• 전기의세계
• /
• v.32 no.5
• /
• pp.263-272
• /
• 1983

본고에서는 음성코우딩 할 때 제일 고려되어야할 음질과, 현재 많이 사용되고 있는 대표적인 음성코우딩 방식, 음성코우딩 방식을 실시간(real-time)에서 동작시키기 위해서 현재 많이 사용하고 있는 DSP(digital-signal-processor)에 대해 설명한다.

• The Journal of Natural Sciences
• /
• v.5 no.2
• /
• pp.33-38
• /
• 1992

A new texture segmentation-based image coding technique which performs segmentation based on roughness of textural regions and properties of the human visual system (HVS) is presented for multime-dia teleconference. The segmentation is accomplished by thresholding the fractal dimension so that textural regions are classified into three texture classes; perceived constant intensity, smooth texture, and rough texture. An image coding system with high compression and good image quality is achieved by developing an efficient coding technique for each segment boundary and each texture class. We compare the coding efficiency of this technique with that of a well established technique (discrete cosine transform (DCT) image coding).

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1986.10a
• /
• pp.133-136
• /
• 1986

• KLA journal
• /
• v.21 no.5
• /
• pp.2-12
• /
• 1980

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1985.10a
• /
• pp.113-117
• /
• 1985

• Proceedings of the KIEE Conference
• /
• 1983.07a
• /
• pp.103-105
• /
• 1983

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.8 no.4
• /
• pp.145-150
• /
• 1983

The runlength coding algorithm widely used for graphic data compression is extended for multi-level general images. It is applied to the quantized prediction error signal obtained by a simple predictor and quantizer. It is shown that this DPCM/RLC algorithm of a modest complexity performs much better than the conventional discrete consine transform and DPCM methods.

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1984.04a
• /
• pp.18-21
• /
• 1984

In this paper, the tree coding using the (M, L) multi-path search algorithm has teen investigated. A hybrid adaptation scheme which employs a block adaptation as well as a sequential dadptation is described for application in quantization and compression of speech signals. Simulation results with the gybrid adaptation scheme indicate that a relatively good speech quality can be obtained at rate about 8Kbps. All necessary parameters such as MlL and filter-order were found from simulation and these parameters turned out to be a good compromise between the complexity and overall performance.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.22 no.5
• /
• pp.29-38
• /
• 1985

In this paper we have designed and implemented in real time a simple, efficient and flexible AOPCM cosec using a high speed digital processor, NEC 7720. For ADPCM system, we have used an instantaneous adaptive quantizer and a first-order fixed predictor. The software for NEC 7720 has been developed and it was found that the NEC 7720 was capable of performing the entire ADPCAt algorithm for 4 channels in real time as optimizing the program. Computer simulation has born made to investigate a computational accuracr of NEC 7720 and to de-termine necessary parameters for a ADPCM codec. Real telephone speech, RC-shaped Gaussian noise and 1004 Hz tone signal were used for simulation. In simulation, the parameters werc optimized from the computed SNR and the informal listening test. The developed software was tested in real time operation using a hardware emulator for NEC 7720. It took a maximum 23.25$\mu$s to encode one sample and 113.5$\mu$s, including all the necessary 1/0 operations, to encode 4 channels. In the case of decoding process, it took 24.75$\mu$s to decode one sample and 119.5$\mu$s to decode 4 channels.