• Journal of Korea Multimedia Society
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• v.11 no.5
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• pp.579-587
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• 2008

Typically, the file sizes of JPEG compressed images with various complexity differ from images regardless of same image size. So, it is not easy to estimate the remaining image counts that should be stored in the limited storage equipped with the digital camera. To solve the problem, the bit rate control employs the modification of quantization table. The previous work assumed that there is linear relation between image activity and modification factor of quantization table, but in this paper, more accurate functional relations based on statistics are employed to improve the bit rate control accuracy. Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.38 no.2
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• pp.87-94
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• 2001

A soft error rate for DRAM was predicted in connection with the leakage current in cell capacitor. The charge in cell capacitor was decreased during the DRAM operation, and soft error retes due to the leakage current were calculated in various operation mode of DRAM. It was found that the soft error rate of the /bit mode was dominant with small leakage current, but as increasing the leakage current memory mode shown the dominant effect on soft error rate. Using the 256M grade DRAM structure it was predicted that the soft error rate was influenced by the change of the cell capacitance, bit line capacitance, and the input voltage sensitivity of sense amplifier, and these results can be used to the design of the optimum cells in the next generation DRAM development.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.42-47
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• 2001

This paper improves the performance of VAD (Voice Activity Detector) in G.723.1 Annex A 6.3kbps/5.3kbps dual rate speech coder, which is developed for Internet Phone and videoconferencing. The VAD decision is based on a three-level energy threshold. We evaluates for processing time, speech quality, and bit rate. The processing time is reduced due to the accuracy of VAD decision on the silence period. On subjective quality test there is almost no difference compared with the G.723.1. In order to measure the bit rate we count the active speech frame (VAD=1) and we can reduce more bit rate as silence periods are shown.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.142-147
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• 1998

Recently, 8 kb/s CS-ACELP coder of G.729 is atandardized by ITU-T SG15 and it has been reported that the speech quality of G729 is better than or equal to that of 32kb/s ADPCM. However G.729 is the fixed rate speech coder, and it does not consider the property of voice activity in mutual conversation. If we use the voice activity, we can reduce the average bit rate in half without any degradations of the speech quality. In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and algorithm, we combine the energy-thresholding method, the phonetic segmentation method by integration of various feature parameters obtained through the analysis procedure, and the variable hangover period method. Through the analysis of noise features, the 1 kb/s sub rate coder is designed for coding the background noise signal. So, we design the 4 kb/s sub rate coder for the unvoiced parts. The performance of the variable rate algorithm is evaluated by the comparison of speed quality and average bit rate with G.729. Subjective quality test is also done by MOS test. Conclusively, it is verified that the proposed variable rate CS-ACELP coder produced the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2019.11a
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• pp.180-181
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• 2019

This paper proposes a rate control algorithm for intra coding frame in HEVC encoder using a deep learning approach. The proposed algorithm is designed for CTU level bit allocation in intra frame by considering visual features spatially and temporally. Our features are generated using visual geometry group (VGG-16) with deep convolutional layers, then it is used for bit allocation per each CTU within an intra frame. According to our experiments, the proposed algorithm can achieve -2.04% Luma component BD-rate gain with minimal bit accuracy loss against the HM-16.20 rate control model.

• Speech Sciences
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• v.10 no.4
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• pp.125-136
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• 2003

Speech coding techniques have been studied to reduce the complexity and bit rate but also to improve the sound quality. CELP type vocoder, has used as a one of standard, supports the great sound quality even low bit rate. In this paper, the preprocessing of input speech to reduce the bit rate is the different with the conventional vocoder. The different kinds of parameter are used for the preprocessing so this paper is compared with theses parameters for finding the more appropriate parameter for the vocoder. The parameters are used to synthesize the speech not to encode or decode for coding technique so we proposed the simple algorithm not to have the influence on the processing time or the computation time. The parameters in used the preprocessing step are speaking rate, duration and PSOLA technique.

• Proceedings of the IEEK Conference
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• 2000.11b
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• pp.164-167
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• 2000

This paper introduces the design of high-speed analog-to-digital converter for hard disk drive (HDD) read channel. This is based on autozero technique for low-error rate, and Double Speed Dual ADC(DSDA) technique lot efficiently increasing the conversion speed of A/D converter. This An is designed by 6bit resolution, 800M sample/s maximum conversion rate, 390㎽ power dissipation, one clock cycle latency in 0.65 $\mu\textrm{m}$ CMOS technology.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.5
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• pp.1384-1396
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• 1998

This paper ivestigates the performance of a relative rate (RR) switch algorithm for the rate-based available bit rate (ABR) flow control in asynchronous transfer mode (ATM) networks. A RR switch may notify the network congestion status to the source by suing the congestion indication (CI) bit or no increase (NI)bit in the backward RM (BRM) cells. A RR switch can be differently implemented according to the congestion detectio and notification methods. In this paper, we propose three implementation schemes for the RR switch with different congestion detection and notification methods, and analyze the allowed cell rate (ACR) of a source and the queue length of a switch in steady state. In addition, we derive the upper and lower bounds for the maximum and minimum queue lengths for each scheme respectively, and evaluate the effects of the ABR parameter values on the queue length. Furthermore, we suggest the range of the rage increase factor (RIF) and rate decrease factor (RDF) parameter values which can prevent buffer overflow and underflow at a switch.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.7
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• pp.125-131
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• 1989

This paper proposes a new ABTC (Adaptive Block Truncation Coding) algorithm which improves the BTC algorithm for image data compression. A new adaptive block truncation coding which adopts a selective coding scheme depending on the local characteristics of an image has been described. The characteristics of the ABTC algorithm can be summarized as high compression ratio and the algorithm simplicity. Using this algorithm, color images can be coded at a variable bit rate from 1.0 (bit/pel) to 2.56 (bit/pel) and high compression rate (1.3-105 bit/pel) can be achieved without conspicuous image degradation compared with original images.

• Journal of the Korean Society of Radiology
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• v.5 no.6
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• pp.329-334
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• 2011

This paper presents improvement of compression rate for SPIHT algorithm based on wavelet compression through redundancy bit removing. Proposed SPIHT algorithm uses a method to select of optimized threshold from feature of wavelet transform coefficients and removes sign bit only if coefficient is LL area. Finally Proposed SPIHT algorithm applies to Huffman coding. Experimental results show that the proposed algorithm achieves more improvement bit rate and more fast progressive transmission with low bit rate.