• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.1107-1110
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• 1999

In speech signal processing, the accurate decision of the voiced/unvoiced sound is important for robust word recognition and analysis and a high coding efficiency. In this paper, we propose the mehod of the voiced/unvoiced decision using the LSP parameter which represents the spectrum characteristics of the speech signal. The voiced sound has many more LSP parameters in low frequency region. To the contrary, the unvoiced sound has many more LSP parameters in high frequency region. That is, the LSP parameter distribution of the voiced sound is different to that of the unvoiced sound. Also, the voiced sound has the minimun value of sequantial intervals of the LSP parameters in low frequency region. The unvoiced sound has it in high frequency region. we decide the voiced/unvoiced sound by using this charateristics. We used the proposed method to some continuous speech and then achieved good performance.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.4
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• pp.286-291
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• 2003

In this Paper, we designed an LSP(Line Spectral Pairs) parameter quantizer with cascaded structure of vector quantizer and scalar quantizer for the wideband speech coder. We have chosen the 16th-order of the LP coefficients. These coefficients are then transformed into the LSP parameters which have the excellent properties for quantization and easy stability checking condition of synthesis filter. In the first stage of quantization, input LSP parameters are split-vector-quantized using two 8-th order codebooks. In the second stage, the components of residual vector are individually quantized by the scalar quantizer utilizing the ordering property of LSP parameters. The designed adaptive VQ-SQ quantizer using 35 bits/frame shows the wideband transparency that the average spectral distortion should be less than 1.6 ㏈ and less than 4% of the frames should have SD above 3 ㏈. The simulation results show that the designed quantizer provides a 2-3 bits/frame saving over the typical vector-scalar quantizer.

• ETRI Journal
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• v.23 no.1
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• pp.16-22
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• 2001

Efficient quantization methods of the line spectrum pairs (LSP) which have good performances, low complexity and memory are proposed. The adaptive quantization range method utilizing the ordering property of LSP parameters is used in a scalar quantizer and a vector-scalar hybrid quantizer. As the maximum quantization range of each LSP parameter is varied adaptively on the quantized value of the previous order's LSP parameter, efficient quantization methods can be obtained. The proposed scalar quantization algorithm needs 31 bits/frame, which is 3 bits less per frame than in the conventional scalar quantization method with interframe prediction to maintain the transparent quality of speech. The improved vector-scalar quantizer achieves an average spectral distortion of 1 dB using 26 bits/frame. The performances of proposed quantization methods are also evaluated in the transmission errors.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.249-252
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• 2010

LPC 계수를 LSP 변환하는 방법에는 복소근, 실근, 비율 필터, 체비셰프 급수, 적응적 순차형 최소제곱 평균 방법(adaptive sequential LMS) 등이 있다. 이 방법들 중 음성 부호화기에서 주로 사용하는 실근 방법은 근을 구하기 위해 주파수 영역을 순차적으로 검색하기 때문에 계산시간이 많이 소요되는 단점을 갖는다. 본 논문에서는 LPC에서 LSP로 변환하는 4가지 고속 알고리즘을 제안한다. 첫 번째 방식에서는 검색간격에 멜 스케일을 적용하였고, 두 번째는 홀수번째 LSP 파라미터의 분포도를 이용하여 검색순서를 조정한 방법이다. 세 번째 방식과 네 번째 방식에서는 각각, 모음 특성, LSP 분포특성과 해상도를 이용하여 계산시간을 단축하였다. LSP 변환시간은 4가지 방법 모두 35~50% 단축되었다. 또한 실험결과에서는 각 알고리즘의 고유한 특성에 대하여 분석한다.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.33 no.8
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• pp.767-772
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• 2009

Laser shock peening(LSP) is proving to be better surface treatment than conventional one such as shot peening. The LSP process has a compressive residual stress into a metal alloy and a significant improvement in fatigue life. Our research is focused on applying finite element method to the prediction of residual stress through the LSP processing in some LSP conditions such as pressure and spot size induced by laser. Two analysis methods are considered to calculating the compressive residual stress. But the explicit solution and the static one after partially explicit solving are almost same. In LSP, because of very high strain rate($10^6s^{-1}$), HEL(Hugoniot Elastic Limit) is the most important parameter in material behavior modeling. As the circular laser spot is considered, 2-D axisymmetric elements are used and the infinite elements are applied to boundaries for no reflection. The relations of material properties and the LSP are also important parts in this study.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.801-804
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• 2001

