• MALSORI
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• no.57
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• pp.153-164
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• 2006

The MMSE-STSA based speech enhancement algorithm is widely used as a preprocessing for noise robust speech recognition. It weighs the gain of each spectral bin of the noisy speech using the estimate of noise and signal power spectrum. In this paper, we investigate the influence of parameters used to estimate the speech signal and noise power in MMSE-STSA upon the recognition performance of noisy speech. For experiments, we use the Aurora2 DB which contains noisy speech with subway, babble, car, and exhibition noises. The HTK-based continuous HMM system is constructed for recognition experiments. Experimental results are presented and discussed with our findings.

• Journal of Biomedical Engineering Research
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• v.4 no.1
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• pp.29-36
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• 1983

In this paper, center frequency down slift of ultrasonic echo signals which for the measurements of frequency dependent attenuation in the biological tissue are estimated. Center frequency down shift of echo-signals are estimated after signal spectrum analysis of whole echo-signals. In case of signal spectrums are simple, estimation of down shift frequency is very simple and in case of complicate spectrum, estimation of down shift frequency is depend on spectral shape. In case of unable to estimate, frequency dependence of medium is nonlinear(n) 1), in which upper shift of spectrums are presented. In case of unable to estimate, spectrum analysis are performed at local position. At consquence, we know that spectral dispersions are caused complicately by biological tissue layer.

• Journal of Positioning, Navigation, and Timing
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• v.9 no.2
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• pp.65-70
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• 2020

For a wireless communication system, the angle-of-arrival (AOA) of the signal has a variety of applications. The signal AOA is estimated utilizing various antenna array structure such as Uniform Linear Array (ULA), Uniform Rectangular Array (URA), and Uniform Circular Array (UCA). In this paper, we introduce a cascade AOA estimation algorithm based on the UCA, which is consisted of CAPON and Beamspace MUSIC. CAPON is employed to estimate approximate AOA groups including multiple AOA signals and Beamspace MUSIC is employed to estimate detailed signal AOAs in the estimated AOA groups. In addition, we provide the computer simulation results for verifying and analyzing the performance of the cascade AOA estimator based on UCA.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.285-288
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• 2001

In the speech analysis, to estimate formant center frequencies exactly is very important. If we know formant frequencies, we can expect which pronunciation is uttered. Generally, the magnitude of first formant frequency in voiced speech is 10dB more than other formant frequency. So, the shape of voice signal in time domain is affected by mainly first formant. Therefore we can get first formant frequency roughly by using ZCR(Zero Cross Rate). In this paper, we proposed the improvement method to get first formant frequency by using ZCR. We did autocorrelation before getting ZCR. This procedure makes voice signal smooth so, first formant in voice signal is emphasized. As a result of this method, we got more exact ZCR and first formant frequency. Conventional method of formant estimate is done in frequency domain but proposed method is done in time domain. So, this is very simple.

• Journal of information and communication convergence engineering
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• v.8 no.2
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• pp.150-154
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• 2010

In this paper, the adaptive interference cancellation system (ICS) in order to cancel the feedback signal in the wireless communication system is proposed. We cancel the interference with the attenuation signal corresponding to the feedback signal and estimate the feedback signal by using Normalized Least Mean Square (NLMS) algorithm. The proposed scheme showed a better performance of interference cancellation in the measurement results.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.3
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• pp.416-420
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• 2008

This paper proposes a new method to improve signal characteristics at branches frequently met in system-level routing. We also introduce the numerical formula which can estimate the time delay due to branches and the simple design guideline for system-level routing. Finally, we propose the routing method which can eliminate the signal reflection for the case of one driver and two receivers (multi-drop topology).

• Proceedings of the KIEE Conference
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• 2005.07a
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• pp.341-343
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• 2005

This paper introduces a compression algorithm for power quality disturbance signal via the discrete wavelet transform, DWT. Algorithm to estimate a time delay from the power quality disturbance signal is proposed. Pseudo-stationary signal is constructed from the estimated time delay. A difference signal or nonstationary signal is obtained by removing a pseudo-stationary signal from a disturbance signal. DWT is applied to a difference signal. The threshold is applied to reduce a number of coefficients. Simulation results show the resonable compression ratio while keep low signal distortion.

• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.148-153
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• 2010

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. A Speech signal consists of Voice and Unvoiced Band in The MBE excitation model. And the energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced. We use the IMBE vocoder for the Voice or Unvoice band of segmented speech signal. Continuously we calculate the segmented SNR using that information and the energy of each band. And we estimate the SNR of continuous speech signal.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.95-98
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• 2001

The adaptive line enhancer (ALE) is widely used for enhancing narrowband signals corrupted by broadband noise. In this paper, we propose novel ALE methods to improve the enhancing capability. The proposed methods are motivated by the fact that the output of the ALE is a fine estimate of the desired narrowband signal with the broadband noise component suppressed. The proposed methods preprocess the input signal using ALE filter to regenerate a finer input signal. Thus the proposed ALE is driven by the input signal with higher signal-to-noise ratio (SNR). The analysis and simulation results are presented to demonstrate that the proposed ALE has better performance than conventional ALE´s.

• Journal of Biomedical Engineering Research
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• v.15 no.3
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• pp.281-288
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• 1994

In this paper, a new decomposition method of the interference EMG signal using MAMDF filtering and digital signal processor. The efficient software and hardware signal processing techniques are employed. The MAMDF filter is employed in order to estimate the presence and likely location of the respective templates which may include in the observed mixture, and high-resolution waveform alignment is employed in order to provide the optimal combination set and time delays of the selected templates. The TMS320C25 digital signal processor chip is employed in order to execute the intensive calculation part of the software. The method is verified through a simulation with real templates which are obtain ed from needle EMG. As a result, the proposed method provides an overall speed improvement of 32-40 times.