• Proceedings of the Korean Information Science Society Conference
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• 1998.10c
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• pp.303-305
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• 1998

MPGE(Moving Picture Expert Group)은 디지털 동영상 압축 부호화의 표준으로 자리잡고 있으며 MPEG-1에 이어 현재는 MPEG-2가 상용화되어 있는 실정이다. 복잡한 압축 기법의 적용으로 이를 재생하기 위해서는 전용의 하드웨어가 필요했지만 CPU의 성능이 향상됨에 따라 소프트웨어적으로 구현이 가능하게 되었다. 본 논문에서는 Software MPEG-1 Playback System에서 가장 큰 문제가 되는 Audio와 Video간의 동기화에 관한 기법을 제시한다.

• ETRI Journal
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• v.31 no.6
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• pp.732-740
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• 2009

This paper describes the implementation of a digital audio effect system-on-a-chip (SoC), which integrates an embedded digital signal processor (DSP) core, audio codec intellectual property, a number of peripheral blocks, and various audio effect algorithms. The audio effect SoC is developed using a software and hardware co-design method. In the design of the SoC, the embedded DSP and some dedicated hardware blocks are developed as a hardware design, while the audio effect algorithms are realized using a software centric method. Most of the audio effect algorithms are implemented using a C code with primitive functions that run on the embedded DSP, while the equalization effect, which requires a large amount of computation, is implemented using a dedicated hardware block with high flexibility. For the optimized implementation of audio effects, we exploit the primitive functions of the embedded DSP compiler, which is a very efficient way to reduce the code size and computation. The audio effect SoC was fabricated using a 0.18 ${\mu}m$ CMOS process and evaluated successfully on a real-time test board.

• Journal of Korea Multimedia Society
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• v.8 no.6
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• pp.807-813
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• 2005

The production and authoring of digital multimedia contents are most important fields in multimedia technology. Nowadays web-based technology and related multimedia software technology are growing in the IT industry and these technologies are evolving most rapidly in our life. The technology of digital audio and video processing is utilizing rapidly to improve quality of our life, Also we are more interested in high sense and artistic feeling in the music and entertainment areas by use of three dimensional (3D) digital sound technology continuously as well as 3D digital video technology. The service field of digital audio contents is increasing rapidly through the Internet. And the society of Internet users wants the audio contents service with better quality. Recently Internet users are not satisfying the sound quality with 2 channels stereo but seeking the high quality of sound with 5,] channels such as 3D audio of the movie films. But it might be needed proper hardware equipments for the service of 3D sound to satisfy this demand. In this paper, we expand the simple 3D audio generator developed and propose a web-based music bank by the software development of 3D audio generation player in 3D sound environment with two speakers minimizing hardware equipments, Also we believe that this study would contribute greatly to digital 3D sound service of high quality for music and entertainment mania.

• ETRI Journal
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• v.22 no.1
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• pp.20-29
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• 2000

In this paper, we present a design of video and audio single chip encoder/decoder for portable multimedia application. The single-chip called as video audio signal processor (VASP) consists of a video signal processing block and an audio single processing block. This chip has mixed hardware/software architecture to combine performance and flexibility. We designed the chip by partitioning between video and audio block. The video signal processing block was designed to implement hardware solution of pixel input/output, full pixel motion estimation, half pixel motion estimation, discrete cosine transform, quantization, run length coding, host interface, and 16 bits RISC type internal controller. The audio signal processing block is implemented with software solution using a 16 bits fixed point DSP. This chip contains 142,300 gates, 22 Kbits FIFO, 107 kbits SRAM, and 556 kbits ROM, and the chip size is $9.02mm{\times}9.06mm$ which is fabricated using 0.5 micron 3-layer metal CMOS technology.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.63 no.6
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• pp.804-809
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• 2014

Recently, the prevention of crime using CCTV draws special in accordance with the higher crime incidence rate. Therefore security systems like a CCTV with audio capability are developing for giving an instant alarm. This paper proposes an analysis on screams and ambient noises for security systems with audio capability. This analysis result will be helpful for security systems to detect screams well with various ambient noises in real environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.4
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• pp.892-898
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• 2017

This paper describes the development of AVN(Audio Video Navigation) software for vehicle information analysis and gesture recognition. The module that examine the CAN(Controller Area Network) data of vehicle in the designed software analyzes the driving state. Using classified information, the AVN software converge vehicle information and hand gesture information. As the result, the derived data is used to match the service step and to perform the service. The designed AVN software was implemented in HW platform that common used in vehicles. And we confirmed the operation of vehicle analysing module and gesture recognition in a simulated environment that is similar with real world.

• Proceedings of the KSLP Conference
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• 1995.11a
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• pp.141-153
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• 1995

CSpeech is a software package that implements an audio waveform/speech analysis workstation on an IBM Personal Computer or hardware compatible computer. Features include digitizing audio waveforms on single or multiple channels, displaying the digitized waveforms, playing back audio waveforms from selected intervals of sing1e channels, saving and retrieving waveforms from binary format disk files, and analysing audio waveforms for their temporal and spectral properties. The distinguishing characteristics of CSpeech are its support for multiple channels, minimal restrictions on sample rate and waveform duration support fur a variety of hardware configurations, fast graphics display, and its user- extensible menu- based command structure.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.541-547
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• 2013

The speech quality of multi-party audio telephony between two, three or more participants is decreased by audio volume imbalance, audio volume saturation and noise level increase. To solve this issue, this paper proposes an advanced audio mixing algorithm for software-based multi-point control unit. Our approach is based on the combined voice activity detection and gain control technique that consists of a set of algorithms that classify audio signals, estimate audio volumes, adjust gain factors and mix audio signals of all channels. The proposed audio mixing algorithm is computationally efficient, delivers high-quality speech, and is suitable for use in any practical multi-party audio telephony.

• Proceedings of the KSLP Conference
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• 1995.11a
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• pp.154-164
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• 1995

CSL, Model 4300B is a highly flexible audio processing package designed to provide a wide variety of speech analysis operations for both new and sophisticated users. Operations include 1) Data acquisition 2) File management 3) Graphics 4) Numerical display 5) Audio output 6) Signal editing 7) A variety of analysis functions, External module include 1) Input control B) Output control 3) Jacks, Software include 1) Wide range of speech display manipulation 2) Editing 3) Analysis (omitted)

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.3
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• pp.307-312
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• 2018

In this paper, we can confirm the result of intuitive interface software implementation for operating a digital PA(Public Address) controller and the performance of audio mixer control part. Developed user interface software provides the maintaining management and control function of digital hybrid mixer. This SW loaded in the integrated control server controls an sound status of the audio mixer TAD-168M and checks the device status for Public Address integrated system. Also, this SW enables the integrated control and the continuous upgrade. Developed SW is connected to TAD-168M with Ethernet and linked to PC Lan port and the 4-port switch, located in the backside of TAD-168M, by LAN cable for communicating with operating PC. Integrated control including system management, audio control and uplink broadcasting control for broadcasting system will be made available with this novel developed system.