• Journal of the Institute of Convergence Signal Processing
• /
• v.19 no.3
• /
• pp.89-96
• /
• 2018

This study aims to develop a fully passive smart sensing tag utilizing RF (Radio Frequency) energy harvesting technology at UHF (Ultra High Frequency) band of 915MHz. To optimize the power collected under various radiated conditions, an efficient energy harvesting module exploiting a boost circuit with maximum power point tracking (MPPT) is employed. Specifically, the proposed tag features two orthogonal antennas to enhance its capability of both energy scavenging and data transmissions. The experimental result shows that the developed smart sensor tag can scavenge an RF input power of as low as 0.19mW at a distance of 4 meters for a 3.6Vdc output. Furthermore, the proposed smart sensor tag performs the feasibility of completely autonomous monitoring food freshness at 2 meters with a low-power sensor array.

• Science of Emotion and Sensibility
• /
• v.9 no.4
• /
• pp.377-383
• /
• 2006

The purpose of this research is to develop a design model of a smart jacket incorporating MP3 functionality for commercial use, as one of the digital lines of textile to apply If technology. In order to conduct this study, the previous studies were analyzed and usability and wearability tests were conducted using five prototype design models of smart wear with MP3 functionality. Accordingly, the smart jacket incorporating MP3 functionality for commercial use, for which user-centered interface, usability and wearability were completed, was developed by optimizing the design and location of the input interface, control module, the path of wires and earphone etc., based on textile wire and keypad. The smart jacket targets the female dress market from late-teens to early twenties. It will be a high value commodity considers all aspects of recent fashion trends, consumers' emotional satisfaction and the value of digital wear.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.10C
• /
• pp.964-972
• /
• 2006

The kidney keeps with close relationship in the internal organs, that the kidney function filtering eliminate the wastes to the urine on the processing to replace the old with the new blood. In case of these problem in the kidney, there is no way to catch out with self-awakening symptom except for serious illness. This problem can solve with keeping the systematic diagnosis method in the kidney trouble shooting. Under the circumstances, the importance of the diagnosis for the kidney disease is growing day after day. In this paper, among the great four diagnosises, using the way of ocular inspection & auscultation, we would like to propose rouble shooting in the way of the kidney. To do this, through the assistance of the input image, extract the value of the color with appropriate output, analysing the color of the face with related to the kidney, using the results we would like to get the accurate symptoms on the kidney's problems. Also, through analysing and comparing with the relationship the kidney and the signal of voice, we would like to realize the proof system of human health. Finally, we'd like to make proof of the usefulness for proposed method from this study.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.22 no.4
• /
• pp.423-429
• /
• 2011

Cooperative spectrum sensing is proposed to overcome some problem such as multipath fading and shadowing and to improve spectrum sensing performance. There are different combining methods for cooperative spectrum sensing: hard decision method and soft decision method. In this paper, we analysis the performance of cooperative spectrum sensing with distance based weight that is kind of a soft decision rule for cognitive radio(CR) systems and CR systems sense the spectrum of the licensed user by using a energy detection method. Threshold is determined in accordance with the constant false alarm rate(CFAR) algorithm for energy detection. The signal of licensed user is OFDM signal and the wireless channel between a licensed user and CR systems is modeled as Gaussian channel. From the simulation results, the cooperative spectrum sensing with distance based weight combining(DWC) and equal gain combing(EGC) methods shows higher spectrum sensing performance than single spectrum sensing does. And the detection probability performance with the DWC is higher than that with the EGC.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.40 no.9
• /
• pp.373-383
• /
• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.44 no.1
• /
• pp.24-31
• /
• 2007

