• Journal of KIISE:Computing Practices and Letters
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• v.5 no.4
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• pp.440-448
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• 1999

본 논문은 객체지향형 분산처리 환경 하에서 오디오나 비디오 등과 같은 실시간(real-time) 스트림(stream) 데이타를 처리하는 데 필요한 소프트웨어 기반구조를 설계하고 구현한 내용을 기술한다. 본 논문에서 제시한 DAViS(Distributed Object Framework supporting Audio/Video Streaming)는, 오디오/비디오 데이타의 처리와 관련된 여러 소프트웨어 구성요소들을 분산객체로 추상화하고, 그 객체들간의 제어정보 교환경로와 오디오/비디오 데이타 전송경로를 서로 분리하여 처리한다. 분산응용프로그램 작성자는 DAViS에서 제공하는 서비스들을 이용하여, 기존의 분산프로그래밍 환경이 제공하는 것과 동일한 수준에서 오디오/비디오 데이타에 대한 처리를 표현할 수 있다. DAViS는, 새로운 형식의 오디오/비디오 데이타를 처리하는 부분을 손쉽게 통합하고, 하부 네트워크의 전송기술이나 컴퓨터시스템 관련 기술의 진보를 신속하고 자연스럽게 수용할 수 있도록 하는 유연한 구조를 가지고 있다. Abstract This paper describes the design and implementation of software framework which supports the processing of real-time stream data like audio and video in distributed object-oriented computing environment. DAViS(Distributed Object Framework supporting Audio/Video Streaming), proposed in this paper, abstracts software components concerning the processing of audio/video data as distributed objects and separates the transmission path of data between them from that of control information. Based on DAViS, distributed applications can be written in the same abstract level as is provided by the existing distributed environment in handling audio/video data. DAViS has a flexible internal structure enough to easily incorporate new types of audio/video data and to rapidly accommodate the progress of underlying network and computer system technology with very little modifications.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.618-629
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• 2009

The MPEG-4 Audio Lossless Coding (ALS) standard belongs to the family MPEG-4 audio coding standards. In contrast to lossy codecs such as AAC, which merely strive to preserve the subjective audio quality, lossless coding preserves every single bit of the original audio data. The ALS core codec is based on forward-adaptive linear prediction, which combines remarkable compression with low complexity. Additional features include long-term prediction, multichannel coding, and compression of floating-point audio material. This paper describes the basic elements of the ALS codec with a focus on prediction, entropy coding, and related tools and points out the most important applications of this standardized lossless audio format.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.6
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• pp.180-187
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• 2013

In this paper, we propose a LED communication based multi-hop audio data transmission network system. The main contribution and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit audio data signal with long distance via multi-hops. Second, the developed system has the following features: In transmitter, audio data is transmitted after encoding with S/PDIF format via a general LED. The relay receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality audio transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted audio quality according to the used LED colors.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1567-1570
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• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.3
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• pp.21-31
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• 2014

In this paper, we propose VLC(visible light communication) based multi-hop multimedia data transmission system. The main contributions and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit multimedia data(audio data, video data) with long distance. Second, the developed system has the following features: In transmitter, audio data and video data are transmitted via multi-hops using two channels. The relay in audio channel receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The relay in video channel receives video signal by using photo diode and then amplify the video signal using OP-AMP and then transmits the signal to receiver. The receiver amplifies the received signal from photo diode and then sends it to the monitor. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality multimedia data transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted multimedia(audio and video) quality according to the used LED colors.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.74-78
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• 2004

As analog audio systems are changing to digital systems, the DAB (Digital Audio Broadcasting) is expected to provide CD quality audio, various data services with interactiveness and excellent mobile reception ability. The DMB (Digital Multimedia Broadcasting), as more advanced successor of the DAB, adds video capability on the audio and data services. The DAB system assures high quality audio services even when the reception is through portable and mobile receivers. In this paper, USB-DAB receiver and PCI-DMB receiver are designed and implemented. The DAB receiver and the DMB receiver incorporate with PC to make use of computational power and application software of Pc. This enables the developed system to be more flexible and to meet various applications easier.

