• 한국멀티미디어학회논문지
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• 제24권8호
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• pp.1012-1019
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• 2021

Currently, since the black-white QR (Quick Response) codes have limited storage capacity, color QR codes have been actively being studied. By multiplexing 3 colors, the color QR codes can allow the code capacity to be increased by three times, however, the color multiplexing brings about the possibility of crosstalk and noises in the acquisition process of the final image, incurring the decrease of bit-recognition rate. In order to improve the bit recognition rate, while keeping the storage capacity high, this paper proposes a new type of color QR code which uses the pattern information as well as the color information, and then analyzes how to increase the bit recognition rate. For this aim, the paper presents an efficient system which extracts embedded information from color QR code and then, through practical experiments, it is shown that the proposed color QR codes improves the bit recognition rate and are useful for commercial applications, compared to the conventional color codes.

• 한국시뮬레이션학회논문지
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• 제18권2호
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• pp.9-17
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• 2009

얼굴인식이 널리 사용되기 시작하면서, 얼굴 데이터베이스는 많은 양의 얼굴정보를 담게 되었다. 이러한 얼굴 데이터의 증가로 인하여 분산처리 방법을 이용한 얼굴인식이 주요 주제로 대두되고 있다. 하지만 기존 방법에서는 대용량의 데이터를 전송하는 방법에 대한 논의가 부족하다. 이에 본 논문은 분산처리 환경에서 퍼지로직 기반 비트압축률 선택을 통한 얼굴인식을 제안한다. 제안한 방법은 얼굴인식률, 얼굴인식 수행시간, 전송된 비트 길이를 바탕으로 퍼지추론을 하여 효과적인 압축률을 선택한다. 우리는 제안한 방법과 압축을 하지 않은 데이터, 고정 압축률을 적용한 데이터에 따른 얼굴인식률과 얼굴인식 수행시간을 측정하여 비교하였다. 실험 결과는 퍼지로직 기반 압축률 선택이 수행시간을 감소시키면서도 합리적인 인식률을 유지하는 효과가 있음을 보여준다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2007년도 하계종합학술대회 논문집
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• pp.423-424
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• 2007

In this paper, we propose a scalable high-quality speech reconstruction method for distributed speech recognition (DSR). It is difficult to reconstruct speech of high quality with MFCCs at the DSR server. Depending on the bit-rate available by the DSR system, we can send additional information associated with speech coding to the DSR sorrel, where the bit-rate is variable from 4.8 kbit/s to 11.4 kbit/s. The experimental results show that the speech quality reproduced by the proposed method when the bit-rate is 11.4 kbit/s is comparable with that of ITU-T G.729 under both ideal channel and frame error channel conditions while the performance of DSR is maintained to that of wireline speech recognition.

• 전자공학회논문지B
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• 제29B권2호
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• pp.1-8
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• 1992

This paper is proposed a Modified WLD (Weighted Levenshtein Distance) algorithm for processor desihn of code-string matching. A proposed MWLD (Modified Weighted Levenshtein Distance) algorithm is consist of 1-dimension bit-serial array processor to pattern matching using a Hamming Distance. The proposed processor is applied to recognition of character with real time input. The recognition rate of Hangul strokes is resulted to 98.65$\%$

• 한국통신학회논문지
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• 제25권3B호
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• pp.529-540
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• 2000

본 논문에서는 비트율이 매우 한정된 응용에 적용되는 초저속 동영상 부호화 표준인 H.236plus에 기반한 의미기반 율-제어 기법을 제안한다. 기존의 율-제어 기법들은 의미기반이 아닌 단순히 프레임에 대해 매크로 블록 단위로 최적의 양자화 파라미터를 설정해 정해진 비트율을 조정하나, 본 논문에서는 부호화 효율을 높이기 위해 의미 영역인식 및 우선권 부여를 통해 영역별로 최적의 양자화 파라미터를 설정함으로써 제약된 비트 내에서 주관적인 화질의 향상을 보이는 기법을 제안한다. 제안하고자하는 기법의 핵심은 부호화 시 배경에 해당하는 영역에 배정되는 비트량을 할당해 전경(얼굴) 영역에 비트 자원을 더 많이 배정함으로써 주관적인 화질을 보호, 개선하기 위한 것이다.

