• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.947-950
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• 2001

This paper implements H.263 codec using SIMD(single instruction multiple data) method in real time based on PC. This system uses INS algorithm previously proposed by the authors as motion estimation module. SIMD method is used in DCT, IDCT, quantization, motion estimation, and display module. The developed algorithms are implemented using TMN5. Using the above algorithm, H.263 Codec can communicate more than 15 frames/sec in CIF resolution on a Pentium-IV 1.7GHz computer.

• Proceedings of the Korean Information Science Society Conference
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• 2010.06b
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• pp.497-502
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• 2010

조합형 실시간 스케줄링은 계층적으로 구성된 실시간 시스템에 대해 실시간성을 보장할 수 있는 방법을 제공한다. 조합형 스케줄링 이론을 통해 여러 개의 실시간 태스크를 하나의 실시간 태스크로 묶어 스케줄링 할 수 있으며, 실시간 보장을 위해 필요한 CPU 요구량을 계산하였다. 하지만, 양자화에 대한 고려가 없어, 틱-기반 스케줄링 시스템에서 정확한 CPU 요구량을 계산할 수 없다. 따라서, 본 연구에서는 양자화에 따른 CPU 할당량의 영향을 정량적으로 보여준다.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.257-258
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• 2006

In this paper, we address several issues in the real time implementation of MPEG-1 Layer II decoder on a fixed-point digital signal processor (DSP), especially TMS320C6416. There is a trade-off between processing speed and the size of program/data memory for the optimal implementation. In a view of the speed optimization, we first convert the floating point operations into fixed point ones with little degradation in audio quality, and then the look-up tables used for the inverse quantization of the audio codec are forced to be located into the internal memory of the DSP. And then, window functions and filter coefficients in the decoder are precalculated and stored as constant, which makes the decoder faster even larger memory size is required. It is shown from the real-time experiments that the fixed-point implementation enables us to make the decoder with a sampling rate of 48 kHz operate with 3 times faster than real-time on TMS320C6416 at a clock rate of 600 MHz.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.48 no.3
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• pp.21-29
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• 2011

This paper presents a method that estimates distance and velocity of an object with reliability regardless of maneuver status of the target in stereo vision system. A stereo vision system can calculate a distance with disparity from left and right images. However, the distance estimation error may occur due to quantization error of image pixel. A sub-pixel interpolation method minimizes the quantization error and estimates accurate disparity with real value. Extended Kalman filter (EKF) was used to minimize the error covariance and estimate the object's velocity. However, divergence problem occurs due to model uncertainty when a target maneuvers highly, which makes the estimation error increase. In this paper, we propose a virtual model extended Kalman filter (VMEKF) method that minimizes the processing time and provides reliable estimation ability regardless of maneuver status. Computer simulations and experimental results in real road environments demonstrate that the proposed method gives a reliable estimation performance and reduces processing time under various maneuver status while comparing other estimation filters.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.9
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• pp.1330-1339
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• 2022

3D object detection generally aims to detect relatively large data such as automobiles, buses, persons, furniture, etc, so it is vulnerable to small object detection. In addition, in an environment with limited resources such as embedded devices, it is difficult to apply the model because of the huge amount of computation. In this paper, the accuracy of small object detection was improved by focusing on local features using only one layer, and the inference speed was improved through the proposed knowledge distillation method from large pre-trained network to small network and adaptive quantization method according to the parameter size. The proposed model was evaluated using SUN RGB-D Val and self-made apple tree data set. Finally, it achieved the accuracy performance of 62.04% at mAP@0.25 and 47.1% at mAP@0.5, and the inference speed was 120.5 scenes per sec, showing a fast real-time processing speed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.2
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• pp.327-334
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• 2016

In this paper, we propose a hardware architecture of transform and quantization for high-perfornamce HEVC(High Efficiency VIdeo Coding) encoder. HEVC transform decides the transform mode by comparing RDCost to search for the best mode of them. But, RDCost is computed using the bit-rate and distortion which is computed by transform, quantization, de-quantization, and inverse transform. Due to the many calculations and encoding time, it is hard to process high resolution and high definition image in real-time. This paper proposes the method of transform mode decision by comparing sum of coefficient after transform only. We use BD-PSNR and BD-Bitrate which is performance indicator. Based on the experimental result, We confirmed that the decision of transform mode can process images with no significant change in the image quality. We reduced hardware area by assigning different values at the same output according to the transform mode and overlapping coefficient multiplied as much as possible. Also, we raise performance by implementing sequential pipeline operation. In view of the larger process that we used compared with the process of reference paper, Our design has reduced by half the hardware area and has increased performance 2.3 times.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.2
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• pp.28-34
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• 1994

In this paper, a new time-domain parameter extraction method for speech recognition is proposed. The suggested emthod is based on the fact that the local peak-to-peak interval, i.e., the interval between maxima and minima of speech waveform is closely related to the frequency component of the speech signal. The parameterization is achieved by a sort of filter bank technique in the time domain. To test the proposed parameter extraction emthod, an isolated word recognizer based on Vector Quantization and Hidden Markov Model was constructed. As a test material, 22 words spoken by ten males were used and the recognition rate of 92.9% was obtained. This result leads to the conclusion that the new parameter extraction method can be used for speech recognition system. Since the proposed method is processed in the time domain, the real-time parameter extraction can be implemented in the class of personal computer equipped onlu with an A/D converter without any DSP board.

• The Journal of the Acoustical Society of Korea
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• v.13 no.1E
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• pp.89-95
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• 1994

This paper describes development of a real-time voice dialing system which can recognize around one hundred word vocabularies in speaker independent mode. The voice recognition algorithm in this system is implemented on a DSP board with a telephone interface plugged in an IBM PC AT/486. In the DSP board, procedures for feature extraction, vector quantization(VQ), and end-point detection are performed simultaneously in every 10 msec frame interval to satisfy real-time constraints after detecting the word starting point. In addition, we optimize the VQ codebook size and the end-point detection procedure to reduce recognition time and memory requirement. The demonstration system has been displayed in MOBILAB of the Korean Mobile Telecom at the Taejon EXPO'93.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.6
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• pp.149-158
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• 2005

This paper proposed a blind watermarking method for video contents which satisfies both the invisibility and the robustness to attacks to prohibit counterfeiting, modification, illegal usage and illegal re-production of video contents. This watermarking algorithm targets MPEG compression system and was designed to control the amount of watermarking to be inserted according to the adaptive quantization scale code to follow the adaptive quantization of the compression system. The inserting positions of the watermark were chosen by considering the frequency property of an image and horizontal, vertical and diagonal property of a $8{\times}8$ image block. Also the amount of watermarking for each watermark bit was decided by considering the quantization step. This algorithm was implemented by C++ and experimented for invisibility and robustness with MPEG-2 system. The experiment results showed that the method satisfied enough the invisibility of the inserted watermark and robustness against attacks. For the general attacks, the error rate of the extracted watermark was less than $10\%$, which is enough in robustness against the attacks. Therefore, this algorithm is expected to be used effectively as a part in many MPEG systems for real-time watermarking, especially in the sensitive applications to the network environments.