• International journal of advanced smart convergence
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• 제11권4호
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• pp.1-9
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• 2022

The recent development of information and communication technology is also affecting audio/video systems used in industry. The audio/video device configuration system changes from analog to digital, and the network-based audio/video system control has the advantage of reducing costs in accordance with system operation. However, audio/video systems released on the market have limitations in that they can only control their own products or can only be performed on specific platforms (Windows, Mac, Linux). This paper is a study on a device (Network Audio Video Integrated Control: NAVICS) that can integrate and control multiple audio / video devices with different functions, and can control digitalized audio / video devices through network and serial communication. As a result of the study, it was confirmed that individual control and integrated control were possible through the protocol provided by each audio/video device by NAVICS, and that even non-experts could easily control the audio/video system. In the future, it is expected that network-based audio/video integrated control technology will become the technical standard for complex audio/video system control.

• 음성과학
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• 제7권1호
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• pp.5-15
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• 2000

In this paper a detailed description of collecting Korean audio-video speech data is presented. The main aim of this experiment is to collect some audio-video materials which can be used for later experiments to estimate and model the actions of the visible human articulatory organs such as mouth, lips and jaw. We collect audio-video data from seven directions separately. Twelve markers are used to trace the movements.

• JSTS:Journal of Semiconductor Technology and Science
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• 제5권4호
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• pp.249-254
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• 2005

This paper presents a system-on-a-chip (SOC) design for digital TV. The single LSI incorporates almost all essential parts such as CPU, ISO/IEC 11172/13818 system/audio/video decoders, a video post-processor, a graphics/OSD processor and a display processor. It has analog IP's inside such as video DACs, an audio PLL, and a system PLL to reduce the system-level implementation cost. Descramblers and Smart Card interface are included to support widely used conditional access systems. The video decoder can decode two video streams simultaneously. The DSP-based audio decoder can process various audio coding specifications. The functional blocks for video quality enhancement also form outstanding features of this SoC. The SoC supports world-wide major DTV services including ATSC, ARIB, DVB, and DIRECTV.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2148-2151
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• 2003

In this paper, we present our experiences in designing and implementing a CX23880 based multichannel video/audio capture device. We try to clarify differences between CX2388x family and 878A, the previous version of Cx2388x, and what one needs to be careful about in developing device drivers for CX2388x based video/audio devices. Our work is expected to help one who will need to develop Cx2388x based video/audio device later.

• 인터넷정보학회논문지
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• 제21권4호
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• pp.51-57
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• 2020

Currently, High-efficient video coding (HEVC) has become the most promising video coding technology. However, the implementation of HEVC in video streaming systems is restricted by factors such as cost, design complexity, and compatibility with existing systems. While HEVC is considering deploying to various systems with different reached methods, H264/AVC can be one of the best choices for current video streaming systems. This paper presents an adaptive method for manipulating video streams using video coding on an integrated circuit (IC) designed with a private network processor. The proposed system allows to transfer multimedia data from cameras or other video sources to client. For this work, a series of video or audio packages from the video source are forwarded to the designed IC via HDMI cable, called Tx transmitter. The Tx processes input data into a real-time stream using its own protocol according to the Real-Time Transmission Protocol for both video and audio, then Tx transmits output packages to the video client though internet. The client includes hardware or software video/audio decoders to decode the received packages. Tx uses H264/AVC or HEVC video coding to encode video data, and its audio coding is PCM format. By handling the message exchanges between Tx and the client, the transmitted session can be set up quickly. Output results show that transmission's throughput can be achieved about 50 Mbps with approximately 80 msec latency.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• ETRI Journal
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• 제22권1호
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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.

• 전자공학회논문지CI
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• 제43권4호
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• pp.123-132
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• 2006

현재 가정에 널리 보급되어 있는 비디오/오디오 기기 중 비디오 레코더 플레이어는 영상 기기로 분류되고 있지만 DVD 플레이어의 경우에는 오디오 기기로 분류될 정도로 오디오의 기능과 성능을 강조하고 있다. 따라서 비디오/오디오 시스템에서 오디오 재생 기능과 성능을 높이는데 많은 연구가 이루어지고 있다. 본 논문에서는 DVD 전용 프로세서를 이용한 비디오/오디오 시스템을 구성하는데 있어서 기존에 비디오 및 오디오 데이터의 처리를 DVD 전용 프로세서가 전담하는 구조에서 발생되는 DVD 전용 프로세서의 시스템 리소스의 한계성으로 인한 기능의 제약을 개선한 오디오 기능을 분화한 비디오/오디오 시스템의 구조를 제안하고 시스템 구현을 통해 제안된 구조의 우수성을 증명한다. 본 논문에서 제안한 시스템은 오디오 기능의 분화로 여러 가지 오디오 포맷이 지원 가능할 뿐만 아니라 원음에 특수한 효과를 부여하여 재생할 수 있도록 함으로써 사용자 취향에 맞는 설정으로 오디오를 감상할 수 있도록 하였고 다양한 음장 모드를 지원하도록 하였다. 다양한 포맷의 오디오를 지원하기 위해서는 각 오디오 포맷에 맞는 디코더를 모두 채용하는 방법이 가장 좋은 방법이겠으나 이러한 방법은 시스템 구성상의 효율성이 떨어지게 된다. 따라서 오디오를 재생할 때 필요한 오디오 포맷만을 재생 가능하도록 하기 위해 오디오 데이터를 검출하고 검출된 포맷만을 지원하는 방식으로 오디오 시스템을 구현함으로써 시스템의 유연성을 개선하였다.

• IEIE Transactions on Smart Processing and Computing
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• 제2권1호
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• 한국정보과학회논문지:컴퓨팅의 실제 및 레터
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• 제5권4호
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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.