• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1409-1418
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• 1998

The recent audio CODEC(Coding/Decoding) algorithms are complex of several coding techniques, and can be divided into DSP tasks, controller tasks and mixed tasks. The traditional DSP processor has been designed for fast processing of DSP tasks only, but not for controller and mixed tasks. This paper presents a new architecture that achieves high throughput on both controller and mixed tasks of such algorithms while maintaining high performance for DSP tasks. The proposed processor, YSP-3, operates four algorithms while maintaining high performance for DSP tasks. The proposed processor, YSP-3, operates functional units (Multiplier, two ALUs, Load/Store Unit) in parallel via 4-issue super-scalar instruction structure. The performance evaluation of YSP-3 has been done through the implementation of the several DSP algorithms and the part of the AC-3 decoding algorithms.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.30A no.3
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• pp.16-33
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• 1993

New decoding algorithm of double-error-correction Reed-Solmon codes over GF(2$^{8}$) for optical compact disks is proposed and decoding algorithm of RS codes with triple-error-correcting capability is presented in this paper. First of all. efficient algorithms for estimating the number of errors in the received code words are presented. The most complex circuits in the RS decoder are parts for soving the error-location numbers from error-location polynomial, so the complexity of those circuits has a great influence on overall decoder complexity. One of the most known algorithm for searching the error-location number is Chien's method, in which all the elements of GF(2$^{m}$) are substituted into the error-location polynomial and the error-location number can be found as the elements satisfying the error-location polynomial. But Chien's scheme needs another 1 frame delay in the decoder, which reduces decoding speed as well as require more stroage circuits for the received ocode symbols. The ther is Polkinghorn method, in which the roots can be resolved directly by solving the error-location polynomial. Bur this method needs additional ROM (readonly memory) for storing tthe roots of the all possible coefficients of error-location polynomial or much more complex cicuit. Simple, efficient, and high speed method for solving the error-location number and decoding algorithm of double-error correction RS codes which reudce considerably the complexity of decoder are proposed by using Hilbert theorems in this paper. And the performance of the proposed decoding algorithm is compared with that of conventional decoding algorithms. As a result of comparison, the proposed decoding algorithm is superior to the conventional decoding algorithm with respect to decoding delay and decoder complexity. And decoding algorithm of RS codes with triple-error-correcting capability is presented, which is suitable for error-correction in digital audio tape, also.

• Journal of KIISE:Computer Systems and Theory
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• v.31 no.9
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• pp.527-535
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• 2004

Handling mixed workload in digital set-top box or streaming server becomes an important issue as integrated file system gets momentum as the choice for the next generation file system. The next generation file system is required to handle real-time audio/video playback while being able to handle text requests such as web page, image file, etc. Legacy file system provides only best effort I/O service and thus cannot properly support the QoS of soft real-time I/O. In this paper, we would like to present our experience in developing the file system which fan guarantee the QoS of multimedia stream. We classify all application I/O requests into two category: periodic I/O and sporadic I/O. The QoS requirement of multimedia stream could be guaranteed by giving a higher priority to periodic requests than sporadic requests. The proto-type file system(Qosfs) is developed on Linux Operating System.

• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.3
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• pp.285-289
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• 2004

Recently, due to the high development of the internet, needs for multimedia streams such as digital audio and video is increasing much more. In case of transmitting multimedia streams using the User Datagram Protocol (UDP), it may cause starvation of TCP traffic on the same transmission path, thus resulting in congestion collapse and enormous delay because UDP does not perform TCP-like congestion control. Because of this problem, diverse researches are being conducted on new transmission schemes and protocols intended to efficiently reduce the transmission delay of real-time multimedia streams and perform congestion control. The TCP-friendly congestion control schemes can be classified into the window-based congestion control, which uses the general congestion window management function, and the rate-based congestion control, which dynamically adjusts transmission rate by using TCP modeling equations and the like. In this paper, we suggest the square-root congestion avoidance algorithm with the hybrid TCP-friendly congestion control scheme which the window-based and rate-based congestion controls are dealt with in a combined way. We apply the proposed algorithm to the existing TEAR. We simulate the performance of the proposed TEAR by using NS, and the result shows that it gives better improvement in the stability needed for providing congestion control than the existing TEAR.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.304-309
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• 2004

Recently, the needs for multimedia stream such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic thorough the Internet, does not apply any congestion control mechanism to regulate the data flow thorough the shared network And it leads to congestion collapse of the Internet and results in long-term transmission delay. To avoid any adverse effect on the current Internet functionality, a now protocol of modification or addition of some functionality to perform congestion control and to reduce huge transmission delay in transmitting of multimedia stream are in study. TCP-friendly congestion control mechanism is classified into two : one is window-based congestion control scheme using general window management functionalities, the other is rate-based congestion control scheme using TCP modeling equation. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and late-based congestion control for multimedia stream. And we also simulate the performance of improved TEAR implementation using NS. With He simulation results, we show that the improved TEAR can provide better fairness and lower rate fluctuations than TCP.

