• Speech Sciences
• /
• v.8 no.4
• /
• pp.45-58
• /
• 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.

• Speech Sciences
• /
• v.12 no.2
• /
• pp.153-170
• /
• 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.

• Proceedings of the IEEK Conference
• /
• 2001.06e
• /
• pp.89-92
• /
• 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.

• Proceedings of the IEEK Conference
• /
• 2002.06e
• /
• pp.9-12
• /
• 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.

• Speech Sciences
• /
• v.12 no.4
• /
• pp.17-29
• /
• 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.

• Proceedings of the KIEE Conference
• /
• 2001.11c
• /
• pp.138-141
• /
• 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.

• Proceedings of the IEEK Conference
• /
• 2003.07e
• /
• pp.2208-2211
• /
• 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

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.1B
• /
• pp.64-72
• /
• 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.

• Journal of KIISE:Computer Systems and Theory
• /
• v.34 no.10
• /
• pp.529-538
• /
• 2007

Fair Queueing algorithms based on Generalized Processor Sharing (GPS) not only guarantee sessions with service rate and delay, but also provide sessions with instantaneous fair sharing. This fair sharing distributes server capacity to currently backlogged sessions in proportion to their weights without regard to the amount of service that the sessions received in the past. From a long-term perspective, the instantaneous fair sharing leads to a different quality of service in terms of delay and bandwidth to sessions with the same weight depending on their traffic pattern. To minimize such long-term unfairness, we propose a delay-bandwidth normalization model that defines the concept of value of service (VoS) from the aspect of both delay and bandwidth. A model and a packet-by-packet scheduling algorithm are proposed to realize the VoS concept. Performance comparisons between the proposed algorithm and algorithms based on fair queueing and service curve show that the proposed algorithm provides better long-term fairness among sessions and that is more adaptive to dynamic traffic characteristics without compromising its service rate and delay guarantees.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.15 no.3
• /
• pp.1-6
• /
• 1978

This paper deals with speech digitization and bandwidth compression techniques, particularly two predictive coding methods-namely, adaptive diferentia1 pulse code modulation(ADPCM) and adaptive delta modulation (ADM). The principle of a typical adoptive quantizer that is used in ADPCM is explained, and two analysis methods for the adaptive predictor coefficents, block and sequential analyses, are discussed. Also, three companding methods (instantaneous, syllabic, and hybrid companding) that are used in ADM are explained in detail, and their performances are compared. In addition, the performances of ADPCM and ADM as speech coders are compared, and the merits of each coder are discussed.