• The KIPS Transactions:PartC
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• v.8C no.1
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• pp.105-112
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• 2001

In this paper, we present the advanced QCELP codebook search improving the qualification of speech, which can make QCELP vocoder used in noise robust system. While conventional QCELP usually searches stochastic codebook once, we can find that two times search is the most suitable for improving the quality of speech after we did 2-5 times search. Consequently, the advanced QCELP vocoder represents excitation signal in detail using two times precise quantization and so improve the qualification of speech. In our experiment, we use the speeches collected from circumstance (such as lecture room, house, street, laboratory etc.) without regarding noise as input dat and measure the speech Qualification using SNR, segSNR. As the result of the experiment, we find that the advanced QCELP makes SNR and segSNR improved by 38.35% and 65.51% respectively compared with conventional QCELP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.8
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• pp.1587-1594
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• 1994

The pitch predictor is one of the most important part for the robust tree coder. The hybrid backward pitch adapation which is a combination of a block adaptation and a recursive adaptation is used for the pitch predictor. In order to improve the error performance and track the pitch period change of the input speech, it is proposed to smooth the input of the pitch predictor. The smoother with three taps can have fixed coefficients or variable coefficients depending on the estimated autocorrelation function of the output of the pitch synthesizer. The inclusion of a variable smoother can track the pitch period change within a block and reduce the effect of channel errors.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.90-94
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• 1996

Code Excited Linear Prediction(CELP) vocoder exhibits good performance at data rates below 8 kbps. The major drawback of CELP type coders is a large amount of computation. In this paper, we propose a new pitch searching method that preserves the quality of the CELP vocoder reducing computational complexity. The basic idea is that grasps preliminary pitches using the first formant of speech signal and performs pitch search only about the preliminary pitches. As applying the proposed method to the CELP vocoder, we can reduce complexity by 64% in the pitch search.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.2
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• pp.62-67
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• 2001

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need a large time delay over the reception process. Moreover, Turbo Code has been known as the robust ending method with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications such as IMT-2000. Therefore, in this paper, we proposed of the method to improve the conventional performance with the parallel concatenated 4-New Turbo Decoder using MAP a1gorithm in spite of complexity increasement. In the real-time video and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and fading channels.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.13-16
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• 2000

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need great many delay over the reception process. Moreover, Turbo Code has been known as the robust coding methods with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications the same as IMT-2000. Therefore, in this paper, we proposed of that has the better performance than existing Turbo Decoder that has the parallel concatenated four-step structure using MAP algorithm. In the real-time voice and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and lading channels.

• The Journal of the Acoustical Society of Korea
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• v.23 no.4
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• pp.333-339
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• 2004

This Paper proposes the vector quantizer-pyramid vector quantizer(VQ-PVQ) structure. Also both predictive structure and safety-net concept are combined into the VQ-PVQ to quantize the IPC parameter of wideband speech codec. The Performance is compared to the LPC vector quantizer used in the AMR-WB(ITU-T G.722.2). demonstrating reduction in both spectral distortion and encoding memory.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.304-309
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• 2002

We propose the computation reduction method of real root method that is mainly used in the CELP (Code Excited Linear Prediction) vocoder. The real root method is that if polynomial equations have the real roots, we are able to find those and transform them into LSP. However, this method takes much time to compute, because the root searching is processed sequentially in frequency region. In this paper, to reduce the computation time of real root, we compare the real root method with two methods. In first method, we use the mal scale of searching frequency region that is linear below 1 kHz and logarithmic above. In second method, The searching frequency region and searching interval are ordered by each coefficient's distribution. In order to compare real root method with proposed methods, we measured the following two. First, we compared the position of transformed LSP (Line Spectrum Pairs) parameters in the proposed methods with these of real root method. Second, we measured how long computation time is reduced. The experimental results of both methods that the searching time was reduced by about 47% in average without the change of LSP parameters.

• The Journal of the Acoustical Society of Korea
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• v.17 no.7
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• pp.101-107
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• 1998

본 논문에서는 잔차신호를 모델링하기 위해 사용되는 동적희박대수코드북에 대해 분석하고 성능이 향상된 새로운 대수코드북 구조 및 검색과정을 제안하였다. 제안된 알고리 즘은 대수 코드북의 단점을 계산량의 증가 없이 개선시켰다. 먼저 기존에 단순히 부호비트 만을 검색하는 것에 대해 다양한 펄스 진폭의 선택을 가능하게 하였다. 그리고 동일 트랙상 에서 두 펄스를 선택하게 하였으며 추가 계산량이 필요없는 무성음에서 유성음으로의 천이 구간 검출기를 이용하여 LSF 보간 시 발생하는 천이구간에서의 LP지연을 최소화하였다. 제 안된 알고리즘을 이용한 5.6kbps음성부호화기는 전화선상의 음질을 시료로 하여 주관적 음 질면에서 6.3kbps MP-MLQ와 동등하였으며 MNRU Q=15dB에서는 MP-MLQ에 비해 약간 의 음질열하가 발생하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.4
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• pp.292-302
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• 2003

We propose a new method to reduce the complexity of excitation codebook search. The preselected excitation pulses by the coarse search method can be updated to pulses with higher quality performance measure. The excitation pulses can arbitrarily be deleted and inserted among the searched pulses until the overall performance achieves. If we use this excitation pulse search method in AMR-WB, the complexity required for excitation codebook search can be reduced to half the original method while the output speech maintains equal speech quality to a conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.8
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• pp.1854-1863
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• 2003

In the recent digital communication systems, the performance of Turbo Code using the mr correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the huh decoder. However, performance depends on the interleaver depth that needs many delays over the reception process. Moreover, turbo code has been blown as the robust coding methods with the confidence over the fading channel. International Telecommunication Union(ITU) has recently adopted it as the standardization of the channel coding over the third generation mobile communications(IMT­2000). Therefore, in this paper, we preposed the interleaver that has the better performance than existing block interleaver, and modified turbo decoder that has the parallel concatenated structure using MAP algorithm. In the real­time voice and video service over third generation mobile communications, the performance of the proposed two methods was analyzed and compared with the existing methods by computer simulation in terms of reduced decoding delay using the variable decoding method over AWGN and fading channels for CDMA environments.