• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.26 no.4
• /
• pp.556-561
• /
• 2022

In this paper, we present a block media communication (BMC) system which employs powerline communication to the equipments used in the welding process for ship-assembly and uses metal block as a communication medium. Inductive couplers are installed on digital feeder and pin jig. Information signal is added to the current generated by the welding gun, and applied to the block. When the welding operation starts, information generated in the field is transmitted to the monitoring server in real-time. The field test on the BMC system confirms that the transmitted data are correctly received at the server. Since the proposed system can be built without any changes to the existing welding process, it is helpful to increase competitiveness of the shipbuilding industry through smart factory of shipyards. It is also possible to quickly respond to emergency situations that may occur to workers in an electromagnetic wave shielding environment or a closed space, the effect of preventing industrial accidents will be great.

• Journal of Korean Academy of Oral and Maxillofacial Radiology
• /
• v.28 no.2
• /
• pp.387-413
• /
• 1998

Image resampling is of particular interest in digital radiology. When resampling an image to a new set of coordinate, there appears blocking artifacts and image changes. To enhance image quality, interpolation algorithms have been used. Resampling is used to increase the number of points in an image to improve its appearance for display. The process of interpolation is fitting a continuous function to the discrete points in the digital image. The purpose of this study was to determine the effects of the seven interpolation functions when image resampling in digital periapical images. The images were obtained by Digora, CDR and scanning of Ektaspeed plus periapical radiograms on the dry skull and human subject. The subjects were exposed to intraoral X-ray machine at 60kVp and 70 kVp with exposure time varying between 0.01 and 0.50 second. To determine which interpolation method would provide the better image, seven functions were compared; (1) nearest neighbor (2) linear (3) non-linear (4) facet model (5) cubic convolution (6) cubic spline (7) gray segment expansion. And resampled images were compared in terms of SNR(Signal to Noise Ratio) and MTF(Modulation Transfer Function) coefficient value. The obtained results were as follows ; 1. The highest SNR value(75.96dB) was obtained with cubic convolution method and the lowest SNR value(72.44dB) was obtained with facet model method among seven interpolation methods. 2. There were significant differences of SNR values among CDR, Digora and film scan(P<0.05). 3. There were significant differences of SNR values between 60kVp and 70kVp in seven interpolation methods. There were significant differences of SNR values between facet model method and those of the other methods at 60kVp(P<0.05), but there were not significant differences of SNR values among seven interpolation methods at 70kVp(P>0.05). 4. There were significant differences of MTF coefficient values between linear interpolation method and the other six interpolation methods (P< 0.05). 5. The speed of computation time was the fastest with nearest -neighbor method and the slowest with non-linear method. 6. The better image was obtained with cubic convolution, cubic spline and gray segment method in ROC analysis. 7. The better sharpness of edge was obtained with gray segment expansion method among seven interpolation methods.

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.45 no.5
• /
• pp.1-11
• /
• 2008

This work proposes a 15b 50MS/s CMOS pipeline ADC based on digital code-error calibration. The proposed ADC adopts a four-stage pipeline architecture to minimize power consumption and die area and employs a digital calibration technique in the front-end stage MDAC without any modification of critical analog circuits. The front-end MDAC code errors due to device mismatch are measured by un-calibrated back-end three stages and stored in memory. During normal conversion, the stored code errors are recalled for code-error calibration in the digital domain. The signal insensitive 3-D fully symmetric layout technique in three MDACs is employed to achieve a high matching accuracy and to measure the mismatch error of the front-end stage more exactly. The prototype ADC in a 0.18um CMOS process demonstrates a measured DNL and INL within 0.78LSB and 3.28LSB. The ADC, with an active die area of $4.2mm^2$, shows a maximum SNDR and SFDR of 67.2dB and 79.5dB, respectively, and a power consumption of 225mW at 2.5V and 50MS/s.

• Journal of the Institute of Electronics Engineers of Korea SC
• /
• v.43 no.5 s.311
• /
• pp.80-88
• /
• 2006

In this paper we developed a novel system for automatic detection of mass type breast cancer on dense digital mammogram images. The new approaches presented in this paper are as follows: 1) we presented a method that stably decides the mass center and radius without being affected by image signal irregularity. 2) We developed a radial directional filter that is suitable to process mass image signal. 3) And we developed the multiple feature function based on mass shape spiculation, mass center homogeneity, and mass eccentricity, so as to determine mass-type breast cancer. When the proposed system is applied to dense mammographic images, the true 기arm rate is improved by 10% over a conventional system while the false alarm is increased by 1 per image.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2014.06a
• /
• pp.157-159
• /
• 2014

In this paper, we introduce SlimSRP, an ultra-low-power digital signal processor (DSP) solution for mobile audio and voice applications. So far, application processors (APs) have taken charge of all the tasks in mobile devices. However, they have suffered from short battery life problems to deal with complex usage scenarios, such as always-on voice trigger with continuous audio playback. From extensive analysis of audio and voice application characteristics, SlimSRP is designed to relive the performance and power burden of APs. It employs three-issue VLIW architecture, and the major low-power and high-performance techniques include: (1) an optimized register-file architecture friendly for constants generation, (2) a powerful instruction set to reduce the number of register file accesses and (3) a unique instruction compression scheme that contributes to saved memory size and reduced cache miss. An implementation of SlimSRP runs at up to 200MHz and the logic occupies 95K NAND2 gates in Samsung 28LPP process. The experimental results demonstrate that a MP3 decoder application with a 128kbps 44.1kHz input can run at 5.1MHz and the logic consumes only 22uW/MHz.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.21 no.7
• /
• pp.1342-1348
• /
• 2017