• Journal of the Korea Society of Computer and Information
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• v.24 no.5
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• pp.59-66
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• 2019

According to the KT telecommunication statistics, people stayed inside their houses on an average of 11.9 hours a day. As well as, according to NSC statistics in the united states, people regardless of age are injured for a variety of reasons in their houses. For purposes of this research, we have investigated an abnormal event detection algorithm to classify infrequently occurring behaviors as accidents, health emergencies, etc. in their daily lives. We propose a fusion method that combines three classification algorithms with vision pattern, audio pattern, and activity pattern to detect unusual user events. The vision pattern algorithm identifies people and objects based on video data collected through home CCTV. The audio and activity pattern algorithms classify user audio and activity behaviors using the data collected from built-in sensors on their smartphones in their houses. We evaluated the proposed individual pattern algorithm and fusion method based on multiple scenarios.

• Journal of KIISE:Computing Practices and Letters
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• v.9 no.4
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• pp.470-485
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• 2003

The steganography is one of methods that users can hide data. Some steganography softwares use audio data among multimedia data. However, these commercialized audio steganography softwares have disadvantages that the existence of hidden messages can or easily recognized visually and only certain-sized data can be hidden. To solve these problems, this study suggested, designed and implemented Dynamic Message Embedding (DME) algorithm. Also, to improve the security level of the secret message, the file encryption algorithm has been applied. Through these, StegoWaveK system that performs audio steganography was designed and implemented. Then, the suggested system and the commercialized audio steganography system were compared and analyzed on criteria of the Human Visilable System (HVS), Human Auditory System (HAS), Statistical Analysis (SA), and Audio Measurement (AM).

• Journal of the Korea Society of Computer and Information
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• v.12 no.6
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• pp.97-104
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• 2007

This paper presents the problem of the feature extraction techniques that has been used a content-based analysis, classification and retrieval in audio data and proposes a course of the preprocessing for a new contents-based retrieval methods. Because the feature vector according to sampling value changes, the existing audio data analysis is problem that same music is appraised by other music. Therefore, we propose waveform information extraction method of PCM data for retrieval audio data of various format to contents-based. If this method is used. we can find that audio datas that get into sampling in various format are same data. And it may be applied in contents-based music retrieval system. To verity the performance of the method, an experiment was done feature extraction using STFT and waveform information extraction using PCM data. As a result, we could know that the method to propose is effective more.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.12
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• pp.108-116
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• 2009

In this paper, we propose a Ethernet based audio distribution system with a new L1/L2 hybrid switching scheme, and evaluate its performance. The proposed scheme not only offers guaranteed low latency and jitter characteristics that are essentially required for the distribution of high-quality uncompressed audio traffic, and but also provide an efficient transmission of data traffic on the Ethernet environment. The audio distribution system with a proposed scheme consists of a master node and a number of relay nodes, and all nodes are mutually connected as a daisy-chain topology through up and downlinks. The master node generates an audio frame for each cycle of 125us, and the audio frame has 24 time slotted audio channels for carrying stereo 24 channels of 16-bit PCM sampled audio. On receiving the audio frame from its upstream node via the downlink, each intermediate node inserts its audio traffic to the reserved time slot for itself, then relays again to next node through its physical layer(L1) transmission - repeating. After reaching the end node, the audio frame is loopbacked through the uplink. On repeating through the uplink, each node makes a copy of audio slot that node has to receive, then play the audio. When the audio transmission is completed, each node works as a normal L2 switch, thus data frames are switched during the remaining period. For supporting this L1/L2 hybrid switching capability, we insert a glue logic for parsing and multiplexing audio and data frames at MII(Media Independent Interlace) between the physical and data link layers. The proposed scheme can provide a good delay performance and transmission efficiency than legacy Ethernet based audio distribution systems. For verifying the feasibility of the proposed L1/L2 hybrid switching scheme, we use OMNeT++ as a simulation tool with various parameters. From the simulation results, one can find that the proposed scheme can provides outstanding characteristics in terms of both jitter characteristic for audio traffic and transmission efficiency of data traffics.