• Journal of the Institute of Electronics Engineers of Korea SC
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• v.40 no.5
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• pp.341-349
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• 2003

A real time interactive controller (spectrometer) for magnetic resonance imaging (MRI) system has been developed using high speed digital signal processors (DSP). The controller generates radio frequency (rf) waveforms and audio frequency gradient waveforms and controls multiple receivers for data acquisition. By employing DSPs having high computational power (e.g., TMS320C670l) real time generation of complicated gradient waveforms and interactive control of selection planes are possible, which are important features in real-time imaging of moving organs, e.g., cardiac imaging. The spectrometer was successfully implemented at a 1.5 Tesla whole body MRI system for clinical application. Performance of the spectrometer is verified by various experiments including high- speed imaging such as fast spin echo (FSE) and echo planar imaging (EPI). These high-speed imaging techniques reduce measurement time, however, usually intensify artifact if there is any systematic phase error or jitter in the synchronization between the transmitter, receiver, and gradients.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.939-945
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• 2008

Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.9
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• pp.817-825
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• 2013

The UHDTV system, which provides realistic service with ultra-high definite video and multi-channel audio, has been studied as a next generation broadcasting service. Since the conventional digital terrestrial transmission system is not capable to cover the increased transmission data rate of the UHDTV service, there are great necessity of researches about increase of data rate. Accordingly, the researches has been studied to increase the transmission data rate of the DVB-T2 system using dual-polarized MIMO technique and high order modulation. In order to optimize the MIMO DVB-T2 system where irregular LDPC codes are used, it is necessary to study the design of the bit mapper that matches the LDPC code and QAM symbols in MIMO channel. However, the research related to the design of the bit mapper has been limited to the SISO system. Therefore, this paper defines a new parameter that indicates the VND distribution of MIMO DVB-T2 system and performs the performance analysis according to the parameter which will be helpful for designing a MIMO bit mapper.

• Journal of Korea Multimedia Society
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• v.12 no.8
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• pp.1142-1154
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• 2009

The brand new type of mobile terminal services are kept being introduced in accordance with the development of mobile communication technology. Among many kinds of mobile application services, the PoC application standard which is using instant messaging service and group calls method with the existing walkie-talkie technology has been finished as the version 1.0 after tremendously active discussion and is being continued to be confirmed as 2.0 and 2.1. The PoC Box, which is discussed for replacing the PoC client and intermediate object as a voice messaging box, is currently being introduced and the biggest issues for PoC Box technology topics include the part of saved informations' processing and effective multimedia contents' transmission in the PoC Box system. In this research, we propose that the PaC client could effectively transmit the media to the end-user by specifying the playback location or range, focusing on the contents and the methods of dynamic controlling for saved media in PoC Box. This paper deals with the way of dynamic controlling method using the RTSP which is appropriate for PoC Box and the effective method for generation, expression, processing of various multimedia contents including audio and video objects.

• The Journal of Korean Institute of Next Generation Computing
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• v.13 no.2
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• pp.18-25
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• 2017

In computer science, an encoding is a standardization of converting information to one format for audio, video or text. Therefore, the encoding information of the data should be known to open and read it and there are algorithms detecting encoder of the data. However, some informations of data could be disappeared by packet loss when transmitted on network, especially, if the data is snatched by packet sniffing or eavesdropping from wireless communications. In this paper, we improve the performance of encoding detection algorithm of 'uchardet' program for multi-byte encoded data with errors based on bit-shift algorithm. To simulate the performance, we generated Korean and Japanese text data with errors that is removed some random bits at random positions. Then the detection algorithm are tested using the data and 'uchardet-bitshift' showed better performance than 'uchardet'. When Korean texts are used, 'uchardet' could detect perfectly with ≤0.005% errors but it showed 0% detection rate with ≥1% errors while 'uchardet-bitshift' detected perfectly with ≤0.05% errors and it showed correct detection cases with ≥1% errors. Japanese texts with errors tend to report falsely as Chinese encoding because Japanese texts include lots of Chinese characters. As a results, we improved encoding detection algorithms by applying bit shift operation.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.197-206
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• 2011

