• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.713-724
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• 2006

The purpose of this study is to design, implement and verify the pre-modulation filter with the variable -3dB cutoff frequency and linear phase response for bandlimiting the allocation of radio frequency bandwidth for PCM/FM transmission. For the design of this required filter, the digital FIR filter, DAC system and tuneable 2nd order LPF have been constructed and simulated according to the attenuation characteristic requirement of the amplitude frequency response by each stage. From these results, we have implemented the filter and verified the analog conversion hardware part which is composed of DAC system and tuneable 2nd order LPF for the interpolation of the discrete sequences. Especially this paper proposes and carries out the verification processes using the tone generator and the calibration procedures for more precise frequency response of the filter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1C
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• pp.65-73
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• 2006

The purpose of this study is to design and analyze the pre-modulation filter with the variable -3dB cutoff frequency and linear phase response for bandlimiting the allocation of radio frequency bandwidth in PCM/FM transmission system. For the implementation of this required filter, the digital FIR filter, DAC and variable 2nd order LPF have been constructed with the filter block which designed and analyzed by each stage in order to satisfy the attenuation characteristic requirement of the analog 7th order bessel filter. The paper also concerned the linear phase properties for the filter block. Especially we have carried out the linear phase simulation with real parts for variable 2nd order LPF and compared this simulation results with the one of the fixed bandwidth 2nd order bessel filter for validating the linear phase requirement.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.1
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• pp.202-211
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• 2001

This paper deals with an analysis of multipath which affect a transmission performance in underwater acoustic channel. Underwater acoustic channel with multipath structure is introduced to mathematical modelling for a basin environment. In this paper, SMR(Signal to Multipath Ratio) which is defined as a parameter of multipath's effect is presented as a mathematical equation, and the equation of SMR is simulated by MATLAB program. Furthermore, this paper is also dealt with an implementation of modulation and demodulation system for acoustic transmission. Acoustic Transmission is limited by frequency bandwidth, so $\pi/4 QPSK$(Quadrature Phase Shift Keying) methods which is very useful at frequency ]imitation and FM(Frequency Modulation) are used at acoustic communication system. This implemented hybrid modulation/demodulation system is used as an analog board of image transmission system. In this system, adaptive equalization for reducing the multipath effect and baseline JPEG used for an image compressing are also stated.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.05a
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• pp.419-425
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• 2000

This paper deals with an analysis of multipath which affect a transmission performance in underwater acoustic channel. Underwater acoustic channel with multipath structure is introduced to mathematical modelling for a basin environment. In this paper, SMR(Signal to Multipath Ratio) which is defined as a parameter of multipath's effect is presented as a mathematical equation, and the equation of SMR is simulated by MATLAB program. Furthermore, this paper is also dealt with an implementation of modulation and demodulation system for acoustic transmission Acoustic Transmission is limited by frequency bandwidth so $\pi$/4 Quadrature Phase Shift Keying) methods which is very useful at frequency limitation and FM(Frequency Modulation) are used at acoustic communication system. This implemented hybrid modulation/demodulation system is used as an analog board of image transmission system

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1999.11b
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• pp.249-254
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• 1999

In this paper we analyze the bit error rate of an in-band DAB(Digital Audio Broadcasting) system which is based upon orthogonal frequency division multiplexing(OFDM) and uses M-PSK and M-QAM modulation schemes in the interference environment of an analog FM signal. Results are derived which take into account the influence of the frequency non- selective Rician fading channel, but they are valid for Rayleigh fading channel and AWGN channel as well. Theoretical bit error rates and simulation results are given for system parameters for DAB proposed by ETRI(Electronics and Telecommunications Research Institute).

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.7
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• pp.1025-1035
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• 1995

In this paper, we realize a real time digital FM synthesizer based on genetic algorithm using a general purpose digital signal processor. Especially, we synthesize diverse music sounds nicely using a synthesis model consisting of a single modulator and multiple carriers. Also we present genetic algorithm-based technique which determines optimal parameters for reconstruction through FM synthesis of a sound after analyzing the spectrum of PCM data as a standard music sound using FFT. Using the suggested parameter extractiuon algorithm, we extract parameters of several instruments and then synthesize digital FM sounds. To verify the validity of the parameter extraction algorithm as well as realization of a real time digital music synthesizer, the evaluation is first done by listening the sound directly as subjective test. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

• ETRI Journal
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• v.33 no.2
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• pp.279-282
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• 2011

Coexistence analysis is extremely important in examining the possibility for spectrum sharing between orthogonal frequency-division multiplexing (OFDM)-based international mobile telecommunications (IMT)-Advanced and other wireless services. In this letter, a new closed form method is derived based on power spectral density analysis in order to analyze the coexistence of OFDM-based IMT-Advanced systems and broadcasting frequency modulation (FM) systems. The proposed method evaluates more exact interference power of IMT-Advanced systems in FM broadcasting systems than the advanced minimum coupling loss (A-MCL) method. Numerical results show that the interference power is 1.3 dB and 3 dB less than that obtained using the A-MCL method at cochannel and adjacent channel, respectively. This reduces the minimum separation distance between the two systems, which eventually saves spectrum resources.

• Proceedings of the KIEE Conference
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• 1999.07f
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• pp.2733-2735
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• 1999

The conducted noises mainly emitted from PWM DC power supply have a periodicity with respect to switching frequency, and especially center at the switching frequency and the discrete harmonic frequencies. The concentration of noise energy at these discrete frequencies makes it harder to meet EMI regulations. To reduce the conducted noises, this paper presents the novel FM-PWM control method by using random switching frequency modulation control. This control makes it to generate the noises of side bands by shattering emission spectrum, and to reduce the peak level of conducted noises.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.823-834
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• 2013

In this paper, we propose a detection method for Digital Radio Mondiale (DRM) Plus suitable for hybrid mode broadcasting which services both DRM Plus and analog FM within the same frequency band. The guard-interval correlation method of Orthogonal Frequency Division Multiplexing (OFDM) is good for DRM Plus signal detection, but the possibility for false alarm increases when FM signal is received. The proposed method includes a reference block in the guard-interval correlation which increases the identification rate of weak DRM Plus signals and decreases the possibility of false alarm when analog FM is received. The performance of the proposed method is verified through simulations.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.63-73
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• 1995

The conventional acoustic sounds can be synthesized by Frequency Modulation which includes the variation of frequency, amplitude, and modulation index. In this paper the number of variable synthesis parameters are limited to easily implement the existing two carrier FM algorithm by hardware. The DSP(Digital Signal Processor), which is able to carry out the modified algorithm and synthesize 16 sounds at a time, is designed with $0.8{\mu}m$ standard sells. The DSP which can synthesize 2 sounds at a time is implemented by ASIC emulator to examine the sound quality of the designed DSP. Through the objective and subjective estimation, it is confirmed that the sounds of many instruments from the implemented DSP are very closed to their real sound. Finally the designed DSP is layouted and simulated by VLSI desgn tool. According to the simulation, the designed DSP has the sufficiently fast speed for synthesizing 16 sounds at a time.