• Proceedings of the KIEE Conference
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• 1987.07b
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• pp.1449-1453
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• 1987

A microprogrammable Bit Slice Sinal Processor for image processing is implemented. Processing speed is increased by the parallelism in horizontal microprogram using 120bits microcode, pipelined architecture, 2 bank memory switching that interfaces with the Host through DMA, a variable clock control, overflow checking H/W,look-up table method and cache memory. With this processor, a DCT algorithm which uses 2-D FFT is performed. The execution time for $512{\times}512{\times}8$ image is 12 sec when 16 bit operation is runned, and the recovered image has acceptable quality with MSE 0.276%.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.930-933
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• 2011

This paper addresses an adaptive postprocessing method applied in the spatial domain for block-based discrete cosine transform (BDCT) coded images. The proposed algorithm is designed by a serial concatenation of a 1D simple smoothing filter and a 2D directional filter. The 1D smoothing filter is applied according to the block type, which is determined by an adaptive threshold. It depends on local statistical properties, and updates block types appropriately by a simple rule, which affects the performance of deblocking processes. In addition, the 2D directional filter is introduced to suppress the ringing effects at the sharp edges and the block discontinuities while preserving true edges and textural information. Comprehensive experiments indicate that the proposed algorithm outperforms many deblocking methods in the literature, in terms of PSNR and subjective visual quality evaluated by GBIM.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.5 s.311
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• pp.48-56
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• 2006

In this paper, we proposed the high quality audio watermarking algorithm using MDCT/IMDCT (Modified DCT/Inverse Modified DCT) with psychoacoustic model. Generally, a digital audio watermark is embedding the frequency domain after frequency transform of the digital audio data but the digital audio quality is affected by watermarking. In our scheme, the digital audio data is spread with PN((Pseudo Noise) code and then audio watermark is embedded in MDCT processing that refers psychoacoustic model. In MDCT processing, according to the shape of filter bank output, the block switching selects a window sequence that has 256, 1,024 or 2,048 points interval for high quality audio. The author confirm that when watermark weight ${\alpha}$ is 2.5 below, the detection ratio of watermark is a satisfied to SDMI's(Secure Digital Music Initiative) recommendation 50% above and SM is $50{\sim}68dB$ range with mainly 4 kind of attacks(Compression, Cropping, FFT and Echo).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.1
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• pp.33-46
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• 1996

Inthis paper, we investigate the compatible coding technique, which receives much interest ever since the introduction of HDTV. First, attempts have been made to analyze the theoretical transform coding gains for various hierarchical decomposition techniques, namely subband, pyramid and DCT-based decomposition techniques. It is shown that the spatical domain techniques proide higher transform coding gains than the DCT-based coding technique. Secondly, we compare the performance of these spatial domain techniques, in terms of the PSNR versus various rate allocations to each layer. Based on these analyses, it is believed that the pyramid decomposition is more appropriate for the compatible coding. Also in this paper, we propose a hybrid prediction pyramid coding technique, by combining the spatio-temporal prediction in MPEG-2[3] and the adaptive MC(Motion Compensation)[1]. In the proposed coding technigue, we also employ an adaptive DCT coefficient scanning technique to exploit the direction information of the 2nd-layer signal. Through computer simulations, the proposed hybrid prediction with adaptive scanning technuque shows the PSNR improvement, by about 0.46-1.78dB at low 1st-layer rate(about 0.1bpp) over the adaptive MC[1], and by about 0.33-0.63dB at high 1st-layer rate (about 0.32-0.43bpp) over the spatio-temporal prediction[3].

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.4
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• pp.442-454
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• 2002

It's noticed that the SA-DCT (Shape-Adaptive Discrete Cosine Transform) produces different 2-dimensional transform coefficients according as the first 1-dimensional transform is applied in horizontal or vertical direction for an arbitrarily shaped boundary block. Performing the first 1-dimensional transform in the direction, which has higher spatial correlation or smaller shifting distance, can compact the energy on the smaller number of AC coefficients around DC. This paper shows, experimentally, the compaction capability improvement by choosing the first 1-dimensional transform direction with higher spatial correlation or smaller shifting distance. Two adaptive selection methods are proposed to decide efficiently the spatial direction with higher correlation in a boundary block. One is based on the gradients between DC coefficients of neighboring and current blocks, and the other is based on the final coding efficiency that means the number of bits required for coding the block. Using the MPEG-4 video coder, the proposed method shows coding efficiency gain up to 10.87% compared to the conventional SA-DCT method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.27 no.9
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• pp.825-833
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• 2016

This paper introduces a direct-conversion CMOS RF transmitter for the IEEE 802.15.4 standard with a low-power high-gain up-conversion mixer designed in $0.18{\mu}m$ process. The designed RF DCT(Direct Conversion Transmitter) is composed of differential DAC(Digital to Analog Converter), passive low-pass filter, quadrature active mixer and drive amplifier. The most important characteristic in designing RF DCT is to satisfy the 2.4 GHz Zigbee standard in low power. The quadrature active mixer inside the proposed RF DCT provides enough high gain as well as sufficient linearity using a gain boosting technique. The measurement results for the proposed transmitter show very low power consumption of 7.8 mA, output power more than 0 dBm and ACPR (Adjacent Channel Power Ratio) of -30 dBc.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.285-288
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• 2002

Compression of magnetic resonance images (MRI) has proved to be more difficult than other medical imaging modalities. In an average sized hospital, many tora bytes of digital imaging data (MRI) are generated every year, almost all of which has to be kept. The medical image compression is currently being performed by using different algorithms. In this paper, Fuzzy C-Means (FCM) algorithm is used for the Vector Quantization (VQ). First, a digital image is divided into subblocks of fixed size, which consists of 4${\times}$4 blocks of pixels. By performing 2-D Discrete Cosine Transform (DCT), we select six DCT coefficients to form the feature vector. And using FCM algorithm in constructing the VQ codebook. By doing so, the algorithm can make good time quality, and reduce the processing time while constructing the VQ codebook.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.2
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• pp.172-179
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• 1991

In this paper we have proposed an efficient coding algorithm using directional filtering. First an image is transformed by using DCT which has better energy compaction and then the transformed image is divided into a low frequency component and several high frequency components. The transformed coefficients of each parts are transmitted respectively by using huffman code and these are transformed inversely at receiver. For the directional components total edge images are reconstructed at zero crossing points. We are able to reduce the amount of data by getting of complex component and making directional angles 90. As a results, this proposed method is better than that of Kunt in respect of processing time and memories. We have 38dB of image quality with objective measurement of PSNR and 0.26bpp of compression ratio which is acceptable.

• Journal of information and communication convergence engineering
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• v.6 no.1
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• pp.29-33
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• 2008

This paper presents the proposal and hardware design of JPEG baseline encoder. The JPEG encoder system consists of line buffer, 2-D DCT, quantization, entropy encoding, and packer. A fully pipelined scheme for JPEG encoder is adopted to speed-up an image compression. The proposed architecture was described in VHDL and synthesized in Xilinx ISE 7.1i and simulated by modelsim 6.1i. The results showed that the performance of the designed JPEG baseline encoder is higher than that demanded by real-time applications for $1024{\times}768$ image size. The designed JPEG encoder IP can be easily integrated into various application systems, such as scanner, PC camera, color FAX, and network camera, etc.