• Title/Summary/Keyword: AM-FM Component

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Digital Demodulator Design and Characteristics Using Algebraic Separation and Energy Operator from Undersampled Two-Component AM-FM Signals (저표본화된 주성분의 AM-FM 신호들로부터 대수적 분리와 에너지 연산자를 사용한 복조기 설계 및 특성)

  • Sohn, Tae-Ho;Lee, Min-Ho
    • The Transactions of the Korean Institute of Electrical Engineers A
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    • v.48 no.5
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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.

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Instantaneous Frequency Estimation of AM-FM Signals using the Inflection Point Detection (변곡점 검출을 이용한 AM-FM 신호의 순간주파수 추정)

  • Iem, Byeong-Gwan
    • Journal of IKEEE
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    • v.24 no.4
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    • pp.1081-1085
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    • 2020
  • Instantaneous frequencies (IF) of the AM-FM signal is estimated based on the inflection point detection (IPD) method. Local maxima/minima are detected using the IPD, and they are exploited to find the IF of AM and FM components, respectively. The envelope of the maxima/minima is obtained to estimate the IF of the AM part. And the distance between neighboring maxima (or minima) is used to estimate the IF of the FM component. Computer simulation shows that the proposed method properly estimates the IF of the AM and FM when the signal has fixed frequencies for both parts. In the case of the time-varying IF of the FM part, the estimated IF shows some deviation from the true IF due to the rough sampling effect of the maximum/minimum points. Thus, the post-processing such as the lowpass filtering of the estimated IF is required to refine the resulting IF estimation.

AM-FM Decomposition and Estimation of Instantaneous Frequency and Instantaneous Amplitude of Speech Signals for Natural Human-robot Interaction (자연스런 인간-로봇 상호작용을 위한 음성 신호의 AM-FM 성분 분해 및 순간 주파수와 순간 진폭의 추정에 관한 연구)

  • Lee, He-Young
    • Speech Sciences
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    • v.12 no.4
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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.

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Decomposition of Speech Signal into AM-FM Components Using Varialle Bandwidth Filter (가변 대역폭 필터를 이용한 음성신호의 AM-FM 성분 분리에 관한 연구)

  • Song, Min;Lee, He-Young
    • Speech Sciences
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    • v.8 no.4
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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.

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On Improving Resolution of Time-Frequency Representation of Speech Signals Based on Frequency Modulation Type Kernel (FM변조된 형태의 Kernel을 사용한 음성신호의 시간-주파수 표현 해상도 향상에 관한 연구)

  • Lee, He-Young;Choi, Seung-Ho
    • Speech Sciences
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    • v.12 no.4
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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.

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Development of the EPG Provider System based on DAB (DAB 기반의 EPG Provider 시스템 개발)

  • Jin Hyun-Joon;Park Nho-Kyung;Hwang Woon-Jae
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.41 no.12
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    • pp.51-60
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    • 2004
  • DAB(Digital Audio Broadcasting) is a new media service that can provide CD quality audio, various data service, interactive and high quality mobile communications through popular media such as terrestrial broadcasting, satellite, cable TV, and internet. In this paper, a new EPG(Electronic Program Guide) application model is proposed. The model is based on DAB and combines a DAB receiver and PCs so that it can take advantages of using various multimedia services and plenty of internet contents. The developed EPSD(EPG Provider System on DAB) has Web-based Server/Client structure and povides EPG functionalities to client PCs over internet. Therefore, the DAB receiver can be smaller and cheaper, and can develop abundant data services on internet. It can also provide high quality video services and be expected to become an important component in future home network systems.