• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.4
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• pp.430-436
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• 1988

LSF(Line Spectral Frequency) derived from LPC has known as a very useful transmission parameter of speech signals, for it has a good linear interpolation characteristics and a low spectrum distortion at low bit rates coding. This paper presents that it is possible to extract directly the formant frequencies of speech signals from LSF parameter without application of FFT algorithm by comparing the distribution of LSF parameter with the frequency distribution of analysis filter. This paper suggests the advanced algorithm that results in improving the speed of convergence at analytic solution method. Also, for the flexibility of parameters, the process that transforms from LSF to LPC is presented.

• The Journal of the Acoustical Society of Korea
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• v.31 no.2
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• pp.93-99
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• 2012

For speech coding, a vocal tract is modeled using Linear Predictive Coding (LPC) coefficients. The LPC coefficients are typically transformed to Line Spectral Frequency (LSF) parameters which are advantageous for linear interpolation and quantization. If multidimensional LSF data are quantized directly using Vector-Quantization (VQ), high rate-distortion performance can be obtained by fully utilizing intra-frame correlation. In practice, since this direct VQ system cannot be used due to high computational complexity and memory requirement, Split VQ (SVQ) is used where a multidimensional vector is split into multilple sub-vectors for quantization. The LSF parameters also have high inter-frame correlation, and thus Predictive SVQ (PSVQ) is utilized. PSVQ provides better rate-distortion performance than SVQ. In this paper, to implement the optimal predictors in PSVQ for voice storage devices, we propose Multi-Frame AR-model based SVQ (MF-AR-SVQ) that considers the inter-frame correlations with multiple previous frames. Compared with conventional PSVQ, the proposed MF-AR-SVQ provides 1 bit gain in terms of spectral distortion without significant increase in complexity and memory requirement.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.29-34
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• 2001

In this paper, we designed an LSF coefficient quantizer of the wideband speech codec that can produce high quality speech service. For the efficient LSF coefficient quantizer, the interframe correlation was used. Also we separately quantized the LSF coefficients with high and low interframe correlation. Predictive pyramid vector quantizer (PVQ) was used for quantizing the LSF coefficients with high interframe correlation, and PVQ was used for quantizing the LSF coefficients with low interframe correlation. Experiments show that the proposed UF quantizer can quantize LSF information in 40 bits/frame, with an average spectral distortion (SD) of 1 dB and less than 3.87% frames having SD greater than 2 dB.

• Phonetics and Speech Sciences
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• v.6 no.4
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• pp.133-139
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• 2014

This paper describes several approaches to transform a speaker's individuality to another's individuality using frequency warping between bilingual formant frequencies on different language environments. The proposed methods are simple and intuitive voice conversion algorithms that do not use training data between different languages. The approaches find the warping function from source speaker's frequency to target speaker's frequency on formant space. The formant space comprises four representative monophthongs for each language. The warping functions can be represented by piecewise linear equations, inverse matrix. The used features are pure frequency components including magnitudes, phases, and line spectral frequencies (LSF). The experiments show that the LSF-based voice conversion methods give better performance than other methods.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.68-77
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• 2000

Line special frequency (LSF) parameters have been widely used in low bit-rate speech coding due to their efficiency for representing the short-time speech spectrum. In this paper, a new distance measure based on the masking properties of human ear is proposed for quantizing LSF parameters whereas most conventional quantization methods are based on the weighted Euclidean distance measure. The proposed method derives the perceptual distance measure from the definition of noise-to-mask ratio (NMR) which has high correspondence with the actual distortion received in the human ear and uses it for quantizing LSF parameters. In addition, we propose an adaptive bit allocation scheme, which allocates minimal bits to LSF parameters maintaining the perceptual transparency of given speech frame for reducing the average bit-rates. For the performance evaluation, we has shown the ratio of perceptually transparent frames and the corresponding average bit-rates for the conventional and proposed methods. By jointly combining the proposed distance measure and adaptive bit allocation scheme, the proposed system requires only 770 bps for obtaining 95.5% perceptually transparent frames, while the conventional systems produce 89.9% at even 1800 bps.

• ETRI Journal
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• v.30 no.5
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• pp.738-740
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• 2008

In this paper, block constrained trellis coded vector quantization (BC-TCVQ) is presented for quantizing the line spectrum frequency parameters of the wideband speech codec. Both a predictive structure and a safety-net concept are combined into BC-TCVQ to develop the predictive BC-TCVQ. The performance of this quantization is compared with that of the linear predictive coding vector quantizer used in the AMRWB codec, demonstrating reductions in spectral distortion.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.153-156
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• 2000

본 논문에서는 멀티미디어 서비스들 중에서 음성 또는 오디오 신호를 저속으로 압축할 때 사용되는 G.723.1 부호화기의 line spectral frequency(LSF) 계수 양자화 방식을 고속으로 처리하는 알고리즘을 제안하였다. 제안된 고속탐색 방법은 LSF 계수의 순서성질을 이용하여 코드북의 탐색 범위를 줄임으로써 계산량을 크게 감소시킨다. 제안된 고속탐색 방법을 predictive split VQ(PSVQ) 구조를 갖는 G.723.1 에 적용한 결과 spectral distortion(SD) 성능 감쇄 및 추가적인 메모리 증가 없이 최적 코드벡터를 찾기 위한 코드북 탐색 과정에서 코드북의 평균 탐색 범위가 $20.1\%$ 감소했으며, 이는 additions, subtractions, multiplies 및 comparisons 수가 각각 $19.1\%$, $20.1\%$, $19.4\%$ 및 $12.2\% 감소하는 결과를 얻었다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.787-790
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• 2000

This paper introduces a new type of representation of the LSPs as a promising alternative used for transmitting the LPC parameters. Major contribution in this paper is that the vocal track information embedded on the spectral envelope can be represented in terms of the reduced number of LSF compared tn the conventional. Hence, it provides a possibility that LPC parameters could be quantized at a reduced bit rate without causing any major spectral distortion. The simulation result illustrates the capability of the proposed LSPs representation as an efficient quantization method via a proper rejection of the redundant pairs of pole and zero along the unit circle.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.115-118
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• 2000

Voice conversion technology is essential for TTS systems because the construction of speech database takes much effort. In this paper. male-to-female voice conversion technology in Korean LPC TTS system has been studied. In general. the parameters for voice color conversion are categorized into acoustic and prosodic parameters. This paper adopts LSF(Line Spectral Frequency) for acoustic parameter, pitch period and duration for prosodic parameters. In this paper. Pitch period is shortened by the half, duration is shortened by $25\%, and LSFs are shifted linearly for the voice conversion. And the synthesized speech is post-filtered by a bandpass filter. The proposed algorithm is simpler than other algorithms. for example, VQ and Neural Net based methods. And we don't even need to estimate formant information. The MOS(Mean Opinion Socre) test for naturalness shows 2.25 and for female closeness, 3.2. In conclusion, by using the proposed algorithm. male-to-female voice conversion system can be simply implemented with relatively successful results.