• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.4
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• pp.112-118
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• 2010

This paper describes a noise suppression system based on a normalization method using a time-delay neural network and line spectrum pair having a parameter of frequency domain. First, a time-delay neural network is trained using line spectrum pair values of noisy speech signals obtained by linear prediction analysis. After trained the time-delay neural network, the proposed system enhances speech signals that are degraded by a background noise. Accordingly, the proposed time-delay neural network restores from the line spectrum pair values of noisy speech signals to the line spectrum pair values of clean speech signals. It is confirmed that this system is effective for speech signals degraded by a background noise, judging from spectral distortion measurement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.6
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• pp.388-395
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• 1986

Dispersion characteristics of shielded single, coupled and edge-offset microstrip structures are investigated by using hybrid mode analysis with Galerkin's method in the spectral domain. Two new basis functions for the longitudinal strip current are proposed and convergence rates of the solutions for the basis functions are compared. Current distribution of the coupled line is obtaind from that of the single line by using shift theorem of the Fourier transform. In addition, effects of off-centered inner strip conductor on dispersion are also discussed Numerical results include various structual parameters and are compared with other available data and good agreements are observed.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.4
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• pp.498-505
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• 1999

The ferrite device that has nonreciprocal characteristics has been used as various microwave components. In this paper, the spectral-domain method was used to analyze the characteristics of structure(strip line on multilayer structure with ferrite material) for which no paper has been published yet. It is assumed that an external dc magnetic field is applied perpendicular to the ground conductor. The propagation constant is calculated for each parameter and frequency. Also we considered convergence by increasing the number of basis functions and verified the numerical analysis results. From the results, we could confirm that as the thicknesses of YIG is increased, the cutoff region is widened and the propagation constant is decreased at the frequency of upper cutoff region. For a larger applied dc magnetic field, the frequency of cutoff region is found to be increased. Also the simulation results for strip line show that the cutoff region is widened and the propagation constant is larger than that of microstrip line.

• 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.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.7
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• pp.28-38
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• 1997

Previously, we have proposed the M-BECM(Modified-Band Edge Component Maximization), which is a symbol synchronization algorithm based on spectral line method for all-digital high speed digital communications. However, Until now, the characteristics of the timing detector based on the spectral line method including M-BECM was not analyzed, particularly the effect of a timing offset at the optimal convergence pont. In this paper, we analyze the timing dtector's characteristics of the M-BECM and present optimal design value. First, the expression for the timing detector's mean value(often called its S-Curver) as a function of the normalized symbol timing offset is derived. Next, the P $D_{bias}$, the value for compensating the timing offset at an optimal convergence point, and the bandwidth of bandpass filter in the timing detector are calculated. It is also shown and analyzed that the P $D_{bias}$ is affected by varuous factors such as the excess bandwidth of input signal, frequency offsets, noise and particularly, the excess bandwidth of input signal is a major parameter to decide P $D_{bias}$. Finally, analytic resutls are compared to simulation results.

• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.1
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• pp.37-46
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• 2003

The transient responses on arbitrarily interconnected digital transmission lines are analyzed by an efficient node discretization technique. Since the proposed node discretization technique offers an efficient means to discretize transmission lines, the transient waveform at any position on the arbitrarily interconnected lines is easily predicted. Dispersive microstrip multiconductor transmission lines arbitrarily connected are analized for generality. The derivation of frequency-dependent equivalent circuit elements of coupled transmission lines have been carried out by the spectral domain approach(SDA). The effects of variations of excited pulse width on the crosstalks of the high-speed microstrip coupled-lines are also investigated. It has been well known that the crosstalk spike level is monotonously increased when the coupling length and effective permittivity of substrate are increased. In this paper, it is found that the variations of crosstalk level are not further monotonous as shortening the exciting pulse width toward several picosecond. The results are verified by the generalized S-parameter technique.