• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.7
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• pp.119-124
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• 2015

Speech enhancement has been required as a preprocessor for a noise robust speech recognition system. Codebook-based Speech Enhancement (CBSE) is highly robust in nonstationary noise environments compared with conventional noise estimation algorithms. However, its performance is severely degraded for the codevector combinations that have lower correlation with the input signal since CBSE depends on the trained codebook information. To overcome this problem, only the reliable codevector combinations are selected to be used to remove the codevector combinations that have lower correlation with input signal. The proposed method produces the improved performance compared to the conventional CBSE in terms of Log-Spectral Distortion (LSD) and Perceptual Evaluation of Speech Quality (PESQ).

• Journal of Broadcast Engineering
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• v.28 no.1
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• pp.141-144
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• 2023

Recently, a joint channel estimation, channel quantization, feedback, and precoding system based on deep-neural network (DNN) was proposed. The corresponding system achieved a joint optimization based on deep learning such that it achieved a higher sum rate than the existing codebook-based precoding systems. However, this DNN-based procoding system is not directly applicable for the environments with many users such that a specific user selection can potentially increase the sum rate of the system. Thus, in this letter, we study an appropriate user selection method suitable for DNN-based precoding.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.17 no.12 s.115
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• pp.1224-1230
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• 2006

In this paper, for spatial multiplexing with limited feedback, a precoding scheme is proposed based on the joint use of minimal instantaneous feedback and long-term feedback of a small number of bits, wherein the long-term feedback is used to convey a selected preceding matrix within a precodercodebook consisting of a number of unitary matrices, and the active column vectors of the selected unitary matrix are conveyed to the transmitter using instantaneous feedback. Focusing on the case of dual multi-input multi-output(MIMO) systems, precoder codebook design for maximizing the average throughput of a spatial multiplexing system with a zero-forcing(ZF) receiver is proposed. It is shown that the proposed scheme provides a considerable throughput enhancement over multi-mode antenna selection and multi-mode basis selection only with the additional long-yterm feedback of a small number of bits. For example, the throughput increases by 11.5 % than antenna selection and 5.1% than basis selection, respectively, when SNR=20 dB.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.7 s.361
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• pp.59-69
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• 2007

In this paper, we propose a MIMO-OFDMA (Multi Input Multi Output-Orthogonal Frequency Division Multiple Access) system based on Grassmannian beamforming for performance improvement of downlink real-time traffic transmission in harsh channel conditions with low CIR (Carrier-to-Interference Ratio). In the proposed system to reduce feedback information for the beamforming, we also apply Grassmannian Beamforming. Furthermore, we propose antenna selection scheme which performs the beamforming with more useful transmit antennas. In the proposed system, the optimal combination of transmit antennas with maximum MRT (Maximum Ratio Transmission) beamforming gain, is selected. Simulation results reveal that the proposed MIMO-OFDMA system achieves significant improvement of spectral efficiency in low CIR region as compared to a typical open-loop MIMO-OFDMA system using pseudo-orthogonal space time block code.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.40 no.4
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• pp.351-359
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• 1991

This paper describes performances of speaker adaptation using a probabilistic spectral mapping matrix in hidden-Markov model(HMM) -based Korean isolated word recognition. Speaker adaptation based on probabilistic spectral mapping uses a well-trained prototype HMM's and is carried out by Viterbi, dynamic time warping, and forward-backward algorithms. Among these algorithms, the best performance is obtained by using the Viterbi approach together with codebook adaptation whose improvement for isolated word recognition accuracy is 42.6-68.8 %. Also, the selection of the initial values of the matrix and the normalization in computing the matrix affects the recognition accuracy.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.7
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• pp.1-5
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• 2009

Random beanforming exploits multiuser diversity gain in static channels. Since the gain is restricted by the user population, some extended works have been proposed. Among them, a codebook-based opportunistic beamforming technique forms multiple random beams with small pilots. The technique however has difficulty in designing beams flexibly by the channel statistics. In this paper, we propose a technique forming the multiple random beams by rotating codebooks. The proposed technique enables the flexible design of beams so that multiuser diversity and beam selection diversity are exploited simultaneously with small pilots robust to the channel statistics.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1E
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• pp.46-55
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• 1995