• Journal of Semiconductor Engineering
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• 제1권1호
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• pp.31-37
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• 2020

We present an approximate DRAM architecture for energy-efficient deep learning. Our key premise is that by bounding memory errors to non-critical information, we can significantly reduce DRAM refresh energy without compromising recognition accuracy of deep neural networks. To validate the key premise, we make extensive Monte-Carlo simulations for several well-known convolutional neural networks such as LeNet, ConvNet and AlexNet with the input of MINIST, CIFAR-10, and ImageNet, respectively. We assume that the highest-order 8-bits (in single precision) and 4-bits (in half precision) are protected from retention errors under the proposed architecture and then, randomly inject bit-errors to unprotected bits with various bit-error-rates. Here, recognition accuracies of the above convolutional neural networks are successfully maintained up to the 10^-5-order bit-error-rate. We simulate DRAM energy during inference of the above convolutional neural networks, where the proposed architecture shows the possibility of considerable energy saving up to 10 ~ 37.5% of total DRAM energy.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.869-870
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• 2008

Illumination variation is the most significant factor affecting face recognition rate. In this paper, we propose adaptive smoothing based on combined most significant bit (CMSB) - plane and local entropy for robust face recognition in varying illumination. Illumination normalization is achieved based on Retinex method. The proposed method has been evaluated based on the CMU PIE database by using Principle Component Analysis (PCA).

• Journal of information and communication convergence engineering
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• 제2권1호
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• pp.36-39
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• 2004

In this paper, a method of improving the learning speed and convergence rate is proposed to exploit the advantages of artificial neural networks and neuro-fuzzy systems. This method is applied to the XOR problem, n bit parity problem, which is used as the benchmark in the field of pattern recognition. The method is also applied to the recognition of digital image for practical image application. As a result of experiment, it does not always guarantee convergence. However, the network showed considerable improvement in learning time and has a high convergence rate. The proposed network can be extended to any number of layers. When we consider only the case of the single layer, the networks had the capability of high speed during the learning process and rapid processing on huge images.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제4권2호
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• pp.149-154
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• 2004

In this research, we selected the speech recognition to implement the electric wheelchair system as a method to control it by only using the speech and used DTW (Dynamic Time Warping), which is speaker-dependent and has a relatively high recognition rate among the speech recognitions. However, it has to have small memory and fast process speed performance under consideration of real-time. Thus, we introduced VQ (Vector Quantization) which is widely used as a compression algorithm of speaker-independent recognition, to secure fast recognition and small memory. However, we found that the recognition rate decreased after using VQ. To improve the recognition rate, we applied ART2 (Adaptive Reason Theory 2) algorithm as a post-process algorithm to obtain about 5% recognition rate improvement. To utilize ART2, we have to apply an error range. In case that the subtraction of the first distance from the second distance for each distance obtained to apply DTW is 20 or more, the error range is applied. Likewise, ART2 was applied and we could obtain fast process and high recognition rate. Moreover, since this system is a moving object, the system should be implemented as an embedded one. Thus, we selected TMS320C32 chip, which can process significantly many calculations relatively fast, to implement the embedded system. Considering that the memory is speech, we used 128kbyte-RAM and 64kbyte ROM to save large amount of data. In case of speech input, we used 16-bit stereo audio codec, securing relatively accurate data through high resolution capacity.

• 대한음성학회지:말소리
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• 제54호
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• pp.27-43
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• 2005

Existing standard speech coders can provide speech communication of high quality while they degrade the performance of speech recognition systems that use the reconstructed speech by the coders. The main cause of the degradation is that the spectral envelope parameters in speech coding are optimized to speech quality rather than to the performance of speech recognition. For example, mel-frequency cepstral coefficient (MFCC) is generally known to provide better speech recognition performance than linear prediction coefficient (LPC) that is a typical parameter set in speech coding. In this paper, we propose a speech coder using MFCC instead of LPC to improve the performance of a server-based speech recognition system in network environments. However, the main drawback of using MFCC is to develop the efficient MFCC quantization with a low-bit rate. First, we explore the interframe correlation of MFCCs, which results in the predictive quantization of MFCC. Second, a safety-net scheme is proposed to make the MFCC-based speech coder robust to channel error. As a result, we propose a 8.7 kbps MFCC-based CELP coder. It is shown from a PESQ test that the proposed speech coder has a comparable speech quality to 8 kbps G.729 while it is shown that the performance of speech recognition using the proposed speech coder is better than that using G.729.