In an era where digital data takes on great importance, images are essential to various media. Noise is generated during the acquisition and transmission of such images, caused by a number of external factors. The removal of noise is an essential step in image processing. There are various methods used to remove noise, in accordance with the cause or form of the noise. AWGN is one of the leading methods. As such, this paper applies the edge detection method using the mean of each pixel after categorizing in detail the partial masks into nine areas as part of the preliminary process, in order to minimize noise that had been added to the image. In addition, the paper suggests an algorithm that applies different filters to the partial masks by using the critical mass value of the transfigured edge detection. To verify the competence of the suggested algorithm, it was compared with existing methods by using magnified images and PSNR(peak signal to noise ratio).

• Journal of Korea Multimedia Society
• /
• v.14 no.11
• /
• pp.1392-1400
• /
• 2011

In most digital imaging applications, high-resolution images or videos are usually desired for later image processing and analysis. The image signal obtained from general imaging system occurs image degradation during the process of image acquirement caused by the optics, physical constraints and the atmosphere effects. Super-resolution reconstruction, one of the solution to address this problem, is image reconstruction technique that produces a high-resolution image from several low-resolution frames in video sequences. In this paper, we propose an improved super-resolution method using Projection onto Convex Sets (POCS) method based on Shift & Add (S&A). The image using conventional algorithms is sensitive to noise. To solve this problem, we propose a fusion algorithm of S&A and POCS. Also we solve the problem using BLPF (Butterworth Low-pass Filter) in frequency domain as optical blur. Our method is robust to noise and has sharpness enhancement ability. Experimental results show that the proposed super-resolution method has better resolution enhancement performance than other super-resolution methods.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.9 no.12
• /
• pp.1337-1344
• /
• 2014

The objective of an echo canceller connected to any end of a communication line such as digital subscriber line (DSL) is to compensate the outgoing transmit signal in the receiving path that the hybrid circuit leaks. The echo canceller working in a full duplex environment is an adaptive system driven by the local signal. Conventional echo canceller that implement the least mean square (LMS) algorithm provides a low computational burden but poor convergence properties. The length of the echo canceller will directly affect both the degree of performance and the convergence speed of the adaptation process. To cancel long time-varying echoes, the number of tap coefficients of a conventional echo canceller must be large, which decreases the convergence speed of the adaptive filter. This paper proposes an alternative technique for the echo cancellation in a telecommunication channel. The new technique employs the bi-directional least mean square (LMS) algorithm for adaptively computing the optimal set of the coefficients of the echo canceller, which is composed of weighted combination of both feedforward and feedback algorithms. Finally, Simulation results as well as mathematical analysis demonstrates that the proposed echo canceller has faster convergence speed than the conventional LMS echo canceller with nearly equivalent complexity of computation.

• Journal of Ocean Engineering and Technology
• /
• v.24 no.1
• /
• pp.166-171
• /
• 2010

In this paper, the front-end of a digital receiver with a 25 kHz carrier frequency, 5 kHz symbol rate, and any excess-bandwidth is designed using two basic facts. The first is known as the uniform sampling theorem, which states that the sampled sequence might not suffer from aliasing even if its sampling rate is lower than the Nyquist sampling rate if the analog signal is a bandpass one. The other fact is that if the sampling rate is 4 times the center frequency of the sampled sequence, the front-end processing complexity can be dramatically reduced due to the half of the sampled sequence to be multiplied by zero in the demixing process. Furthermore, the designed front-end is simplified by introducing sub-filters and sub-sampling sequences. The designed front-end is composed of an A/D converter, which takes samples of a bandpass filtered signal at a 20 kHz rate; a serial-to-parallel converter, which converts a sampled bandpass sequence to 4 parallel sub-sample sequences; 4 sub-filter blocks, which act as a frequency shifter and lowpass filter for a complex sequence; 4 synchronized switches; and 2 adders. The designed front-end dramatically reduces the computational complexity by more than 50% for frequency shifting and lowpass filtering operations since a conventional front-end requires a frequency shifting and two lowpass filtering operations to get one lowpass complex sample, while the proposed front-end requires only four filtering operation to get four lowpass complex samples, which is equivalent to one filtering operation for one sample.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2011.05a
• /
• pp.827-830
• /
• 2011

In this paper, a wind speed & direction module and the DSP(Digital Signal Processor) sensor interface circuit board are proposed. This DSP system accepts and process the informations from a wind speed & direction module, the atmospheric pressure sensor, the ambient air temperature sensor and transfers it to the PC monitering system. Especially, a wind speed & direction module and a DSP hardware are directly designed and applied. A wind speed & direction module have a construction that it have four film type RTD(Resistive Temperature Detectors) or film type Piezo sensors adhered around the circular metal body for obtaining vector informations about wind. By this structure, the module is enabled precise measurement having a robustness about vibration, humidity, corrosion. A sensor signal processing circuit is using TMS320F2812 TI(Texas Instrument) Corporation high speed DSP.