LSP 파라미터는 음성코덱(codec)이나 인식기에서 음성 신호를 분석하여 전송형이나 저장형 파라미터로 변환되어, 주로 저전송률 음성부호화기에 사용된다. 그러나 LPC 계수를 LSP로 변환하는 방법이 복잡하여 계산시간이 많이 소요된다는 단점이 있다[1]. 기존의 LSP 변환 방법 중 음성 부호화기에서 주로 사용하는 real root 방법은 근을 구하기 위해 주파수 영역을 순차적으로 검색하기 때문에 계산시간이 많이 소요되는 단점을 갖는다. 본 논문에서 비교 평가한 알고리즘은 첫 번째, 기존의 real root 알고리즘, 두 번째는, LSP 파라미터의 분포 특성을 조사하여 이를 토대로 검객구간의 순서와 검색간격을 달리한 경우, 세 번째는 검색 시 mel scale을 사용한 알고리즘이다. 실험결과, 기존의 real root 방식에 비하여 두 가지 방식 모두가 변환시간의 40% 이상이 감소되는데 반하여 통일한 관을 찾음을 알 수가 있었고, 특히 분포특성을 이용하여 검색순서와 간격조절을 한 경우에 있어서, 기존의 방식보다 40%이상이 감소되었다.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.405-408
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• 2002

LSP(Line Spectrum Pairs) Parameter is used for speech analysis in vocoders or recognizers since it has advantages of constant spectrum sensitivity. low spectrum distortion and easy linear interpolation. However the method of transforming LPC(Linear Predictive Coding) into LSP is so complex that it takes much time to compute. Among conventional methods, the real root method is considerably simpler than others, but nevertheless, it still suffers from its jndeterministic computation time because the root searching is processed sequentially in frequency region. We suggest a method of reducing the LSP transformation time using voice characteristics The proposed method is to apply search order and interval differently according to the distribution of LSP parameters. in comparison with the conventional real root method, the proposed method results in about 46.5% reduction. And, the total computation time is reduce to about 5% in the G.723.1 vocoder.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.71-83
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• 2003

In this paper, a novel transcoding algorithm for the G.729A and the Selectable Mode Vocoder(SMV) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. In transcoder from SMV to G.729A, LSP conversion algorithm, pitch delay conversion algorithm and transcoding algorithm in lower rate are proposed, and in transcoder from G.729A to SMV, LSP conversion algorithm, pitch delay conversion algorithm and rate selection algorithm are proposed. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent or Improved speech quality to that produced by the tandem transcoding algorithm.

• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.87-92
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• 2007

The conventional CELP(code excited linear prediction) type vocoder has no V/UV(voiced/unvoiced) classifier. So, the unvoiced speech is processed like the voiced speech. In this paper, to reduce the bit rate, we propose a new V/UV decision algorithm minimized error rate and preprocessing computation. This V/UV classifier use the LSP(line spectrum pair) parameter which is acquired spectrum analysis process in CELP vocoders. Applying this method to the 5.3kbps ACELP(algebraic code excited linear prediction) in the G.723.1, we can get the transmission bits rate reduction of 6% approximately without degradation of speech quality.

• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.199-202
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• 2000

Speaking Rate has variety depends on the situation and habit of speakers. It has been many studied about speaking rate In speaker recognition. The study of speaking rate in speech recognition is one of considerable matter when It is recognized the speakers and it is measured by many speech data base and complicate estimation for accuracy. In this paper, conventional vocoder process the speech signal when encoding and transmitting without regard to speaking rate so in order to apply the speaking rate for vocoder It should be considered the simpler algorithm and less computation amount than the conventional method of speaking rate used In speech recognition. We proposed the speaking rate algorithm which is used the simple parameter with Line Spectrum Pair (LSP). The proposed peaking rate method is measured by the information of LSP in speech. We measured the variety rate of phenomenon about utterances which have different velocity, respectively. As a result, It has distinct variation rate of phenomenon between utterances uttered fast and slow and the rate is 42.8% higher in case of uttered fast than in case of uttered slow.