• 대한전기학회논문지:전력기술부문A
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• 제48권5호
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• pp.643-649
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• 1999

In this paper, we proposed that ｉ) noise-tolerant four kinds of AM(Amplitude Modulation)-FM(Frequency Modulation) demodulators are designed, ⅱ) we derived undersampling frequency through the product via energy operator of the monocomponent AM-FM signals separated form two-component AM-FM signals, and ⅲ) these four kinds of AM-FM demodulators detect respectively information signals of the IA(Instantaneous Amplitude) and IF(Instantaneous Frequency) by undersampling frequency to be different each other from the undersampled monocomponet AM-FM signals. Particularly, the proposed algorithm can control undersampling frequency by an integer factor. And these efficient AM-FM demodulators are well worked with the undersampled AM-FM signals.

• 전기전자학회논문지
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• 제24권4호
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• pp.1081-1085
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• 2020

변곡점 추출방법을 활용하여 AM-FM신호의 순간주파수를 추정하는 방법을 제안한다. 제안된 방법은 AM-FM신호의 국부적인 최대값/최소값을 변곡점 추출법을 활용하여 검출하며, 검출된 인접한 최대값/최소값 사이의 시간 간격을 계산하여 FM성분의 순간주파수를 추정한다. AM성분의 경우, 검출된 최소값/최대값의 포락선을 먼저 구하고 포락선의 최고값 간의 시간을 계산하여 순간주파수를 추정한다. 컴퓨터 모의실험으로 FM성분의 주파수가 고정된 경우와 변화하는 경우에 제안된 방법의 유용성을 검증한다. FM성분의 주파수가 일정한 AM-FM신호의 경우, 제안된 방법은 두 주파수 성분을 적절하게 추정한다. FM성분의 주파수가 변화하는 AM-FM신호의 경우, AM성분의 순간주파수는 적절히 추정되었으나 FM성분의 순간 주파수는 이론적인 주파수에서 편이되는 현상을 보인다. 이는 국소적인 최고점/최대점 검출로 표본수가 적어지는 효과에 따른 것으로, 저역통과 필터와 같은 후처리로 개선될 수 있다.

• 음성과학
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• 제12권4호
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• pp.53-70
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• 2005

A Vowel of speech signals are multicomponent signals composed of AM-FM components whose instantaneous frequency and instantaneous amplitude are time-varying. The changes of emotion states cause the variation of the instantaneous frequencies and the instantaneous amplitudes of AM-FM components. Therefore, it is important to estimate exactly the instantaneous frequencies and the instantaneous amplitudes of AM-FM components for the extraction of key information representing emotion states and changes in speech signals. In tills paper, firstly a method decomposing speech signals into AM - FM components is addressed. Secondly, the fundamental frequency of vowel sound is estimated by the simple method based on the spectrogram. The estimate of the fundamental frequency is used for decomposing speech signals into AM-FM components. Thirdly, an estimation method is suggested for separation of the instantaneous frequencies and the instantaneous amplitudes of the decomposed AM - FM components, based on Hilbert transform and the demodulation property of the extended Fourier transform. The estimates of the instantaneous frequencies and the instantaneous amplitudes can be used for modification of the spectral distribution and smooth connection of two words in the speech synthesis systems based on a corpus.

• 음성과학
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• 제8권4호
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• pp.45-58
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• 2001

Modulated components of a speech signal are frequently used for speech coding, speech recognition, and speech synthesis. Time-frequency representation (TFR) reveals some information about instantaneous frequency, instantaneous bandwidth and boundary of each component of the considering speech signal. In many cases, the extraction of AM-FM components corresponding to instantaneous frequencies is difficult since the Fourier spectra of the components with time-varying instantaneous frequency are overlapped each other in Fourier frequency domain. In this paper, an efficient method decomposing speech signal into AM-FM components is proposed. A variable bandwidth filter is developed for the decomposition of speech signals with time-varying instantaneous frequencies. The variable bandwidth filter can extract AM-FM components of a speech signal whose TFRs are not overlapped in timefrequency domain. Also, amplitude and instantaneous frequency of the decomposed components are estimated by using Hilbert transform.

• 음성과학
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• 제12권4호
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• pp.17-29
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• 2005

Time-frequency representation reveals some useful information about instantaneous frequency, instantaneous bandwidth and boundary of each AM-FM component of a speech signal. In many cases, the instantaneous frequency of each component is not constant. The variability of instantaneous frequency causes degradation of resolution in time-frequency representation. This paper presents a method of adaptively adjusting the transform kernel for preventing degradation of resolution due to time-varying instantaneous frequency. The transform kernel is the form of frequency modulated function. The modulation function in the transform kernel is determined by the estimate of instantaneous frequency which is approximated by first order polynomial at each time instance. Also, the window function is modulated by the estimated instantaneous. frequency for mitigation of fringing. effect. In the proposed method, not only the transform kernel but also the shape and the length of. the window function are adaptively adjusted by the instantaneous frequency of a speech signal.

• 대한전자공학회논문지TC
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• 제41권12호
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• pp.51-60
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• 2004

디지털 오디오 방송은 기존의 AM/FM 방송에 이은 고품질의 차세대 라디오 방송으로 CD수준의 음질, 다양한 데이터 서비스, 양방향성 및 우수한 이동 수신 품질 등을 제공하며 지상파 방송, 위성 방송, 케이블 TV 및 인터넷 등 다양한 매체를 통하여 서비스가 가능한 뉴미디어 서비스이다. 본 논문에서는 DAB(Digital Audio Broadcasting) 시스템이 오디오 방송뿐만 아니라 다양한 멀티미디어 서비스가 가능하다는 장점과 인터넷 단말기로서의 PC를 결합한 새로운 형태의 DAB 기반 EPG(Electronic Program Guide) 응용 기술에 대하여 연구하였다. 개발된 EPSD(EPG Provider System on DAB)시스템은 웹 기반의 서버/클라이언트 구조를 가지며 서버에서 클라이언트 PC로 다양한 EPG 기능을 제공한다. 따라서 DAB 수신기가 보다 저렴하고 소형화될 수 있으며 인터넷과 연결되어 풍부한 데이터 서비스의 개발이 가능하게 된다. 특히 고품질 영상 서비스가 가능하고 기존의 디지털 장비와의 연결 또한 수월하여 추후 홈 네트워크의 중요한 단말 장치로서의 역할을 담당할 것이다.