In this paper, we proposes the virtual sound source panning algorithm in the multichannel system. Recently, High-definition (HD) and Ultrahigh-definition (UHD) video formats are accepted for the multimedia applications and they provide the high-quality resolution pixels and the wider view angle. The audio format also needs to generate the wider sound field and more immersive sound effects. However, the conventional stereo system cannot satisfy the desired sound quality in the latest multimedia system. Therefore, the various multichannel systems that can make more improved sound field generation are proposed. In the mutichannel system, the conventional panning algorithms have acoustic problems about directivity and timbre of the virtual sound source. To solve these problems in the arbitrary positioned multichannel loudspeaker system, we proposed the virtual sound source panning algorithm using multiple vectors base nonnegative amplitude panning gains. The proposed algorithm can be easily controlled by the gain control function to generate an accurate localization of the virtual sound source and also it is available for the both symmetric and asymmetric loudspeakers format. Its performance of sound localization is evaluated by subjective tests comparing with conventional amplitude panning algorithms, e.g. VBAP and MDAP, in the symmetric and asymmetric formats.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.573-578
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• 2008

Recently, we should control various devices such as TV, audio, DVD player, video player, and set-top box simultaneously to manipulate home theater system. To execute the function the user want in this situation, user should know functions and positions of the buttons in several remote controllers. Normally, people feel difficult due to these realistic problems. Besides, the number of the devices that we can control shall increase, and people will confuse more if the ubiquitous home environment is realized. Therefore, user adaptive interface that provides the summarized functions is required. Moreover there can be a lot of mobile and stationary controller devices in ubiquitous computing environment, so user interface should be adaptive in selecting the functions that user wants and in adjusting the features of UI to fit in specific controller. To implement the user and controller adaptive interface, we modeled the ubiquitous home environment and used modeled context and device information. We have used Bayesian network to get the degree of necessity in each situation. Behavior selection network uses predicted user situation and the degree of necessity, and it selects necessary functions in current situation. Selected functions are used to construct adaptive interface for each controller using presentation template. For experiments, we have implemented ubiquitous home environment and generated controller usage log in this environment. We have confirmed the BN predicted user requirements effectively as evaluating the inferred results of controller necessity based on generated scenario. Finally, comparing the adaptive home UI with the fixed one to 14 subjects, we confirmed that the generated adaptive UI was more useful for general tasks than fixed UI.

• Journal of Broadcast Engineering
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• v.22 no.6
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• pp.744-754
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• 2017

This paper compares transport overhead of MPEG-2 TS, MMT and ROUTE in order to compare transport efficiency between the DTV and UHDTV. The MPEG-2 TS standard, widely used, was established for multiplexing and synchronizing encoded audio and video, additional information. In recent years, MMT and ROUTE was established as a next generation multimedia transport standard for the new broadcasting communication environment. In this paper, we compare and analyze transport overhead about three protocol. In order to analysis, we captured the UHD A/V stream in real-time broadcasting service using ROUTE and MMT, and we calculated and analyzed transport overhead using the overhead analysis program which was developed in our laboratory. Furthermore, for comparison under the same conditions, we assumed the MPEG-2 TS stream by extracting ES of UHD A/V stream based on the DTV standard. In this paper, we show the results of protocol transport efficiency in case of basic A/V stream except for additional services. And result show that MMT and ROUTE have similar overhead and MPEG-2 TS is relatively small overhead. However, since MPEG-2 TS result does not consider null packets, it is expected that the relative overhead difference will be reduced.

• Smart Media Journal
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• v.11 no.10
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• pp.65-75
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• 2022

Deep learning is used as a creative tool that could overcome the limitations of existing analysis models and generate various types of results such as text, image, and music. In this paper, we propose a method necessary to preprocess audio data using the Niko's MIDI Pack sound source file as a data set and to generate music using Bi-LSTM. Based on the generated root note, the hidden layers are composed of multi-layers to create a new note suitable for the musical composition, and an attention mechanism is applied to the output gate of the decoder to apply the weight of the factors that affect the data input from the encoder. Setting variables such as loss function and optimization method are applied as parameters for improving the LSTM model. The proposed model is a multi-channel Bi-LSTM with attention that applies notes pitch generated from separating treble clef and bass clef, length of notes, rests, length of rests, and chords to improve the efficiency and prediction of MIDI deep learning process. The results of the learning generate a sound that matches the development of music scale distinct from noise, and we are aiming to contribute to generating a harmonistic stable music.