• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.11
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• pp.1209-1216
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• 2009

This paper presents an efficient Quantized Equal Gain Transmission(QEGT) codebook index searching technique for MISO beamforming system in a Rayleigh flat fading channel. The searching time for the optimum weight vector among the codebook vectors increases exponentially when the codebook size increases linearly, although the bit error rate decreases. So, newly defined simple metric is proposed for fast searching, which determines a few candidates. Then the conventional method combined with accurate search algorithm selects the optimal index. This strategy significantly reduces the overall search time, while maintaining almost the same bit error rate performance. Furthermore, as the codebook size increases, the search time is considerably decreased compared to that of the conventional approach.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.7
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• pp.107-115
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• 1996

The average distortio of the vector quantizer is calcualted using a probability function F of the input source for a given codebook. But, since the input source is unknown in geneal, using the sample vectors that is realized from a random vector having probability function F, a time-average opeation is employed so as to obtain an approximation of the average distortion. In this case the size of the smple set should be large so that the sample vectors represent true F reliably. The theoretical inspection about the approximation, however, is not perfomed rigorously. Thus one might use the time-average distortion without any verification of the approximation. In this paper, the convergence characteristics of the time-average distortions are theoretically investigated when the size of sample vectors or the size of codebook gets large. It has been revealed that if codebook size is large enough, then small sample set is enough to obtain the average distortion by approximatio of the calculated tiem-averaged distortion. Experimental results on synthetic data, which are supporting the analysis, are also provided and discussed.

• Phonetics and Speech Sciences
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• v.1 no.2
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• pp.3-8
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• 2009

In an effort to improve speech recognition, we investigated the effect of the number of clusters. In usual LBG clustering, the number of codebook clusters is doubled on each bifurcation and hence cannot be chosen arbitrarily in a natural way. To have the number of clusters at our control, we combined adaptive resonance theory (ART2) with LBG and perform the clustering in two stages. The codebook thus formed was used in subsequent processing of fuzzy vector quantization (FVQ) and HMM for speech recognition tests. Compared to conventional LBG, our method was shown to reduce the best recognition error rate by 0${\sim$}0.9% depending on the vocabulary size. The result also showed that between 400 and 800 would be the optimal number of clusters in the limit of small and large vocabulary speech recognitions of isolated words, respectively.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.5
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• pp.747-755
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• 1995

When codebooks designed by a clustering algorithm using training sets, a time-averaged distortion, which is called the inside-training-set- distortion (ITSD), is usually calculated in each iteration of the algorithm, since the input probability function is unknown in general. The algorithm stops if the ITSD no more significantly decreases. Then, in order to test the trained codebook, the outside-training-set-distortion (OTSD) is to be calculated by a time-averaged approximation using the test set. Hence codebooks that yield small values of the OTSD are regarded as good codebooks. In other words, the calculation of the OTSD is a criterion to testing a trained codebook. But, such an argument is not always true if some conditions are not satisfied. Moreover, in order to obtain an approximation of the OTSD using the test set, it is known that a large test set is requared in general. But, large test set causes heavy calculation com0plexity. In this paper, from the analyses in [16], it has been revealed that the enough size of the test set is only the same as that of the codebook when codebook size is large. Then a simple method to testing trained codebooks is addressed. Experimental results on synthetic data and real images supporting the analysis are also provided and discussed.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.26 no.7
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• pp.627-635
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• 2015

We propose a dual-layer differential equal-gain codebook design methodology for LTE-Advanced(LTE-A), IEEE802.ac, and radar system having multiple transmit and receive antennas, and make computer simulations to evaluate its link-level performaces. M-ary phase shift keying constellation is used as its codeword elements to utilize low-cost power amplifiers at mobile stations. Especially, the proposed codebook can meet radar systems requirement for the high-powered equal-gain transmission property. Due to the temporal correlation of the adjacent channel, the proposed differential codebook can quantize only the differential information of the channel instead of the whole channel subspace, which virtually increase the codebook size to realize more accurate quantization of the channel. The proposed codebook has the same properties of LTE codebook that is, constant modulus, complexity reduction, and nested property. Computer simulations show that the proposed codebook performs better than the conventional 8-ary codebooks with the same amount of feedback information.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.583-589
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• 2004

