these coders in the Rayleigh fading environment has not been made yet. In this paper, the performances of the four spech coders in the random bit error and burst error environment are investigated. For the channel coding of SBC-APCM, RPE-LPC and MPE-LTP, the sensitivity of output bit stream is measured and a bit selective forward error correction is provided acording to the measured bit sensitivity. And for an attempt to improve the performance of CELP, an optimum quantizer is applied for transmitting scalar quantities in CELP. However, an improvement over the conventional approach is found to be negligible. For the channel coding of CELP, Reed-Solomon code, Golay code, convolutional code of rate 1/2 shows the best performance. Finally, from the simulation results, it is concluded that CELP is the best candidate for digital mobile radio and is followed by MPE-LTP, SBC-APCM and RPE-LPC.

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