• International Journal of Highway Engineering
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• v.15 no.6
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• pp.41-47
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• 2013

PURPOSES : The purpose of this study is about noise which is generated from roads and is consist of irregular frequency variation from low frequency to various band. The existing methods of noise reduction are sound barrier that uses insulation material and absorbing material or have applied passive technology of noise reduction by devices. The total frequency band is needed to apply active noise control. METHODS : In this study applies to the field of road traffic environment, signal processing controller and various analog signal input/output, the amplifier module is based on parallel-core embedded processor designed. DSP performs the control algorithm of the road traffic noise. Noise sources in the open space performance of evaluation were applied. In this study, controller of active signal processor was designed based on the module of audio input/output and main controller of embedded process. The controller of active signal processor operates noise reduction algorithm and performance tests of noise reduction in inside and outside environment were executed. RESULTS : The signal processing controller with OMAP-L137 parallel-core processors as the center, DSP processors in the active control operations dealt with quickly. To maximize the operation speed of an object and ARM processor is external function keys and display for functions and evaluating the performance management system was designed for the purpose of the interface. Therefore the reduction of road traffic noise has established an electronic controller-based noise reduction. CONCLUSIONS : It is shown that noise reduction is effective in the case of pour tonal sound and complex tonal sound below 500Hz by appling to Fx-LMS.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.4077-4091
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• 2017

This paper presents the design of a broadcasting scenario and system for an 8K-resolution content. Due to an 8K content is four times larger than the 4K content in terms of size, many technologies such as content acquisition, video coding, and transmission are required to deal with it. Therefore, high-quality video and audio for 8K (ultra-high definition television) service is not possible to be transmitted only using the current terrestrial broadcasting system. The proposed broadcasting system divides the 8K content into four 4K contents by area, and each area is hierarchically encoded by Scalable High-efficiency Video Coding (SHVC) into three layers: L0, L1, and L2. Every part of the 8K video content divided into areas and hierarchy is independently treated. These parts are transmitted over heterogeneous networks such as digital broadcasting and broadband networks after going through several processes of generating signal messages, encapsulation, and packetization based on MPEG media transport. We propose three methods of generating streams at the sending entity to merge the divided streams into the original content at the receiving entity. First, we design the composition information, which defines the presentation structure for displays. Second, a descriptor for content synchronization is included in the signal message. Finally, we define the rules for generating "packet_id" among the packet header fields and design the transmission scheduler to acquire the divided streams quickly. We implement the 8K broadcasting system by adapting the proposed methods and show that the 8K-resolution contents are stably received and serviced with a low delay.

• The Journal of the Convergence on Culture Technology
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• v.8 no.4
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• pp.297-302
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• 2022

Contemporary viewers familiar with the digital space show their desire for self-expression and use voice, text and gestures as tools for expression. The purpose of this study is to create interactive art that expresses the narrative uttered by the viewer in the form of a collage using the viewer's figure, and reproduces and expands the story by the viewer's movement. The proposed interactive art visualizes audio and video information acquired from the viewer in a text collage, and uses gesture information and a natural user interface to easily and conveniently interact in real time and express personalized emotions. The three pieces of information obtained from the viewer are connected to each other to express the viewer's current temporary emotions. The rigid narrative of the text has some degree of freedom through the viewer's portrait images and gestures, and at the same time produces and expands the structure of the story close to reality. The artwork space created in this way is an experience space where the viewer's narrative is reflected, updated, and created in real time, and it is a reflection of oneself. It also induces active appreciation through the active intervention and action of the viewer.

• Journal of Korean Society of Archives and Records Management
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• v.22 no.4
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• pp.129-149
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• 2022

Records have temporal characteristics, including the past and present; linguistic characteristics not limited to a specific language; and various types categorized in a complex way. Processing records such as text, video, and audio in the life cycle of records' creation, preservation, and utilization entails exhaustive effort and cost. Primary natural language processing (NLP) technologies, such as machine translation, document summarization, named-entity recognition, and image recognition, can be widely applied to electronic records and analog digitization. In particular, Korean deep learning-based NLP technologies effectively recognize various record types and generate record management metadata. This paper provides an overview of Korean NLP technologies and discusses considerations for applying NLP technology in records management. The process of using NLP technologies, such as machine translation and optical character recognition for digital conversion of records, is introduced as an example implemented in the Python environment. In contrast, a plan to improve environmental factors and record digitization guidelines for applying NLP technology in the records management field is proposed for utilizing NLP technology.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2021.05a
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• pp.584-588
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• 2021

The impedance of the speaker is likely to be recognized as a fixed value. However, speaker impedance continues to vary with frequency variation, especially larger in resonant frequency region. The sound pressure level of loudspeakers is determined by the current flowing throughout the coil that consists loudspeakers. If loudspeakers are driven by voltage, sound pressure level of the loudspeaker is distorted by the variation of loudspeaker impedance. Current-drive of loudspeakers can solve this problem, but distortion of sound pressure level occurs in low frequencies due to resonance. The distortion can degrade the sound quality of the sound system. So to solve this problem, In this paper, we propose a resonance compensation circuit using DSP. we simulates audio systems using an equivalent model of loudspeakers to verify distortion of sound pressure level due to impedance variation and propose a circuit to compensate it. The proposed circuit is configured using a state variable filter and it can adjust the center frequency and output, so it will be used various sound systems.