• The Journal of the Acoustical Society of Korea
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• v.16 no.1E
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• pp.41-48
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• 1997

The very purpose of subband codec is the attainment of data rate compression through the use of quantizer and optimum bit allocation for each decimated signal. Yet the question of floating-point quantization effects in subband codec has received scant attention. There has been no direct focus on the analysis of quantization errors, nor on design with quantization errors embedded explicitly in the criterion. This paper provides a rigorous theory for the modelling, analysis and optimum design of the general M-band subband codec in the presence of the floating-point quantization noise. The floating-point quantizers are embedded into the codec structure by its equivalent multiplicative noise model. We then decompose the analysis and synthesis subband filter banks of the codec into the polyphase form and construct an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed output. The optimum design criteria of the subband codec is given to the design of the analysis/synthesis filter bank and the floating-point quantizer to minimize the output mean square error. Specific optimum design examples are developed with two types of filter of filter banks-orthonormal and biorthogonal filter bank, along with their perpormance analysis.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.9
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• pp.1219-1228
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• 1995

In this paper, we propose a dynamic bandwidth allocation scheme for the transmission of VBR video source through ATM based BISDN. In order to efficiently evaluate the required bandwidth, the characteristics of the VBR video sources generated by the MPEG coder are analyzed with variations in the number of GOP and quantizer scale. The required bandwidth for VBR video source is estimated by a prediction algorithm using scene and frame correlations as well as the statistical properties of the VBR video source. Scene correlation represents a strong correlation among the adjacent slices in a frame, and frame correlation represents a strong correlation among the frames. The statistical properties based on the traffic parameters of video sources such as peak rate, average rate, and standard deviation of source are determined through simulation. The performance of the proposed bandwidth allocation scheme is evaluated in terms of the over-allocated bandwidth, bandwidth utilization, and cell loss rate with variations in the quantizer scale and the number of GOP. It is shown that the results of simulation based on the proposed scheme are superior to those of the conventional methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.2C
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• pp.100-109
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• 2003

A differential index assignment scheme is proposed for the image encoding system in which a variable-length tree-structured vector quantizer is adopted. Each source vector is quantized into a terminal node of VLTSVQ and each terminal node is represented as a unique binary vector. The proposed index assignment scheme utilizes the correlation between interblocks of the image to increase the compression ratio with the image quality maintained. Simulation results show that the proposed scheme achieves a much higher compression ratio than the conventional one does and that the amount of the bit rate reduction of the proposed scheme becomes large as the correlation of the image becomes large. The proposed encoding scheme can be effectively used to encode R images whose pixel values we, in general, highly correlated with those of the neighbor pixels.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1461-1470
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• 1998

Bit allocation or quantizer assigning problem is a basic and essential issue in lossy picture coding. It could be represented as minimizing overall distortion with the given constraint that total bits should not exceed allowed bit-budget. Optimal solution can be found by Lagrangian method. However this method needs much computational time and memory. This paper presents an approximation method that uses the activity measure. The comparison between the existing activity measuring techniques are made, and mapping function from activity value to the quantizer is proposed. Under MPEG-1 Intra coding situation, simulations show almost identical results compared to the optimal ones obtained by Lagrangian method with reduced computational time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38C no.6
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• pp.540-549
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• 2013

An adaptive non-uniform quantization scheme is proposed for soft-decision error correction in NAND flash memory. Even though the conventional maximizing mutual information (MMI) quantizer shows the optimal post-FEC (forward error correction) bit error rate (BER) performance, this quantization scheme demands heavy computational overheads due to the exhaustive search to find the optimal parameter values. The proposed quantization scheme has a simple structure that is constructed by only six parameters, and the optimal values of them are found by maximizing the mutual information between the input and the output symbols. It is demonstrated that the proposed quantization scheme improves the BER performance of soft-decision decoding with only small computational overheads.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.36-40
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• 1997

In speech signal processing, it is necessary and important to detect exactly the pitch. The algorithms of pitch extraction which have been proposed until now are difficult exactly pitches over wide range speech signals. In this paper, thus, we proposed a new pitch detection algorithm that finds the fundamental period of speech signal in the residual signal quantized by the uniform quantizer as PCM. The proposed method shows little gross error of average 0.25% for clean speech and average 3.39% for SNR of 0dB. It also achieves results of the pitch contours, improving the accuracy of pitch detection in transient phonemes and noise environments.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.10-15
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• 1997

In this paper, an efficient LSF quantizer, named improved PSVQ(IPSVQ), is proposed to apply in the 8 kbps QCELP speech coder. By using 27 bits IPSVQ instead of 40 bits DPCM quantizer per frame, we can save 13 bits/frame and allocate those bits to the codebook gain and the pitch gain parameters. Hence we improve the overall performance of the QCELP codec. The enhanced QCELP shows the performance improvement of 0.9 dB SNR and 0.4 dB SEGSNR. Informal listening tests also confirm the improvement in the speech quality.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.405-408
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• 2005

This paper proposed the method to images of losses using restorable wavelet transformation. The algorithm proposed in this work stars by processing the pre-quantizer on the original images to organize an image that matches the gray level. The wavelet transformation filter to the original image which is already pre-quantized in order to segment bands. Considering the lowest coding of bands influencing the most to the overall condition of the reconstructed image, it only uses the Huffman coding using prediction. Reconstructed images by proposed algorithm showed higher PSNR when coding images of JPEG or non pre-quantized images. Applying pre-quantizer can control the peak errors and is expected to be useful at mass image compression.

• The KIPS Transactions:PartB
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• v.12B no.4 s.100
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• pp.437-442
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• 2005

For the purpose of speaker normalization in speaker independent speech recognition systems, experiments are conducted on a method based on Gaussian mixture model(GMM). The method, which is an improvement of the previous study based on vector quantizer, consists of modeling the probability distribution of canonical feature vectors by a GMM with an appropriate number of clusters, and of estimating the warp factor of a test speaker by making use of the obtained probabilistic model. The purpose of this study is twofold: improving the existing ML based methods, and comparing the performance of what is called 'soft decision' method with that of the previous study based on vector quantizer. The effectiveness of the proposed method is investigated by recognition experiments on the TIMIT corpus. The experimental results showed that a little improvement could be obtained tv adjusting the number of clusters in GMM appropriately.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.1
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• pp.113-118
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• 2001

In this paper, we implement modules of the MPEG-2 AAC decoder using VHDL. Tools of Huffman decoder, inverse quantizer and high-density filter bank which are necessary for the AAC decoder. We designed the high speed Huffman decoder using the method of octal tree search algorithm, and reduced computational time of filter bank using IFFT. Also, we use table of computation result for an exponential calculation of Inverse quantizer in fixed-point hardware, and reduced the size of table using linear interpolation. Modules implemented by hardware through optimization work in real time at low clock frequency are possible to reduce the system size.