• Title/Summary/Keyword: 비트 스펙트럼

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A Temporal Decomposition Method Based on a Rate-distortion Criterion (비트율-왜곡 기반 음성 신호 시간축 분할)

  • 이기승
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.3
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    • pp.315-322
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    • 2002
  • In this paper, a new temporal decomposition method is proposed. which takes into consideration not only spectral distortion but also bit rates. The interpolation functions, which are one of necessary parameters for temporal decomposition, are obtained from the training speech corpus. Since the interval between the two targets uniquely defines the interpolation function, the interpolation can be represented without additional information. The locations of the targets are determined by minimizing the bit rates while the maximum spectral distortion maintains below a given threshold. The proposed method has been applied to compressing the LSP coefficients which are widely used as a spectral parameter. The results of the simulation show that an average spectral distortion of about 1.4 dB can be achieved at an average bit rate of about 8 bits/Frame.

A Study on Estimation of Beat Frequency in a Water Level Measurement Radar (수위 측정 레이다에서의 비트 주파수 추정에 관한 연구)

  • Lee, Jonggil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.8
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    • pp.1791-1797
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    • 2014
  • A water level measurement radar estimates beat frequencies to extract the corresponding water level information for water resource management. Therefore, it is required to maintain the high degree of reliability and accuracy in beat frequency estimates. However, Beat spectra of water surface return echoes can have very widely varying shapes according to system frequency linearity, measurement environments and weather conditions. Therefore, serious problems may arise in maintaining the reliability and accuracy of the conventional level estimation method. Therefore, in this paper, high resolution spectrum estimates are analyzed for improvement of the conventional method. These methods show that the more accurate level measurement can be possible on the condition that SNR is higher than a certain required threshold. However, if SNR is too low, the conventional method shows that estimates are more reliable than the suggested method though absolute errors are too large.

Perceptual and Adaptive Quantization of Line Spectral Frequency Parameters (선 스펙트럼 주파수의 청각 적응 부호화)

  • 한우진;김은경;오영환
    • 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.

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A Study on Estimation of a Beat Spectrum in a FMCW Radar (FMCW 레이다에서의 비트 스펙트럼 추정에 관한 연구)

  • Lee, Jong-Gil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.12
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    • pp.2511-2517
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    • 2009
  • Recently, a FMCW radar is used for the various purposes in the short range detection and tracking of targets. The main advantages of a FMCWradar are the comparative simplicity of implementation and the low peak power transmission characterizing the very low probability of signal interception. Since it uses the frequency modulated continuous wave for transmission and demodulation, the received beat frequency represents the range and Doppler information of targets. Detection and extraction of useful information from targets are performed in this beat frequency domain. Therefore, the resolution and accuracy in the estimation of a beat spectrum are very important. However, using the conventional FFT estimation method, the high resolution spectrum estimation with a low sidelobe level is not possible if the acquisition time is very short in receiving target echoes. This kind of problems deteriorates the detection performance of adjacent targets having the large magnitude differences in return echoes and also degrades the reliability of the extracted information. Therefore, in this paper, the model parameter estimation methods such as autoregressive and eigenvector spectrum estimation are applied to mitigate these problems. Also, simulation results are compared and analyzed for further improvement.

A Study on Analysis of Beat Spectra in a Radar System (레이다 시스템에서의 비트 스펙트럼 분석에 관한 연구)

  • Lee, Jong-Gil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.14 no.10
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    • pp.2187-2193
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    • 2010
  • A specific radar system can be implemented more easily using the frequency modulated continuous wave comparing with the pulse Doppler radar. It also has the advantage of LPI (low probability of interception) because of the low power and wide bandwidth characteristics. These radars are usually used to cover the short range area and to obtain the high resolution measurements of the target range and velocity information. The transmitted waveform is used in the mixer to demodulate the received echo signal and the resulting beat signal can be obtained. This beat signal is analyzed using the FFT method for the purpose of clutter removal, detection of a target, extraction of velocity and range information, etc. However, for the case of short signal acquisition time, this FFT method can cause the serious leakage effect which disables the detection of weaker echo signals masked by strong side lobes of the clutter. Therefore, in this paper, the weighting window method is analyzed to suppress the strong side lobes while maintaining the proper main lobe width. Also, the results of FFT beat spectrum analysis are shown under various environments.

