• The Journal of the Acoustical Society of Korea
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• 제15권3E호
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• pp.78-85
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• 1996

The set of admissible time-variations in the input signal can be separated into two categories : slow parameter changes and large parameter changes which occur infrequently. A common approach used in the tracking of slowly time-varying parameters is the exponential recursive least-squares(RLS) algorithm. There have been a variety of research works on the error analysis of the exponential RLS algorithm for the slowly time-varying parameters. In this paper, the focus has been given to the error analysis of exponential RLS algorithms for the input data with abrupt property changes. The voiced speech signal is chosen as the principal application. In order to analyze the error performance of the exponential RLS algorithm, deterministic properties of the exponential RLS algorithms is first analyzed for the case of abrupt parameter changes, the impulsive input(or error variance) synchronous to the abrupt change of parameter vectors actually enhances the convergence of the exponential RLS algorithm. The analysis has also been verified through simulations on the synthetic speech signal.

• 대한후두음성언어의학회지
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• 제19권2호
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• pp.101-106
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• 2008

The speech signal has been put into a form suitable for storage and analysis by computer, several different operation can be performed. Filtering, sampling and quantization are the basic operation in digiting a speech signal. The waveform can be displayed, measured and even edited, and spectra can be computed using methods such as the Fast Fourier Transform (FFT), Linear predictive Coding (LPC), Cepstrum and filtering. The digitized signal also can be used to generate spectrograms. The spectrograph provide major advantages to the study of speech. So, author introduces the basic techniques for the acoustic recording, digital signal processing and the principles of spectrum and spectrogram.

• 전자공학회논문지CI
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• 제41권2호
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• pp.1-8
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• 2004

자동차 텔레매틱스 용 음성 HMI(Human Machine Interface) 기술은 차량 내 음성정보기술 활용을 위하여 차량 잡음환경에 강인한 내장형 음성 기술을 통합한 음성 HMI 기반 텔레매틱스 용 DSP 시스템의 개발을 포함한다. 개발된 내장형 음성 인식엔진을 바탕으로 통합 시험을 위한 자동차 텔레매틱스 용 DSP 시스템 구현 개발을 수행하는 본 논문은 자동차용 음성 HMI의 요소 기술을 통합하는 기술 개발로 자동차용 음성 HMI 기술 개발에 중심이 되는 연구이다.

• 한국지능시스템학회논문지
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• 제15권2호
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• pp.173-178
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• 2005

본 논문은 보다 편리한 가정 생활환경 구축을 목적으로 한 가상대화시스템 구현에 관한 연구이다. 이를 실현하기 위하여 본 논문은 음성인식, 음성합성, 비디오 신호 및 센서신호처리 등의 멀티미디어 신호처리에 그 기술적 기반을 두고 있다. 대화시스템의 중요한 모듈로서의 음성합성기, HM-Net(Hidden Markov Network)에 기반한 실시간 음성인식기, 픽셀의 밝기차를 이용한 실시간 움직임 검출 및 터치센서 등을 대화시스템에 통합함으로써 이루어진다. 실제 구동 실험에서 주위 노이즈 환경의 영향으로 시뮬레이션 결과보다는 성능이 떨어지나, 소파에 앉아있는 동안 자동되는 시스템의 실험 평가에서 가전제품 능의 컨트롤이 비교적 사용하기 쉬웠다는 결과를 얻었다.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2005년도 추계학술대회 논문집
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• pp.302-305
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• 2005

Recently, many methods to analyze signal have been proposed and representative methods are the Fourier transform and wavelet transform. In these methods, the Fourier transform represents signal with combination cosine and sine at all locations in the frequency domain. However, it doesn't provide time information that particular frequency occurs in signal and denpends on only the global feature of the signal. So, to improve these points the wavelet transform which is capable of multiresolution analysis has been applied to many fields such as speech processing, image processing and computer vision. And the wavelet transform, which uses changing window according to scale parameter, presents time-frequency localization. In this paper, we proposed a new approach using a wavelet of cosine and sine type and analyzed features of signal in a limited point of frequency-time plane.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2002년도 합동 추계학술대회 논문집 정보 및 제어부문
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• pp.529-532
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• 2002

This research treated development of real time pitch tracer for training of musical tune of speech signal and pre-processing and post-processing technics were proposed to get higher accuracy in extraction of pitch. Autocorrelation Function was used to get pitch frequency from 64Hz to 980Hz in real time. Half Rectifier method and Envelop extraction method as a pre-processing was used to get higher accuracy in pitch detection, and improved results were obtained on noised speech signal. Post-processing method using periodicity of Autocorrelation was proposed to get higher accuracy in the high frequency region.

