• Proceedings of the Korea Contents Association Conference
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• 2003.11a
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• pp.347-351
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• 2003

It is important to consider not only audio/video but presentation way, in video conferencing, seminar, lecture based network communication. The method to transmit image of a chalkboard used in traditional seminar, conference, lecture are one of difficulties for remote environment since it requires techniques high-resolution transmission and talking films. In this paper we implemented real-time multimedia shared board system for shared effective multimedia presentation using connection-oriented socket of TCP. In this system, decreasing cost of related system construction, improve of data reliableness and interaction, presentation delay time be able to minimize.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.1
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• pp.129-137
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• 2010

There are a lot of multimedia functions getting included in mobile devices recently, not only simple audio but visual functions as well, such as video playing, game and web browsing. Such the visual multimedia functions brought the strong requirement for much bigger LCD Display and high resolution. And the brightness of LCD is also one of major requirements and it is required to be more bright. However, LCD brightness is limited to increase because of power consumption. It occupies nearly 30% of the whole system power. This thesis suggests an effective and simple method of adjusting the LCD backlight brightness, considering a characteristic of battery and user pattern.

• Proceedings of the KIEE Conference
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• 1996.07b
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• pp.1295-1297
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• 1996

CCITT G.721, G.723 standard ADPCM algorithm is implemented by using TI's fixed point DSP start kit (DSK). ADPCM can be implemented on a various rates, such as 16K, 24K, 32K and 40K. The ADPCM is sample based compression technique and its complexity is not so high as the other speech compression techniques such as CELP, VSELP and GSM, etc. ADPCM is widely applicable to most of the low cost speech compression application and they are tapeless answering machine, simultaneous voice and fax modem, digital phone, etc. TMS320C50 DSP is a low cost fixed point DSP chip and C50 DSK system has an AIC (analog interface chip) which operates as a single chip A/D and D/A converter with 14 bit resolution, C50 DSP chip with on-chip memory of 10K and RS232C interface module. ADPCM C code is compiled by TI C50 C-compiler and implemented on the DSK on-chip memory. Speech signal input is converted into 14 bit linear PCM data and encoded into ADPCM data and the data is sent to PC through RS232C. The ADPCM data on PC is received by the DSK through RS232C and then decoded to generate the 14 bit linear PCM data and converted into the speech signal. The DSK system has audio in/out jack and we can input and out the speech signal.

• Journal of the Society of Disaster Information
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• v.7 no.1
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• pp.51-63
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• 2011

The purpose of this study is to prevent some accidents from the tragedy related to school accidents in elementary, middle and high schools. However, it is necessary to establish the program of school safety and change the view of the safety because safety measures are being taken after the accident, not prevention. There are some resolution to prevent school accidents by developing manuals, educating school safety for teachers, utilizing audio-video education and private organization, it is difficult for teachers to handle a fire alarm and electric safety inspection, and check condition of health, training experts through qualifying exams.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.906-909
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• 2006

This paper proposes a new body-effect compensated switch configuration for low voltage and low distortion switched-capacitor (SC) applications. The proposed circuit allows rail-to-rail switching operation for low voltage SC circuits and has better total harmonic distortion than the conventional bootstrapped circuit by 19 dB. A 2-1 cascaded sigma-delta modulator is provided for performing the high-resolution analog-to-digital conversion on audio codec in a communication transceiver. An experimental prototype for a single-stage folded-cascode operational amplifier (opamp) and a 2-1 cascaded sigma-delta modulator has been implemented in a 0.25 micron double-poly, triple-metal standard CMOS process with 2.7 V of supply voltage.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.05a
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• pp.451-454
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• 2009

There are a lot of multimedia functions getting included in mobile devices recently, not only simple audio but visual functions as well, such as video playing, game and web browsing. Such the visual multimedia functions brought the strong requirement for much bigger LCD Display and high resolution. And the brightness of LCD is also one of major requirements and it is required to be more bright However, LCD brightness is limited to increase because of power consumption. It occupies nearly 30% of the whole system power. This thesis suggests an effective and simple method of adjusting the LCD backlight brightness, considering a characteristic of battery and user pattern.

• Journal of Korea Entertainment Industry Association
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• v.15 no.4
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• pp.267-277
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• 2021

Both teachers and learners should participate in non-face-to-face class due to COVID-19. The non-face-to-face class has brought about many problems, where they made adequate preparations for such abrupt situation. This study attempted to understand and improve problems occurring during non-face-to-face midi class. The findings are as follows: First, there were differences in equipment available to contact and non-face-to-face class. Such a problem could be improved by using Reaper, DAW which can be installed and freely utilized without any functional limits, regardless of the types of operating systems. Second, latency could not be reduced, when the screen share function of Zoom was used, since it was impossible to select audio interface's drivers in DAW. This problem was improved by again receiving audio output as input and sending it, from the perspectives of teachers. In addition, learners who used the operating system of Windows and have no audio interfaces usually suffer from latency during practices. The latency can be reduced by installing Asio4all. Third, image degradation and screen disconnection phenomena occurred due to the lack of resource. Two computers were connected by using a capture board and the screen disconnection phenomena could be improved by distributing resources and maintaining high-resolution. The system for allowing non-face-to-face midi class could be successfully established, as one more computer was connected by using Vienna Ensemble Pro and more plug-ins were used by securing additional resources. Consequently, the problems of non-face-to-face midi class could be understood and improved.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.4 no.2
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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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8B
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• pp.1460-1468
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• 2000

The goal of this research is the real-time implementation of MPEG-1 Layer III audio decoder using the fixed-point digital signal processor of TMS320C6201 The main job for this work is twofold: one is to convert floating-point operation in the decoder into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real-time with memory size as small as possible. We, especially, devote much time to the descaling module in the decoder for conversion of floating-point operation into fixed-point operation with high accuracy. The inverse modified cosine transform(IMDCT) and synthesis polyphase filter bank modules are optimized in order to reduce the amount of computation and memory size. After the optimization process, in this paper, the implemented decoder uses about 26% of maximum computation capacity of TMS320C6201. The program memory, data ROM, data RAM used in the decoder are about 6.77kwords, 3.13 kwords and 9.94 kwords, respectively. Comparing the PCM output of fixed-point computation with that of floating-point computation, we achieve the signal-to-noise ratio of more than 60 dB. A real-time operation is demonstrated on the PC using the sound I/O and host communication functions in the EVM board.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.1
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• pp.1-9
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• 2007

Digital Multimedia Broadcasting is the next generation broadcasting service which enables various digital multimedia contents, i.e., audio and video, and data access for mobile users. However, due to the bandwidth limitation, the spatial resolution is limited to CIF(Common Interleaved Frame). The Advanced Terrestrial DMB (AT-DMB) secures additional bandwidth by adopting hierarchical modulation transmission technology and provides high data rate and quality for mobile multimedia broadcasting services with scalable video coding(SVC). This paper proposes scalable video coding technology for AT-DMB which enables high quality mobile multimedia broadcasting services that exceeds current DMB service's quality and contents capability.