• Journal of Institute of Control, Robotics and Systems
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• v.12 no.9
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• pp.912-917
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• 2006

A strain gauge weight measuring instrumentation system was designed with RF sensor network facilities. In the sensor module system data conversion and a series of signal processing were totally equipped. 16 strain gauges are incoming sensors and each output of the strain gauge was amplified and filtered for proper analog signal processing. Several measuring instrumentation OP amps and general purposed OP amps were used. 12 bits A/D converters converted analog signals to digital bits and a PIC microprocessor controlled the 16 channels of strain gauges. RF RS232 modules were used for wireless communication between the PIC microprocessor and an Ethernet host far a remote sensor monitoring system development.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.7
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• pp.628-637
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• 1990

In this paper we present on architecture for a high sppeed CMOS multiplier which can be used for real-time digital signal processing. And a synthesis method for designing highly parallel algorithms in VLSI is presented. A parallel multiplier design based on the modified Booth's algorithms and Ling's algorthm. This paper addresses the design of multiplier capable of accpting data in 2's complement notation and coefficients in 2's complement notation. Multiplier consists of an interative array of sequential cells, and are well suited to VLSI implementation as a results of their modularity and regularity. Booth's decoders can be fully tested using a relatively small number af test vector.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.5
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• pp.1041-1046
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• 2009

In this paper, we proposed a robust watermark scheme for image based on SVD(Singular Value Transform) and Triplet. First, the original image is decomposed by using 3-level DWT, and then used the singular values changed for embedding and extracting of the watermark sequence in LL3 band. Since the matrix of singular values is not easily altered with various signal processing noises, the embedded watermark sequence has the ability to withstand various signal processing noise attacks. Nevertheless, this method does not guarantee geometric transformation(such as rotation, cropping, etc.) because the geometric transformation changes the matrix size. In this case, the watermark sequence cannot be extracted. To compensate for the above weaknesses, a method which uses the triplet for embedding a barcode image watermark in the middle of frequency band is proposed. In order to generate the barcode image watermark, the pattern of the watermark sequence embedded in a LL3 band is used. According to this method, the watermark information can be extracted from attacked images.

• Journal of Information Processing Systems
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• v.15 no.1
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• pp.169-180
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• 2019

Transmitting visual information over a broadcasting network is not only prone to a copyright violation but also is a forgery. Authenticating such information and protecting its authorship rights call for more advanced data encoding. To this end, electronic watermarking is often adopted to embed inscriptive signature in imaging data. Most existing watermarking methods while focusing on robustness against degradation remain lacking of measurement against security loophole in which the encrypting scheme once discovered may be recreated by an unauthorized party. This could reveal the underlying signature which may potentially be replaced or forged. This paper therefore proposes a novel digital watermarking scheme in temporal-frequency domain. Unlike other typical wavelet based watermarking, the proposed scheme employed the Lorenz chaotic map to specify embedding positions. Effectively making this is not only a formidable method to decrypt but also a stronger will against deterministic attacks. Simulation report herein highlights its strength to withstand spatial and frequent adulterations, e.g., lossy compression, filtering, zooming and noise.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38C no.7
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• pp.623-629
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• 2013

This paper presents the implementation result of the EEG(electroencephalogram) signal transmission protocol and its test platform. EEG measured by a dry-type electrode is directly converted into digital signal by ADC(analog-to-digital converter). Thereafter it is transferred DSP(digital signal processor) platform by $I^2C$(inter-integrated circuit) protocol. DSP conducts the pre-processing of EEG and extracts feature vectors of EEG. In this work, we implement the $I^2C$ protocol with 16 channels by using 10 or 12-bit ADC. In the implementation results, the overhead ratio for the 4 bytes data burst transmission measures 2.16 and the total data rates are 345.6 kbps and 414.72 kbps with 10-bit and 12-bit 1 ksps ADC, respectively. Therefore, in order to support a high speed mode of $I^2C$ for 400 kbps, it is required to use 16:1 and $(8:1){\times}2$ ratios for slave:master in 10-bit ADC and 12-bit ADC, respectively.