• 한국해양공학회지
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• 제25권2호
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• pp.1-6
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• 2011

Vibration-based Structural Health Monitoring (SHM) systems use field measurements of operational signals, which are distorted by noise from many sources. Reducing this noise allows a more accurate assessment of the original "clean" signal and improves analysis results. The implementation of a noise reduction methodology for the Modal Distribution Method (MDM) is reported here. The spectral subtraction method is a popular broadband noise reduction technique used in speech signal processing. Its basic principle is to subtract the magnitude of the noise from the total noisy signal in the frequency domain. The underlying assumption of the method is that noise is additive and uncorrelated with the signal. In speech signal processing, noise can be measured when there is no signal. In the MDM, however, the magnitude of the noise profile can be estimated only from the magnitude of the Power Spectral Density (PSD) at higher frequencies than the frequency range of the true signal associated with structural vibrations under the additional assumption of white noise. The implementation of the spectral subtraction method to MDM may decrease the energy of the individual mode. In this work, a modification of the spectral subtraction method is introduced that enables the conservation of the energies of individual modes. The main difference is that any (negative) bars with a height below zero after subtraction are set to the absolute value of their height. Both noise reduction methods are implemented in the MDM, and an application example is presented that demonstrates its effectiveness when used with a signal corrupted by noise.

• 한국정보통신학회논문지
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• 제20권9호
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• pp.1673-1678
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• 2016

음성인식처리 분야에서 배경잡음으로 인하여 음성입력이 배경잡음으로 잘못 판단되는 원인이 되어 음성인식율의 저하를 초래한다. 이러한 종류의 잡음대책은 단순하지 않으므로 보다 고도한 잡음처리기술이 필요하게 된다. 따라서 본 논문에서는 잡음환경 중에서 정상적인 배경잡음 혹은 비정상적인 배경잡음과 지속 시간이 짧은 음성을 구별하는 알고리즘에 대하여 기술한다. 본 알고리즘은 다른 종류의 잡음과 음성을 구별하는 중요한 수단으로서 개량된 음성의 특징파리미터를 사용한다. 다음으로 다층퍼셉트론 네트워크에 의하여 잡음의 종류를 추정하는 알고리즘에 대해서 기술한다. 본 실험에서는 잡음과 음성이 구별이 가능하도록 실험적으로 확인하였다.

• 음성과학
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• 제12권3호
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• pp.125-136
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• 2005

Automatic speech recognition is much more difficult in real world. Speech recognition according to SIR (Signal to Interface Ratio) is difficult in situations in which noise of surrounding environment and multi-speaker exists. Therefore, study on main speaker's voice extractions a very important field in speech signal processing in binaural sound. In this paper, we used directional filter and harmonic filter among other existing methods to extract the main speaker's information in binaural sound. The main speaker's voice was extracted using directional filter, and other remaining speaker's information was removed using harmonic filter through main speaker's pitch detection. As a result, voice of the main speaker was enhanced.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2001년도 추계학술발표논문집 (상)
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• pp.801-804
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• 2001

This article is concerned with automatic segmentation of two adjacent vowels for speech signals. All kinds of transition case of adjacent vowels can be characterized by spectrogram. Firstly the voiced-speech is extracted by the histogram analysis of vowel indicator which consists of wavelet low pass components. Secondly given phonetic transcription and transition pattern spectrogram, the voiced-speech portion which has consecutive vowels automatically segmented by the template matching. The cross-correlation function is adapted as a template matching method and the modified correlation coefficient is calculated for all frames. The largest value on the modified correlation coefficient series indicates the boundary of two consecutive vowel sounds. The experiment is performed for 154 vowel transition sets. The 154 spectrogram templates are gathered from 154 words(PRW Speech DB) and the 161 test words(PBW Speech DB) which are uttered by 5 speakers were tested. The experimental result shows the validity of the method.