In this paper, we present a new wavelet transform-based CELP coder, called band selection wavelet transform CELP (BS-WTCELP) operated at 4.8 kbps. The proposed algorithm uses a band selection scheme of frequency bands of wavelet transform and selective vector quantization (VQ). The band selection and selective VQ structure is implemented by using a classified VQ structure. The proposed algorithm has about 0.5-1.0 dB improvement in segmental SNR compared with the conventional CELP that uses the random codebook search, while is has significantly reduced computational and storage complexity. Many experimental results have shown that the proposed algorithm is more suitable for most real-applications than the conventional CELP and wavelet transform CELP.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.11
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• pp.4623-4643
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• 2015

This paper considers physical-layer security protocols in multicast cognitive radio (CR) networks. In particular, we propose dual-hop cooperative decode-and-forward (DF) and randomize-and-forward (RF) schemes using partial relay selection method to enhance secrecy performance for secondary networks. In the DF protocol, the secondary relay would use same codebook with the secondary source to forward the source's signals to the secondary destination. Hence, the secondary eavesdropper can employ either maximal-ratio combining (MRC) or selection combining (SC) to combine signals received from the source and the selected relay. In RF protocol, different codebooks are used by the source and the relay to forward the source message secretly. For each scheme, we derive exact and asymptotic closed-form expressions of secrecy outage probability (SOP), non-zero secrecy capacity probability (NzSCP) in both independent and identically distributed (i.i.d.) and independent but non-identically distributed (i.n.i.d.) networks. Moreover, we also give a unified formula in an integral form for average secrecy capacity (ASC). Finally, our derivations are then validated by Monte-Carlo simulations.

• Journal of the Korean Institute of Intelligent Systems
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• v.8 no.2
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• pp.60-69
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• 1998

This paper proposes a speech synthesis method using Fuzzy VQ, and then study how to make choice of fuzziness value which optimizes (controls) the performance of FVQ in order to obtain the synthesized speech which is closer to the original speech. When FVQ is used to synthesize a speech, analysis stage generates membership function values which represents the degree to which an input speech pattern matches each speech patterns in codebook, and synthesis stage reproduces a synthesized speech, using membership function values which is obtained in analysis stage, fuzziness value, and fuzzy-c-means operation. By comparsion of the performance of the FVQ and VQ synthesizer with simmulation, we show that, although the FVQ codebook size is half of a VQ codebook size, the performance of FVQ is almost equal to that of VQ. This results imply that, when Fuzzy VQ is used to obtain the same performance with that of VQ in speech synthesis, we can reduce by half of memory size at a codebook storage. And then we have found that, for the optimized FVQ with maximum SQNR in synthesized speech, the fuzziness value should be small when the variance of analysis frame is relatively large, while fuzziness value should be large, when it is small. As a results of comparsion of the speeches synthesized by VQ and FVQ in their spectrogram of frequency domain, we have found that spectrum bands(formant frequency and pitch frequency) of FVQ synthesized speech are closer to the original speech than those using VQ.

• Journal of Satellite, Information and Communications
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• v.11 no.3
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• pp.133-137
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• 2016

Millimeter wave wireless communication systems, especially those targeting indoor high rate data transfer, have a strong requirement for high quality wireless link. Unfortunately, in this frequency band, the electromagnetic wave has to sustain the high propagation loss caused by the smaller wavelengths. In this scenario, beamforming technique, which enhances the link quality by focusing the radiation power on a direction, becomes one of the most important techniques in millimeter wave band wireless communication. In recent year, there been conducted many research on beamforming to improve the performance of wireless system. In this paper, we evaluate the performance of a simplified codebook-based beamforming scheme which is based on multiple-procedure and three-state beam selection. The simplified scheme significantly reduces beamforming setup time, comparing to the exhaustive searching, two-level searching adopted in IEEE 802.15.3c standard, and also conventional multi-level scheme.