Performance Analysis of a Trellis coded 4-CPFSK over an Statistical Underwater Acoustic Channel (통계적 수중음향 채널에서 트렐리스 부호화된 4-CPFSK의 성능분석)

  • Kang, Hee-hoon
    • Journal of the Institute of Electronics and Information Engineers
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    • v.54 no.6
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    • pp.140-145
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    • 2017
  • A channel is very sever in an underwater acoustic communication. Therefore, a modulation method with high spectrum efficiency is needed in an underwater acoustic communication. PSK(phase shift keying) for transmitting 1 bit or 2 bits is robust to noise, but sensitive to noise about more than 3 bits. CPFSK error performance for transmitting 1 bit or 2 bits is similar to that of PSK and CPFSK decreases high frequency components in modulation signal. In the paper, I analyze the performance of trellis coded 4-CPFSK modulation.

Analytic derivation of the finite wordlength errors in fixed-point implementation of SDFT (SDFT 고정소수점 연산에 대한 유한 비트 오차영향 해석)

  • Chang, Tae-Gyu;Kim, Jae-Hwa
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.37 no.4
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    • pp.65-71
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    • 2000
  • Finite wordlength effect of the recursive implementation of SDFT(sliding-DFT) is analytically derived in this paper. Representation errors of the twiddle coefficients and the data registers are the two major causes of the spectral errors in the recursive implementation. The noise-to-signal ratio is analytically derived in terms of the coefficients wordlength, the data registers wordlength, and the DFT's block-length used in the computation Error dynamic equation is obtained from the recursive DFT and the probabilistic models for the coefficients error and the round-off error are introduced for the NSR derivation, The result of the NSR derivation is verified with the simulation data.

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비트겐슈타인 철학의 현상학적 측면 - 박정일 박사와 김영건 박사의 비판에 대한 반론 -

  • Park, Byeong-Cheol
    • Korean Journal of Logic
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    • v.7 no.2
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    • pp.47-70
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    • 2004
  • 필자는 이 글에서 박병철의 저서 Phenomenological Aspects of Wittgenstein's Philosophy (Kluwer 1998)에 대한 박정일 박사의 서평과 김영건 박사의 비판에 대한 반론을 제시하고 있다. 먼저 현상학과 현상론의 구분이 불분명하다는 비판에 대한 반론에서 시작하여, 유아론에 대한 비판을 반박하고 있으며, "논리-철학 논고"에서의 대상의 본성에 대한 견해도 밝히고 있다. 필자는 무엇보다도 자신의 시도가 여전히 불투명한 비트겐슈타인의 철학을 놓고 벌어지는 다양한 해석의 스펙트럼의 한 부분을 이룰 수 있을 것이라고 주장한다.

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A Study on Analysis of Phase Noise Effects in a FM-CW Radar System (FM-CW 레이다 시스템에서의 위상잡음 영향 분석에 관한 연구)

  • Lee, Jong-Gil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.15 no.9
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    • pp.1840-1846
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    • 2011
  • It is necessary to estimate the Doppler spectrum for each range cell for the extraction of useful information from the return echoes in radar systems used for the remote sensing purpose such as detection of moving targets and weather surveillance. The signal amplitude in the beat frequency band is the important parameter in the detection and tracking of targets. However, strong clutter echoes do exist in most radar operation environments and the system phase noise spreads both the clutter and signal echoes of the target. In this paper, the effects of this system phase noise are analyzed concerning the clutter and the signal beat spectrum. It is shown that the separation capability of adjacent beat signal depends on the degree of spread in the clutter and beat signal caused by the radar system phase noise

Intelligibility Improvement of Low Bit-Rate Speech Coder Using Stochastic Spectral Equalizer (통계적 스펙트럼 이퀄라이저를 이용한 저 비트율 음성부호화기의 명료도 향상)

  • Lee, Jeong Hun;Yun, Deokgyu;Choi, Seung Ho
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.41 no.10
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    • pp.1183-1185
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    • 2016
  • Low bit-rate speech coder in digital speech communications synthesizes speech using vocal tract model parameters. In this case, the spectra of the synthesized speech can be much distorted since the allocated bits for the parameters are considerably limited, which results in the degradation of speech intelligibility. In this paper, we propose a speech intelligibility improvement method using stochastic spectral equalizer. This method stochastically obtains the weight vector of each speech coder using spectral ratios between original and synthesized speech, then applies this weight vector to synthesized speech. From the experiments of objective speech intelligibility tests, we found that the performance of the proposed method is better than that of the conventional method.