• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.107-112
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• 2006

The previous acquisition method of GPS receiver for reference station adopts not only the coherent integration method but also the non-coherent integration method in order to enhance sensitivity under noisy environment. However, under noisy environment, the previous GPS signal acquisition method causes the non-coherent integration loss which is a major factor among losses that can be caused during GPS signal acquisition. The non-coherent integration loss also increases with the strength of the received noise. This paper has intention of analyzing the performance of the GPS signal acquisition method proposed to effectively enhance sensitivity of DGPS reference receiver under noisy environment. This paper presents that the proposed GPS signal acquisition method suppresses the non-coherent integration loss through post-processing simulation. Furthermore, with regard to the mean acquisition time, it is shown that the number of search cells of the proposed GPS signal acquisition method is much fewer than that of the previous GPS signal acquisition method.

• Journal of the Institute of Convergence Signal Processing
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• v.20 no.1
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• pp.45-50
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• 2019

When PPG signal is used in mobile healthcare devices, the accuracy of the measured heartbeat decreases from the influence by the movement of the user. The reason is that the frequency band of the noise overlaps the frequency band of the PPG signal. In order to remove these same noises, the methods using frequency analysis method or application of acceleration sensor have been investigated and showed excellent performance. However, in applying these methods to low-cost healthcare devices, it is difficult to apply these methods because of much processing time and sensor's cost. In order to solve these problems, this study proposed the filter design method using an adaptive notch filter and the fuzzy inference system to extract more accurate heart rate in real time and evaluated its performance. As results, it showed better results than the other methods. Based on the results, when applying the proposed method to design the mobile healthcare device, it is possible to measure the heartbeat more accurately in real time.

• Proceedings of the Acoustical Society of Korea Conference
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• 1984.12a
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• pp.91-95
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• 1984

Because of layer and scattering in the ocean, there are some problem in algorithm currently used for the recognition of targets. Those are time delay of processing and circuit design. The simple method of detecting direct sound wave in noise caused by time delay is proposed-recognized, estimated, and then direcxt sound wave is reconstructed by the AMDF and $\mu$-processor. 2KHz, 4KHz, 8KHz, 12KHz, 16KHz sound waves are used in experiment. To obtain a reference signal, anechoic water tank is used is processing and aluminium water tank used instead of real ocean. As a result, there are a few errors which caused by anechoic water tank error, decreasing of frequency make errors. Possibility of application to Sonar Signal Processing is proved.

• ETRI Journal
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• v.34 no.2
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• pp.256-259
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• 2012

The double-base number system has been used in digital signal processing systems for over a decade because of its fast inner product operation and low hardware complexity. This letter proposes an innovative multiplier architecture using hybrid operands. The multiplier can easily be linked to flash analog-to-digital converters or digital systems through a double-base number encoder (DBNE) for realtime signal processing. The design of the DBNE and the multiplier enable faster digital signal processing and require less hardware resources compared to the binary processing method.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.383-384
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• 2006

The P-ECLMS algorithm adapted the existing Post-Processing method occurs the distortion of the near-end signal at the double-talk situationt. To solve this problem, we propose the SP-ECLMS(Selective Post-Processing ECLMS) algorithm which makes the Post-Processing coefficient differently at the case of the single-talk and the double-talk. When the correlation level is not output less than 30%, the proposed algorithm output the original signal to prevent the signal's distortion.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.50 no.12
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• pp.590-596
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• 2001

The pre-emphasis is an essential process for speech signal processing. Widely used two methods are the typical method using a fixed value near unity and te optimal method using the autocorrelation ratio of the signal. This study proposes a new pre-emphasis method using the short-term energy difference of speech signal, which can effectively compensate the glottal source characteristics and lip radiation characteristics. Using the proposed pre-emphasis, speech analysis, such as spectrum estimation, formant detection, is performed and the results are compared with those of the conventional two pre-emphasis methods. The speech analysis with 5 single vowels showed that the proposed method enhanced the spectral shapes and gave nearly constant formant frequencies and could escape the overlapping of adjacent two formants. comparison with FFT spectra had verified the above results and showed the accuracy of the proposed method. The computational complexity of the proposed method reduced to about 50% of the optimal method.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.3
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• pp.169-175
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• 2011

In this paper, an automatic scoring system of the OMR answer sheet is implemented using Gray Scale and image segmentation method. The proposed method was used to extract the OMR data on multiple-choice answer sheet from captured image. In addition, On-line scoring system is developed and implemented to mark the short-answer type on the reverse side. Therefore, teachers can mark the short-answer type for anytime and anywhere within the available time. There were many advantages to mark of the multiple-choice answer sheet without additional OMR reader. In the future, the grading of short-answer type will be more efficient if it were performed by using an automatic scoring system based on image processing.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.3
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• pp.15-20
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• 2002

In this paper, the method of processing a blurred noisy signal has been researched. The conventional method of processing signal has faults, which are slow-convergence speed and long time-consuming process at the singular point and/or in the ill condition. There is the process, the Gauss-Seidel's method to remove these faults, but it takes too much time because it processes signal repeatedly. For overcoming the faults, this paper shows a signal process method which takes shorter than the Gauss-Seidel's by comparing the Gauss-Seidel's with proposed algorithm and accelerating convergence speed at the singular point and/or in the ill condition. 

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.169-174
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• 2003

A novel subspace speech enhancement using subband whitening filter is proposed. Previous subspace speech enhancement method either assumes additive white noise or uses whitening filter as a pre-processing for colored noise. The proposed method tries to minimize the signal distortion while reducing residual noise by processing the signal using subband whitening filter. By incorporating the notion of subband whitening filter, spectral resolution in Karhunen-Loeve(KL) domain is improved with the negligible additional computational load. The proposed method outperforms both the subspace method suggested by Ephraim and the spectral subtraction suggested by Boll in terms of segmental signal-to-noise ratio (SNRseg) and perceptual evaluation of speech quality (PESQ).

• Journal of Positioning, Navigation, and Timing
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• v.12 no.1
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• pp.75-81
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• 2023

In this study, we design an optimized Graphics Processing Unit (GPU)-based GNSS signal processing technique with the goal of designing and implementing a GNSS Software Defined Receiver (SDR) that can operate in real time all-in-view mode under multi-constellation and multi-frequency signal environment. In the proposed structure the correlators of the existing GNSS SDR are processed by the GPU. We designed a memory structure and processing method that can minimize memory access bottlenecks and optimize the GPU memory resource distribution. The designed GNSS SDR can select and operate only the desired GNSS or desired satellite signals by user input. Also, parameters such as the number of quantization bits, sampling rate, and number of signal tracking arms can be selected. The computing capability of the designed GPU-based GNSS SDR was evaluated and it was confirmed that up to 2400 channels can be processed in real time. As a result, the GPU-based GNSS SDR has sufficient performance to operate in real-time all-in-view mode. In future studies, it will be used for more diverse GNSS signal processing and will be applied to multipath effect analysis using more tracking arms.