• 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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• 제12권2호
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• pp.153-170
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• 2005

The sub-Nyquist nonuniform sampling (SNNS) and the perfect reconstruction (PR) formula are proposed for the development of a systematic method to obtain minimal representation of a speech signal. In the proposed method, the instantaneous sampling frequency (ISF) varies, depending on the least upper boundary of spectral support of a speech signal in time-frequency domain (TFD). The definition of the instantaneous bandwidth (IB), which determines the ISF and is used for generating the set of samples that represent continuous-time signals perfectly, is given. Also, the spectral characteristics of the sampled data generated by the sub-Nyquist nonuniform sampling method is analyzed. The proposed method doesn't generate the redundant samples due to the time-varying property of the instantaneous bandwidth of a speech signal.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 하계종합학술대회 논문집(5)
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• pp.89-92
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• 2001

For the diagnosis of arrhythmia in the heart system, the QRS complex of ECG signals is used in many cases. The rejection of the noise in ECG signals is important to acquisition of exact QRS complex. This paper presents some experimental results about instantaneous bandwidth estimation and noise rejection of ECG signals with the purpose of rejection of the 60 Hz power noise and the motion artifacts such as EMG signals and contact noise. ECG signals corrupted by noise are cleaned by using the variable bandwidth filter. For the filtering of ECG signals with noise, the instantaneous bandwidth of the signals is estimated by analysis of time-frequency representation of ECG signal.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(5)
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• pp.9-12
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• 2002

The problem of sub-Nyquist nonuniform sampling for the perfect reconstruction of signals with time-varying spectral contents is studied. The signals are assumed to have a known instantaneous bandwidth in time-frequency domain. As the function of time, the nonuniform sampling pattern of a given signal, that is, the instantaneous sampling frequency is determined by the observation of instantaneous bandwidth based on time-frequency analysis. The proposed sampling pattern guarantees the perfect reconstruction of nonuniform sampled signals under Nyquist-sampling rate in average.

• 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.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2001년도 합동 추계학술대회 논문집 정보 및 제어부문
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• pp.138-141
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• 2001

In this paper, a VBF(variable bandwidth filter) is suggested and analyzed in time-frequency domain. There are four kinds of VBF, which are low-pass VBF, high-pass VBF, band-pass VBF and band-stop VBF. The proposed VBF can extract the components of signals within variable instantaneous bandwidth at a specific time instant. Instantaneous bandwidth is estimated in time-frequency domain. The VBF is represented by rational form in extended Fourier frequency domain.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2208-2211
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• 2003

In this paper, a variable bandwidth filter(VBF) is realized with the purpose of the decomposition of speech signals with time-varying instantaneous of frequencies. The proposed VBF can extract AM-FM components of a speech signal whose time-frequency representations(TFRs) are not overlapped in time-frequency domain

• 한국통신학회논문지
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• 제29권1B호
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• pp.64-72
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• 2004

Multiprotocol Label Switching (MPLS) has been developed as a key technology to enhance the reliability, manageability and overall quality of service of core If networks with connection-oriented tunnel LSP and traffic engineering such as constraint-based routing, explicit routing, and restoration. In this paper, we propose a control bandwidth borrowing scheme that maximizes the utilization of tunnel LSPs or physical links by an extension to the RSVP-TE label distribution protocol. MPLS-based core switching network and VPN services rely on the establishment of connection-oriented tunneled LSPs that are configured or predefined by network management systems. The mechanism of network management system varies from (i) a relatively static LSP establishment accounting, to (ii) a dynamic QoS routing mechanisms. With the use of hierarchical LSPs, the extra bandwidth that is unused by the trunk (outer) LSPs should be fully allocated to their constituent end-to-end user traffic (inner) LSPs in order to maximize their utilization. In order to find out the unused extra bandwidth in tunnel LSP or physical link and redistribute these resources to constituent LSPs, we expend the functionality of RSVP-TE and the found unused extra bandwidth is redistributed with a weight-based recursive redistribution scheme. By the extended RSVP-TE and proposed recursive redistributed scheme, we could achieve the instantaneous maximized utilization of tunnel LSP or physical link suffering from the potential under-utilization problem and guarantee the end-to-end QoS requirements. With the proposed scheme, network manager can manage more effectively the extra available bandwidth of hierarchical LSPs and maximize the instantaneous utilization of the tunneled LSP resources.

• 한국정보과학회논문지:시스템및이론
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• 제34권10호
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• pp.529-538
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• 2007

Generalized Processor Sharing(GPS) 기반의 공정큐잉(Fair Queueing) 알고리즘들은 세션들에게 서비스율과 지연시간 보장 서비스를 제공할 뿐만 아니라, 순시적 공유(instantaneous sharing)를 통해 각 세션에게 공정서비스를 제공한다. 이 공정서비스는 현재 서버에 대기중인 세션들의 과거에 받은 서비스 양에 관계없이 그 세션의 가중치에 비례하여 서버 용량을 분배한다. 그러나 이 공정서비스는 장기적 측면에서 같은 가중치를 가지는 세션에게 세션의 트래픽 패턴에 따라 다른 지연시간과 대역폭 QoS(Quality of Service)를 제공한다. 이러한 장기적 측면의 불공정 서비스를 최소화하기 위해, 본 논문에서는 지연시간과 대역폭 관점에서 서비스 가치(Value of Service)를 정의한 지연시간-대역폭 정규화 모델을 제안한다. 이 정규화 모델에서 정의한 서비스 가치 개념을 사용하여 각 세션에게 지연시간-대역폭 관점의 공정한 서비스를 제공하는 스케줄링 알고리즘을 제안한다. 제안된 알고리즘과 기존의 공정큐잉 및 서비스 커브 기반의 알고리즘과 비교를 통해 제안된 알고리즘은 세션들에게 장기적 측면의 공정서비스를 제공하고, 다양한 트래픽 특성을 갖는 세션에 대해 서비스율과 지연시간 보장에 대한 재조정 없이 동적으로 트래픽 특성에 적응하여 서비스하는 것을 관찰할 수 있다.

• 대한전자공학회논문지
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• 제15권3호
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• pp.1-6
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• 1978

본 논문에서는 음성신호의 디지탈화와 대역폭축소의 한 방법으로 예측부호화 원리를 사용하는 adaptive differential pulse code modulation(ADPCM)과 adoptive delta modulation(ADM)에 관하여 고찰하였다. ADPCM에서 사용되는 대표적인 적응양자기의 원리를 설명하고 적응예측기의 계수를 얻는 두 방법, 즉 브록해석과 연차해석 방법을 검토하였다. 또한 ADM에서 사용되는 세가지 압신방법(instantaneous, syllabic, hybrid commanding)을 구체적으로 설명하고 그의 성능을 비교하였다. 마지막으로 ADPCM과 ADM을 음성신호의 부호화기로 쓸 때의 성능과 장단점들을 비교 검토하였다.