• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.8
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• pp.148-155
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• 2014

In this research a signal processing system is designed for detecting the ultrasonic signal envelope using Xilinx's Zynq SoC(system on chip). As a design tool, Vivado IDE(integrated design environment) is used to hierarchically design the whole signal processing system. The proposed system consists of a Zynq-internal ADC, an FIR(finite impulse response) BPF(band pass filter), an absolute value calculator, an FIR LPF(lpw pass filter), and the Kalman filter. Under this configuration, two design schemes, HW design scheme with LPF as a final stage and HW/SW co-design scheme with a Kalman filter as a final stage, are compared in terms of the performance and efficiency. As a result, envelope detecting performances of the two schemes are proved to be almost same, but the HW/SW co-design is verified to be much more efficient than the HW design considering the much smaller time consumption during system design.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.8
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• pp.1818-1826
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• 2014

Time-variant sea surface causes a forward scattering and Doppler spreading in received signal on underwater acoustic communication system. This results in time-varying amplitude, frequency and phase variation of the received signal. In such a way the channel coherence bandwidth and fading feature also change with time. Consequently, the system performance is degraded and high-speed coherent digital communication is disrupted. In this paper, quadrature phase shift keying (QPSK) performance is examined in two different sea surface conditions. The impact of sea surface scattering on performance is analyzed on basis of the channel impulse response and temporal coherence using linear frequency modulation (LFM) signal. The impulse response and the temporal coherence of the rough sea surface condition were more unstable and less than that of the calm sea surface condition, respectively. By relating these with time variant envelope, amplitude and phase of received signal, it was found that the bit error rate (BER) of QPSK are closely related to time variation of sea surface state.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.77-85
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• 2001

This paper proposes a method of time-scale modification of polyphonic audio signals based on a sinusoidal model. The signals are modeled with sinusoidal component and noise component. A multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in time-scale modification a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. For extracting sinusoidal components and calculating their parameters matching pursuit algorithm is applied to each analysis frame of subband signal. In accordance with spectrum analysis a psychoacoustic model implementing the effect of frequency masking is incorporated with matching pursuit to provide a resonable stop condition of iteration and reduce the number of sinusoids. The noise component obtained by subtracting the synthesized signal with sinusoidal components from the original signal is modeled by line-segment model of short time spectrum envelope. For various polyphonic audio signals the result of simulation shows suggested sinusoidal modeling can synthesize original signal without loss of perceptual quality and do more robust and high quality time-scale modification for large scale factor because of representing transients without any perceptual loss.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.70-77
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• 2017

Bandwidth extension is a technique to improve speech quality, intelligibility and naturalness, extending from the 300 ~ 3,400 Hz narrowband speech to the 50 ~ 7,000 Hz wideband speech. In this paper, an Artificial Bandwidth Extension (ABE) module embedded in the Opus audio decoder is designed using the information of narrowband speech to reduce the computational complexity of LPC (Linear Prediction Coding) and LSF (Line Spectral Frequencies) analysis and the algorithm delay of the ABE module. We proposed a spectral envelope extension method using DBN (Deep Belief Network), one of deep learning techniques, and the proposed scheme produces better extended spectrum than the traditional codebook mapping method.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.7
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• pp.987-993
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• 2013

Integrity of the control and instrumental cables in nuclear power plant is important to maintain the stability of the nuclear power plants. In order to diagnose the integrity of the cables, the diagnostic methods based on reflectometry have been studied. The reflectometry is a non-destructive method and it is applicable to diagnose the live cables. We introduce a Gaussian enveloped linear chirp reflectometry to diagnose the cables in the nuclear power plants. In this paper, we estimate the instantaneous frequency of the Gaussian enveloped linear chirp signal by using the weighted robust least squares filtering to localize the impedance discontinuities in the class 1E instrumental cable.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.3
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• pp.582-590
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• 1994

This paper introduce a highly verstile and simple data transmission system designed for commercial power distribution lines. The system operates on the DSSS principle but utilizes two independent carrier frequencies each represents polarity of DS MODEM outiputs. At the receiving terminal, outputs of two envelope detectors are directly applied to separate DS correlators before the two components are compared. The recovered signals which represents data and line noise are then compared at comparator. With the noise power greatly rudused at the correlator, the error rate of the data observed at comparator desplays great improvement comparing to the conventional FSK-DS system in which the detector output are compared before the correlator stage. Despite its simplest structure, the prototype MODEM transmitts 2400 bps with the error rate 10 , about 10dB improved compared to conventional FSK system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.8
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• pp.1847-1853
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• 2014

The PAPR of the input signal is increased due to OFDMA signal in a mobile communication system. High efficiency of a power amplifier, which accounts for power consumption, is a very important key technology. Digital Pre-Distortion techniques were used to improve the linearity of the power amplifier. The Asymmetric Doherty scheme was used to improve the efficiency of the power amplifier. In this paper, we propose a new structure of Asymmetric Doherty. Drive power amplifier part is separated as main path and peak path, and phase shifter is employed to improve power combine characteristics of the Doherty Amplifier. Also, envelope tracking technology for drive gate bais in drive peak amplifier is used to improve efficiency.

• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.133-140
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• 2010

We improve the performance of cardiac disorder classification by adding new temporal features extracted from continuous heart sound signals. We add three kinds of novel temporal features to a conventional feature based on mel-frequency cepstral coefficients (MFCC): Heart sound envelope, murmur probabilities, and murmur amplitude variation. In cardiac disorder classification and detection experiments, we evaluate the contribution of the proposed features to classification accuracy and select proper temporal features using the sequential feature selection method. The selected features are shown to improve classification accuracy significantly and consistently for neural network-based pattern classifiers such as multi-layer perceptron (MLP), support vector machine (SVM), and extreme learning machine (ELM).

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.17-23
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• 2004

The mobile radio channel can be simulated as a complex-valued random process with narrow-band spectrum. This paper describes a real-time implementation of that process using a INS320C6414 digital signal processor and XC2VP30 Virtex FPGA. The simulator presented here is not only a comprehensive model of the flat fading but also frequency selective fading mobile channel conditions. To replicate the statistical characteristics of the multipath fading environment with the minimum computational burden, multi-rate techniques are employed to resolve practical problems such as variable sampling rate. The simulator produces accurate and consistent results due to digital implementation. It is very flexible and simple to program for various field conditions in mobile communications with a graphical user interface.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.8
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• pp.851-856
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• 2010

This paper presents an efficiency enhancement for the balanced power amplifier using DBS(Dynamic Bias Switching) method which dynamically provides the power amplifier with two bias voltage levels according to the input envelope signal. In order to apply the dynamic biases to each side of the balanced power amplifier, two switching stages are adopted. Using an OFDM signal with a bandwidth of 20 MHz and a PAR(Peak to Average Ratio) of 8.5 dB, 6 % of PAE(Power-Added Efficiency) is improved at an output power of 42.5 dBm.