DVB-T (Digital Video Broadcasting-Terrestrial) is an important multimedia broadcasting technology capable of high data-rate transmission and adopted by Europe. OFDM (Orthogonal Frequency Division Multiplexing) is the backbone technique employed in DVB-T to support multimedia services that have various bandwidths. Unfortunately, an OFDM signal has a large PAPR (Peak-to-Average Power Ratio). In this paper, we investigate the performance of a simple PAPR reduction scheme for the DVB-T system, which requires no change of a receiver structure or no additional information transmission. The approach we employed is clipping in the time and frequency domains. The time-domain clipping is carried out with a predetermined clipping level while the frequency-domain clipping is done within EVM (Error Vector Magnitude). This approach is suboptimal with lower computational complexity compared to the optimal method. The simulation results demonstrate that the proposed one is getting more effective at lower modulation levels and with more allowed constellation error.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.6A
• /
• pp.593-602
• /
• 2006

This paper propose to methodology for deciding suitable bit size that minimizes hardware complexity and performance degradation from floating-point design the fixed-point implementation of downlink traffic channel of IEEE 802.16e OFDMA TDD system. One of the major considering issues for implementing fixed-point design is to select Saturation or Quantization properly with the knowledge of signal distribution by pdf or histogram. Also, through trial and error, we should execute exhaustive computer simulation for various bit sizes, hence obtain appropriate bit size while minimizing performance degradation. We carry out computer simulation to decide the optimized bit size of downlink traffic channel under AWGN and ITU-R M.1225 Veh-A channel model.

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.48 no.2
• /
• pp.55-60
• /
• 2011

Most of neural network chips use an analog-type maximum selector circuit (MS). As the increase of integration level, the analog MS has difficulties in achieving sufficient resolution. Contrary, the digital-type MS is easy to get high resolution but slower than its analog counterparts. A new high-speed digital MS circuit called MSIT (Maximum Selector with Internal Trigger-signal) is presented in this paper. The MSIT has been designed to achieves both the high reliability by using trigger-signals and high speed by removing the unnecessary waiting times. The response time of MSIT is 3.4ns for 32 data with 10-bit resolution in the simulation with 1.2V, $0.13{\mu}m$-process model parameters, which is much faster than its analog counterparts. It shows that digital MS circuits like MSIT can achieve higher speed as well as higher resolution than analog MS circuits.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.1
• /
• pp.56-61
• /
• 2001

This paper presents the real-time implementation of an AMR (Adaptive Multi Rate) vocoder which is included in the asynchronous International Mobile Telecommunication (IMT)-2000 mobile ASIC. The implemented AMR vocoder is a multi-rate coder with 8 modes operating at bit rates from 12.2kbps down to 4.75kbps. Not only the encoder and the decoder as basic functions of the vocoder are implemented, but VAD (Voice Activity Detection), SCR (Source Controlled Rate) operation and frame structuring blocks for the system interface are also implemented in this vocoder. The DSP for AMR vocoder implementation is a 16bit fixed-point DSP which is based on the TeakLite core and consists of memory block, serial interface block, register files for the parallel interface with CPU, and interrupt control logic. Through the implementation, we reduce the maximum operating complexity to 24MIPS by efficiently managing the memory structure. The AMR vocoder is verified throughout all the test vectors provided by 3GPP, and stable operation in the real-time testing board is also proved.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.12C
• /
• pp.1617-1622
• /
• 2004

MPEG/Audio decoder for mobile multimedia systems requires low power consumption. Implementations of AV decoder using a single RISC processor often need high power consumption owing to cash-miss in case of insufficient cash memory. In this paper, we present a MPEG/Audio decoder for mobile handset applications and implement it on a RISC processor embedding a minimized DSP accelerator. Audio decoding algorithm is splined into two parts; computation intensive and control intensive parts. Those parts we, respectively, allocated to DSP and RISC core, which are designed to run in parallel to increase the processing efficiency. The proposed system implements MP3 and AAC decoders at l7MHz and 24MHz clocks, which are reductions of 48% and 40% of complexities in comparison with implementations on a single RISC processor. The proposed method is adequate for mobile multimedia applications with insufficient cash memory.