• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.76-80
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• 1995

Differential pulse code modulation 구조의 Line spectrum pairs 양자화기에 높은 차수 LSP 파라미터 값은 낮은 차수 LSP 파라미터 값보다 커야한다는 LSP 파라미터의 순서화 특징을 이용하여 양자화하는 방법을 제시한다. 31 비트/프레임에서 1 dB 이하의 스펙트럴 왜곡을 갖는 명료한 음질을 보였다. 또한 LSP 주파수의 순서화 특성은 벡터-스칼라 결합 양자화기에 적용하였다. 벡터-스칼라 결합 양자화기에서는 26비트/프레임에서 명료한 음질을 보였다. 새로운 LSP 양자화 방법들은 전송에러 발생시 성능을 시험하고, 전송 에러에 강인하도록 프레임간 예측을 위해 Moving Average 예측기를 사용한다.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.265-272
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• 2011

In most of the ISF (or LSF) based real time speech codec, SVQ (split vector quantization) method is used to decrease the quantizer complexity and memory size of codebook. However, it produces drawback that the level of correlation between code vectors can not be used during vector splits. This paper presents a new method of adaptive ISF vector quantization, which compensates the drawbacks of SVQ structured quantizer for wideband speech codec. In each different frame, the proposed method makes use of the correlation between splitted vectors by adaptively changing codebook distribution according to ordering property of ISF. The algorithm is evaluated in AMR-WB, and shows about 1.5 bit per frame improvement.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.3C
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• pp.234-240
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• 2007

In this paper, low complexity block constrained trellis coded quantization (BC-TCQ) structures are introduced, and a predictive BC TCQ encoding method is developed for quantization of line spectrum frequencies (LSF) parameters for narrowband speech coding applications. Trellis-coded quantization(TCQ) is a form of VQ that builds the VQ codebook from interleaved constituent scalar quantization codebooks. The performance is compared to the other VQ, demonstrating reduction in spectral distortion and significant reduction in encoding complexity. The predictive BC-TCQ is about 0.47107 dB superior to the IS-641 split-VQ, 26bits/frame, in spectral distortion sense. The BC-TCQ is 64.54%, 76.93%, 2.35% of the IS-641 split-VQ, respectively, in the complexity of the additions, multiplies, comparisons.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.293-296
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• 1998

본 논문에서는 다양한 멀티미디어 서비스를 위해 가변율 half rate 음성 부호화기를 설계하였다. 유, 무성음과 묵음의 구분을 위해 본 논문에서는 프레임 에너지와 음성 파라메터들을 이용한 효과적인 voicing 결정 알고리즘을 사용하였다. 유성음을 위한 half rate 음성 부호화기는 저속에서 좋은 특성을 보이는 generalized AbS구조를 이용하였다. LPC 계수는 LSP 계수로 변환한 후 predictive 2-stage VQ를 통해서 양자화하며, 여기 신호는 음질저하를 최소화하며 복잡도를 감소시킨 shift 방식의 대수적 고정 코드북 구조를 사용하고, 적응코드북과 여기코드북의 이득은 VQ로 양자화 하였다. 무성음을 위한 부호화기는 대부분이 유성음을 위한 부호화기와 동일하지만, 무성음에서는 피치간 상관도가 매우 낮으므로 피치 보간 방법을 사용하지 않고 개루프로 피치 lag를 찾은 후 전체 프레임에 사용한다. 1 kb/s 부호화기는 묵음 구간과 주변소음 구간에 사용되며 이 구간의 신호를 피치 성분이 미약한 주변소음들로 제한하고 이에 최적인 부음성 부호화기를 설계하였다. 최종적으로 완성된 가변율 half rate 부호화기는 voice activity factor(VAF)가 0.47인 시험음성에서 약 2.6 kb/s의 평균 전송률을 보였다. 주관적 음질 평가의 일환으로 IS-96 표준 코덱인 가변율 8 kb/s QCELP와 A-B preference 시험을 실시하였다. 시험 결과 평균전송률이 약 2배인 가변율 8 kb/s QCELP 보다 우수한 음질 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.4
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• pp.50-56
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• 1997

In this paper, efficient quantization methods of line spectrum pairs (LSP) which has good performances and low complexity and memory are proosed for vocoder of mobile communication system. The adaptive quantization method utilizing the ordering property of LSP parameters is used in a scalar quantizer and a vector-scalar hybrid quantizer. The proposed scalar quantization algorithm needs 31 bits/frame to maintain the transparent quality of speech. The improved vector-scalar quantizer achieves an average spectral distortion of 1dB using 26 bits/frame. The proposed methods are evaluated in the channel errors and changed the predictor structure to maintain the robustness to channel errors.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.85-88
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• 2001

Line spectrum pair(LSP) 계수는 양자화 오류에 강하고. 선형 릴간에 효율적이며, 필터의 안정성 판정이 용이하므로 LPC를 대신하여 음성 부호화에 널리 사용되고 있다. 일반적으로 LSP 계수간에는 일정한 상관관계가 나타나고, 이 특성을 이용하면 LSP 계수의 부호량을 줄일 수 있는 가능성이 있나. 본 논문에서는 LSP 계수를 압축하기 위해 principal component analysis(PCA)를 사용한 방법을 제안한다. 제안된 방법에서는 LSP 계수를 Karhunen-Loeve(KL) 변환해 에너지가 집중되는 고유치(eigenvalue)와 고유벡터(eigenvector)를 찾고 값을 양자화 한다. 성능 평가를 위해 2.4kbps MELP(mixed excitation linear prediction)와 8kbps QCELP(qualcumn code excited linear prediction) 음성 부호화기를 사용해 결과 값을 비교했고, 압축률이 증가하는 것을 확인했다.