• Journal of the Korean Institute of Telematics and Electronics A
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• v.29A no.4
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• pp.74-82
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• 1992

This paper deals with the scheduling problems, which are the most important subtask in High-level Synthesis. ILP(integer linear programming) formulations are used as a scheduling problem approach.For practical application to digital system design, we have concentrated our attentions on pipelined datapath scheduling. For experiment results, we choose the 5-th order digital wave filter as a benchmark and do the schedule. Finally, we can obtain better and near-optimal scheduling results.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.91-94
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• 1994

This paper is a study on the design and implementation of the Korean Tet-to-Speech system. In this paper, parameter symthesis method is chosen for speech symthesis method and PARCOR coeffient, one of the LPC analysis, is used as acoustic parameter, We use a diphone as synthesis unit, it include a basic naturalness of human speech. Diphone DB is consisted of 1228 PCM files. LPC synthesis method has defect that decline clearness of synthesis speech, during synthesizing unvoiced sound In this paper, we improve clearness of synthesized speech, using residual signal as ecitation signal of unvoiced sound. Besides, to improve a naturalness, we control the prosody of synthesized speech through controlling the energy and pitch pattern. Synthesis system is implemented at PC/486 and use a 70Hz-4.5KHz band pass filter for speech imput/output, amplifier and TMS320c30 DSP board.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1299-1304
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• 2000

In a speed coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including UnVoiced Consonant) searching, extraction ad approximation-synthesis method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This method based on a zerocrossing rate and pitch detector using FIR-STREAK Digital Filter. As a result, the extraction rates of TSIUVC are 84.8% (plosive), 94.9%(fricative), 92.3%(affricative) in female voice, and 88%(plosive), 94.9%(fricative), 92.3%(affricative) in male voice respectively, Also, I obain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. This method has the capability of being applied to speech coding of low bit rate, speech analysis and speech synthesis.

• The KIPS Transactions:PartB
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• v.12B no.4 s.100
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• pp.379-386
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• 2005

This paper provides a rigorous theory for analysis of quantization effects and optimum filter bank design in quantized multidimensional subband filter banks. Even though subband filter design has been a hot topic for last decades, a few results have been reported on the subband filter with a quantizer. Each pdf-optimized quantizer is modeled by a nonlinear gain-plus-additive uncorrelated noise and embedded into the subband structure. Using polyphase decomposition of the analysis/synthesis filter banks, we derive the exact expression for the output mean square quantization error. Based on the minimization of the output mean square error, the technique for optimal filter design methodology is developed. Numerical design examples for optimum nonseparable paraunitary and biorthogonal filter banks are presented with a quincunx subsampling lattice. Through the simulation, $10\~20\;\%$ decreases in MSE have been observed compared with subband filter with no quantizers especially for low bit rate cases.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.17 no.2 s.105
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• pp.95-100
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• 2006

In this paper, we designed amplitude equalizer which is composed of amplifier, complementary shaping filter and attenuator in order to improve flatness of high order bandpass filter. We modified Chebyshev polynomial and calculated the prototype elements for complementary shaping filters by network synthesis. The amplitude equalizer is realized that it connects the 4th order complementary shaping filter designed by using calculated the prototype elements to the amplifier compensating for insertion loss and improving return loss, and with the attenuator for gain control. Using proposed amplitude equalizer, We certificated improvement in flatness of 13th order bandpass filter at WiBro band.

• Smart Structures and Systems
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• v.6 no.2
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• pp.115-133
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• 2010

A modal filter is a tool used to extract the modal coordinates of each individual mode from a system's output. This is achieved by mapping the response vector from the physical space to the modal space. It decomposes the system's responses into modal coordinates, and thus, on the output of the filter, the frequency response with only one peak corresponding to the natural frequency to which the filter was tuned can be obtained. As was shown in the paper (Deraemecker and Preumont 2006), structural modification (e.g. a drop in stiffness or mass due to damage) causes the appearance of spurious peaks on the output of the modal filter. A modal filter is, therefore, a great indicator of damage detection, with such advantages as low computational effort due to data reduction, ease of automation and lack of sensitivity to environmental changes. This paper presents the application of modal filters for the detection of stiffness changes. Two experiments were conducted: the first one using the simulation data obtained from the numerical 7DOF model, and the second one on the experimental data from a laboratory stand in 4 states of damage.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.557-562
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• 2003

An output multiplexer of manifold type is widely used in a recent satellite transponder for its mass and volume reduction. For correct operation, the filters of such a multiplexer must be singly terminated. In this paper, a simple synthesis method of a 6-pole dual-mode singly-terminated filter is described. From the transfer function of the filter, network parameters such as in/output terminations and coupling coefficients are obtained easily without complicated matrix algebra such as orthogonal projection and similarity transformation. Two different-structure filters are taken into consideration and the network parameters of each filter have been extracted from the same transfer function. It is shown that the responses of two filters are same to each other since their network parameters are obtained from the same transfer function. The method described in this paper can be applied to the other degree singly terminated filter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.12A
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• pp.1214-1218
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• 2010

A new compact UWB bandpass filter with sharp-rejection and low insertion-loss is designed in this paper, using the parallel transmission line with SIR structure. An Example filter is firstly theoretically designed by the proposed synthesis procedure, the optimized by EM simulator, fabricated in microstrip line and finally characterized by a network analyzer. A good agreement between the simulation results and prototype validates the proposed UWB bandpass filter. The designed filter also realized a sharp rejection of higher order with a relatively small size which is proper for the UWB communication system.

• The Transactions of the Korea Information Processing Society
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• v.3 no.4
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• pp.938-946
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• 1996

Hierarchical subband codec has been widely promoted in the field of data compression/decompression because of their simplicity and modular nature. Over the past years, the study has received great attention to the perfect reconstruction (PR)system which perfectly recovers the original input signal at the reconstructed output. However, in the actual subband codec system, the signals that passed through the analysis filter bank are quantized before transmission to the receiver side and reconstructed by the synthesis filter bank. Thus the PR system is impossible and the quantization effects must be carefully considered in the system design such that the system recovers the reconstructed output as possible to the the original input signal with minimum quantization error.In this paper, we propose an optimum hierarchical subband codec structure in the presence of quantizer. The optimality criteria of the code is given to the deign of the hierarchical analysis/synthesis subband filter bank and the quantizer that minimize then output mean square error due to the quantizer in the codec. Specific opti-mum design esamples are shown with level-1, level-2 hierarchical structure. The optimal designs are verified by computer simulation.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.223-230
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• 2002

In a speech coding system using excitation source of voiced and unvoiced, it would be involves a distortion of speech waveform in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new method of TSIUVC (Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. The TSIUVC extraction is based on a zero crossing rate and IPP (Individual Pitch Pulses) extraction algorithm using residual signal of FIR-STREAK Digital Filter. As a result, This method obtain a high Quality approximation-synthesis waveform by using Least Mean Square. The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.