• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.33-37
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• 1997

This paper describes experimental results from emotional speech materials, which is analysed by various signal processing methods. Speech materials with emotional informations are collected from actors. Analysis is focused to the variations of pitch informations and durations. From the analysed results we can observe the characteristics of emotional speech. The materials from this experiment provides valuable resources for analysing emotional speech.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.47-50
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• 2000

비상관적인 가산잡음에 오염된 음성으로부터 향상된 음성을 얻기 위한 방법 중 Soft Decision에 근거한 음성 향상 기법이 뛰어난 성능을 가진다고 알려져 있다. Soft Decision은 주파수 영역에서 음성에 가산된 잡음을 처리하며, 잡음 환경에 대한 사전정보에 의존적이다. 본 연구에서는 Soft Decision을 근거로 음성에 가산된 잡음신호를 비선형 처리를 하여 효과적으로 음성에 포함된 잡음을 추정하도록 하였으며, 잡음환경에 대한 사전 정보 없이 효율적으로 잡음을 억제하는 방법을 제안한다. 본 연구에서 제안한 음성향상 기법은 주관적인 음질평가에서 기존의 방법들보다 나은 성능을 나타내었다

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.11
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• pp.2512-2520
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• 1997

The classes of Reverse Jacket matrix [RJ]$_{N}$ and the corresponding Restclass Reverse Jacket matrix ([RRJ]$_{N}$) are defined;the main property of [RJ]$_{N}$ is that the inverse matrices of them can be obtained very easily and have a special structure. [RJ]$_{N}$ is derived from the weighted hadamard Transform corresponding to hadamard matrix [H]$_{N}$ and a basic symmertric matrix D. the classes of [RJ]$_{2}$ can be used as a generalize Quincunx subsampling matrix and serveral polygonal subsampling matrices. In this paper, we will present in particular the systematical block-wise extending-method for {RJ]$_{N}$. We have deduced a new orthorgonal matrix $M_{1}$.mem.[RRJ]$_{N}$ from a nonorthogonal matrix $M_{O}$.mem.[RJ]$_{N}$. These matrices can be used to develop efficient algorithms in IMT2000 signal processing, multidimensional subsampling, spectrum analyzers, and signal screamblers, as well as in speech and image signal processing.gnal processing.g.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.743-749
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• 2007

Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1983.04a
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• pp.16-17
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• 1983

In Speech signal processing, ARMA spectral estimation method is used. It has been demonstrated that the ARMA model provides better spectral estimation then the more specialized AR model and MA model. Dynamic program is used to achieve time algnment. Speech sound similarity is defined to be proportional to the distance seperating to sound in a vector space defined by ARMA model. AS a result, the recognition rate of 97.3% for three speaker is obtained.

• Proceedings of the Acoustical Society of Korea Conference
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• 1996.10a
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• pp.70-75
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• 1996

In this paper, the pre-precessing method for the recognition of single vowels by use of spectrum envelope is presented , we use new method of an extrating spectrum envelope using the diffusion filter bank. We reduced the total processing time, and got higher enhancement of discrimination . By getting 88.3% of average recognition rate for single vowels of real voice through computer simulation, we confirmed it to be useful for speech recongition which use spectrum analysis for voice signal to have many frequency components.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.39-42
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• 1988

In speech signal processing, it is necessary to estimate exactly the pitch. We propose a new algorithm which uses the correlation coefficient between the original and the repeated segmental waveform in the frame as a parameter in the pitch extraction. The correlation coefficient in the frame reflects the periodic component and the transient ratio of the waveform.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.179-182
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• 2002

In this paper, we propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the proposed algorithm is suitable for adaptive signal processing area using speech or acoustic field.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.1
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• pp.23-28
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• 2006

Pitch period that is a important factor in speech signal processing is used in various applications such as speech recognition, speaker identification, speech analysis and synthesis. So many pitch detection algorithms have been studied until now. AMDF which is one of pitch period detection algorithms chooses the time interval from valley point to valley point as pitch period. In selection of valley point to detect pitch period, complexity of the algorithm is increased. So in this paper we proposed the simple algorithm using rotation transform of AMDF that detects global minimum valley point as pitch period of speech signal and compared it with existing methods through simulation.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.310-314
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• 2002

The exact pitch (fundamental frequency) extraction is important in speech signal processing like speech recognition, speech analysis and synthesis. However the exact pitch extraction from speech signal is very difficult due to the effect of formant and transitional amplitude. So in this paper, the pitch is detected after the elimination of formant ingredients by flattening the spectrum in frequency region. The effect of the transition and change of phoneme is low in frequency region. In this paper we proposed the new flattening method of log spectrum and the performance was compared with LPC method and Cepstrum method. The results show the proposed method is better than conventional method.