Proposed is a speaker normalization method based on vector quantizer for continuous speech recognition (CSR) system in which no acoustic information is made use of. The proposed method, which is an improvement of the previously reported speaker normalization scheme for a simple digit recognizer, builds up a canonical codebook by iteratively training the codebook while the size of codebook is increased after each iteration from a relatively small initial size. Once the codebook established, the warp factors of speakers are estimated by comparing exhaustively the warped versions of each speaker's utterance with the codebook. Two sets of phones are used to estimate the warp factors: one, a set of vowels only. and the other, a set composed of all the Phonemes. A Piecewise linear warping function which corresponds to the estimated warp factor is adopted to warp the power spectrum of the utterance. Then the warped feature vectors are extracted to be used to train and to test the speech recognizer. The effectiveness of the proposed method is investigated by a set of recognition experiments using the TIMIT corpus and HTK speech recognition tool kit. The experimental results showed comparable recognition rate improvement with the formant based warping method.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.107-112
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• 1996

In this paper we propose a new fast codebook training algorithm for reducing the searching time of LBG algorithm. For each training data, the proposed algorithm stores the indexes of codewords that are close to that training data in the first iteration. It reduces computation time by searching only those codewords, the indexes of which are stored for each training data. Compared to one of the previous fast training algorithm, FSLBG, it obtains a better codebook with less exccution time. In our experiment, the performance of the codebook generated by the proposed algorithm in terms of peak signal-to-noise ratio(TSNR) is very close to that of LBG algorithm. However, the codewords to be searched for each training data of the proposed algorithm is only about 6%, for a codebook size of 256 and 1.6%, for a codebook size of 1.24, of LBG algorithm.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.24-31
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• 2015

A differential codebook using M-ary phase shift keying (M-PSK) constellation as its codeword elements, is proposed for Long term evolution (LTE), LTE-Advanced (LTE-A), and/or WiMAX systems. Due to the temporal correlation of the adjacent channel, the consecutive precoding matrices are likely to be similar. This approach quantize only the differential information of the channel instead of the whole channel subspace, which virtually increase the codebook size to realize more accurate quantization of the channel. Especially, the proposed codebook has the same properties of LTE release-8 codebook that is, constant modulus, complexity reduction, and nested property. The mobile station can be designed by using a less expensive non-linear amplifier utilizing the constant modulus property. Computer simulations show that the capacity of the proposed codebook performs better than LTE release-8 codebook with the same amount of feedback information.

• The Journal of the Acoustical Society of Korea
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• v.30 no.1
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• pp.1-8
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• 2011

This paper presents a method for improving the performance of ISF coefficients quantizer through compensating the defect of the split structure vector quantization using the ordering property of ISF coefficients. And design the ISF coefficients quantizer for wideband speech codec using proposed method. The wideband speech codec uses split structure vector quantizer which could not use the correlation between ISF coefficients fully to reduce complexity and the size of codebook. The proposed algorithm uses the ordering property of ISF coefficients to overcome the defect. Using the ordering property, the codebook redundancy could be figured out. The codebook redundancy is replaced by the adaptive-extended codebook to improve the performance of the quantizer through using the ordering property, ISF coefficient prediction and interpolation of existing codebook. As a result, the proposed algorithm shows that the adaptive-extended codebook algorithm could get about 2 bit gains in comparison with the existing split structure ISF quantizer of AMR-WB (G.722.2) in the points of spectral distortion.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2000.11a
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• pp.467-470
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• 2000

Vector quantization (VQ) is widely used in many high-quality and high-rate data compression applications such as speech coding, audio coding, image coding and video coding. When the size of a VQ codebook is large, the computational complexity for the full codeword search method is a significant problem for many applications. A number of complexity reduction algorithms have been proposed and investigated using such properties of the codebook as the triangle inequality. This paper proposes a new structured VQ search algorithm that is based on a multi-stage structure for searching for the best codeword. Even using only two stages, a significant complexity reduction can be obtained without any loss of quality.