• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.9
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• pp.1293-1298
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• 2013

This paper proposes a noise reduction algorithm using Wiener filter to remove the noise components from the noisy speech in order to improve the speech signal. The proposed algorithm first removes the noise spectrums of white noise from the noisy signal based on the noise reshaping and reduction method at each frame. And this algorithm enhances the speech signal using Wiener filter based on linear predictive coding analysis. In this experiment, experimental results of the proposed algorithm demonstrate using the speech and noise data by Japanese male speaker. Based on measuring the spectral distortion (SD) measure, experiments confirm that the proposed algorithm is effective for the speech by contaminated white noise. From the experiments, the maximum improvement in the output SD values was 4.94 dB better for white noise compared with former Wiener filter.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.81-84
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• 2000

In this paper, a new technology for extracting the feature of the speech signal of an isolated word by the analysis on the frequency domain is proposed. This technology can be applied efficiently for the limited speech domain. In order to extract the feature of speech signal, the number of peaks is calculated and the value of the frequency for a peak is used. Then the difference between the maximum peak and the second peak is also considered to identify the meanings among the words in the limited domain. By implementing this process hierarchically, the feature of speech signal can be extracted more quickly.

• Speech Sciences
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• v.3
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• pp.132-147
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• 1998

Previously developed methods for pitch modification have not been based on the voice source model. Therefore, the synthesized speech often sounds unnatural although it may be highly intelligible. The purpose of this paper is to analyze the alteration of a voice source signal with pitch period and to establish the pitch-modification rule based on the result of this analysis. We examine the alteration of the interval of closing phase, closed phase and open phase using the excitation waveform as the pitch increases. In comparison to the previous methods which performed directly on the speech signal, the pitch modification method based on a voice source model shows high intelligibility and naturalness. This study might benefit the application to the speaker identification and the voice color conversion. Therefore the proposed method will provide high quality synthetic speech.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.91-94
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• 1994

This paper is a study on the design and implementation of the Korean Tet-to-Speech system. In this paper, parameter symthesis method is chosen for speech symthesis method and PARCOR coeffient, one of the LPC analysis, is used as acoustic parameter, We use a diphone as synthesis unit, it include a basic naturalness of human speech. Diphone DB is consisted of 1228 PCM files. LPC synthesis method has defect that decline clearness of synthesis speech, during synthesizing unvoiced sound In this paper, we improve clearness of synthesized speech, using residual signal as ecitation signal of unvoiced sound. Besides, to improve a naturalness, we control the prosody of synthesized speech through controlling the energy and pitch pattern. Synthesis system is implemented at PC/486 and use a 70Hz-4.5KHz band pass filter for speech imput/output, amplifier and TMS320c30 DSP board.

• Journal of Digital Convergence
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• v.11 no.7
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• pp.209-214
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• 2013

Improve the recognition performance of speech recognition systems as a method for recognizing human listening skills were incorporated into the system. In noisy environments by separating the speech signal and noise, select the desired speech signal. but In terms of practical performance of speech recognition systems are factors. According to recognized environmental changes due to noise speech detection is not accurate and learning model does not match. In this paper, to improve the speech recognition feature extraction using gamma tone and learning model using acoustic model was proposed. The proposed method the feature extraction using auditory scene analysis for human auditory perception was reflected In the process of learning models for recognition. For performance evaluation in noisy environments, -10dB, -5dB noise in the signal was performed to remove 3.12dB, 2.04dB SNR improvement in performance was confirmed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.6
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• pp.994-1003
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• 1994

In this paper, a scheme of noise cancellation using spectral subreaction method with single input in an autombile noise environment is proposed. In order to remove the changing automonile noise components form the noisy speech signal, the noise of various states is analyzed and its characteristics are presented. For the decision of speech/nonspeech and the estimation of noise spectrum, a classification function is proposed on the basis of noise analysis. This function presents the precise decision of speech/nonspeech and the optimal estimation of noise spectrum with less computation. As the result of the estimation of noise spectrum by the proposed classification function, the clean speech signal is extracted from the noisy speech signal with high signal-to-ratio.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1799-1802
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• 2002

This paper proposes an improved method for formant frequencies estimation based on the complex autocorrelation function of the speech signal. Instead of using the incoming signal as an input fur the LPC analysis, the analytic signal of the autocorrelation function of the speech signal is computed and itself used as an input for the LPC analysis. Due to the properties of the analytic signal, which occupies half of the bandwidth of the original signal, the required model order for the LPC analysis is halved. The accuracy of the proposed method in noisy environments is examined on five natural vowels. The effectiveness of the proposed method is shown by the estimated spectral shapes and the estimation errors of the formant frequencies.

• Journal of the Korean Institute of Telematics and Electronics
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• v.12 no.3
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• pp.30-36
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• 1975

Distortions are analyzed and compared for two TDM channel expansion methods- periodic sample skipping and sampling frequency reduction. Signal is assumed to be stationary random signal with zero.mean. Channel noise and interference are not considered in the analysis. For speech signal, it is shown that the periodic sample skipping method could be a better choice under practical design constraints.

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.5
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• pp.811-816
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• 2019

This paper first proposes a technique to separate clean speech signals and mixed noise signals by using an independent vector analysis algorithm of frequency band for 4 channel speech source signals with a noise. An improved output speech signal from the proposed independent vector analysis algorithm is obtained by using the cross-correlation between the signal outputs from the frequency domain delay-sum beamforming and the output signals separated from the proposed independent vector analysis algorithm. In the experiments, the proposed algorithm improves the maximum SNRs of 10.90dB and the segmental SNRs of 10.02dB compared with the frequency domain delay-sum beamforming algorithm for the input mixed noise speeches with 0dB and -5dB SNRs including white noise, respectively. Therefore, it can be seen from this experiment and consideration that the speech quality of this proposed algorithm is improved compared to the frequency domain delay-sum beamforming algorithm.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.28-33
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• 2000

For efficient quantization of speech representation, it is common to incorporate Perceptual characteristics of human hearing. However, the focus has been confined only to the magnitude information of speech, and little attention has been paid to phase information. This paper presents a novel approach, termed perceptually irrelevant phase elimination (PIPE), to find out irrelevant phase information of acoustic signals in terms of perception. The proposed method, which is based on the observation that the relative phase relationship within a critical band is perceptually important, is derived not only for stationary Fourier signal but also for harmonic signal. The proposed method is incorporated into the analysis/synthesis system based on harmonic representation of speech, and subjective test results demonstrate the effectiveness of proposed method.