• Korean Journal of Optics and Photonics
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• v.18 no.5
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• pp.309-316
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• 2007

We demonstrate the wavelength swept source and signal processing for the frequency domain optical coherence tomography. The laser output performance is improved by using a semiconductor optical amplifier with a booster amplifier. The laser generates 14 mW of average power of which wavelength shift in the lasing spectral shape is compensated. Adopting a Fabry-Perot etalon and digital signal processing, the broadening of the beat frequency due to the variance of wavelength-sweep-velocity is calibrated. The optical coherence tomography system shows 154.4 kHz of axial scanning speed, 0.95mm of depth range, and $12{\pm}0.37{\mu}m$ of axial resolution.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.8C
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• pp.806-815
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• 2006

It is studied to design a low pass filter of the SRC(sample rate converter), which is used to change the sampling rate of digital signals such as in digital modulation and demodulation systems. The larger the conversion ratio of the sample rate becomes, the more signal processing is needed for the filter, which corresponds to the more complexity in circuit realization. Thus it is important to reduce the amount of signal processing for the case of high conversion ratio. In this paper it is presented a design method of a two-stage cascaded FIR filter, which proved to have reduced amount of signal processing in comparison with a conventional single-stage one. The reduction effect of signal processing turned out to be more noticeable for larger value of conversion ratio, for instance, giving down to 72% in complexity for the conversion ratio of 32. It has been shown that the reduction effect is dependent to specific combination of conversion ratios of the cascaded filters. So an exhaustive search has been performed in order to obtain the optimal combination for various values of the total conversion ratio. In this paper every filter is considered to be implemented in the form of a polyphase FIR filter, and its coefficients are determined by use of the Parks-McCllelan algorithm.

• Journal of Sensor Science and Technology
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• v.15 no.4
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• pp.223-230
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• 2006

An FIR filter for digital voice signal processing has been designed and implemented using a microcontroller in implantable middle ear hearing devices (IMEHDs). The designed digital voice signal processing filter which has fast and accurate filtering operation and controllable filter characteristics has been implemented using a hardware multiplier and a direct memory access (DMA) in the low power microcontroller, MSP430F169. It has been confirmed that each of the implemented 6-orders Remez FIR filters with 1 channel and 2 channels can be applied to the voice signal processing module of IMEHDs based on the evaluation results of the filtering performance experiment.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.5
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• pp.26-33
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• 1994

Phase retrieval is concerned with the reconstruction of a signal from its Fourier transform magnitude (or intensity), which arises in many areas such as X-ray crystallography, optics, astronomy, or digital signal processing. In such areas, the Fourier transform phase of the desired signal is lost while measuring Fourier transform magnitude (F.T.M.). However, if a reference 'signal is added to the desired signal, then, in the Fourier trans form magnitude of the added signal, the Fourier transform phase of the desired signal is encoded. This paper addresses uniqueness and retrieval of the encoded Fourier phase of a multidimensional signal from the Fourier transform magnitude of the added signal along with the Fourier transform magnitude of the desired signal and the information of the additive reference signal. In Part I, several conditions under which the desired signal can be uniquely specified from the two Fourier transform magnitudes and the additive reference signal are presented. In Part II, the development of non-iterative algorithms and an iterative algorithm that may be used to reconstruct the desired signal(s) is considered.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.5
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• pp.34-41
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• 1994

Phase retrieval is concerned with the reconstruction of a signal from its Fourier transform magnitude (or intensity), which arises in many areas such as X-ray crystallography, optics, astronomy, or digital signal processing In such areas, the Fourier transform phase of the desired signal is lost while measuring Fourier transform magnitude (F.T.M.). However, if a reference 'signal is added to the desired signal, then, in the Fourier trans form magnitude of the added signal, the Fourier transform phase of the desired signal is encoded This paper addresses uniqueness and retrieval of the encoded Fourier phase of a multidimensional signal from the Fourier transform magnitude of the added signal along with Fourier transform magnitude of the desired signal and the information of the additive reference signal In Part I, several conditions under which the desired signal can be uniquely specified from the two Fourier transform magnitudes and the additive reference signal are presented In Part II, the development of non-iterative algorithms and an iterative algorithm that may be used to reconstruct the desired signal (s) is considered