• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.209-212
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• 2003

본 논문에서는 PC 환경에서 지상파 디지털 멀티미디어 방송(Digital Multimedia Broadcasting, DMB)을 수신할 수 있는 PC 기반 지상파 DMB 수신기용 백엔드 시스템 개발에 대해 서술한다. 지상파 DMB는 기존의 지상파 아날로그 또는 디지털 TV에 비해 탁월한 이동 수신 성능을 보인다. 본 논문에서는 국내 지상파 DMB 표준안에 부합하는 수신기의 백엔드 (back-end)를 PC 환경에서 소프트웨어로 구현하였다. 지상파 DMB는 유럽의 디지털 오디오 방송(Digital Audio Broadcasting, DAB) 표준인 EUREKA-147을 기반으로 MPEG-4 표준에 의한 멀티미디어 서비스를 제공한다. 지상파 DMB의 멀티미디어 서비스는 MPEG-4 AVC(Advance Video Coding) 압축 비디오와 BSAC(Bit Slice Arithmetic Coding) 압축 오디오를 MPEG-4 시스템의 SL(Sync Layer) 표준으로 패킷화 후 MPEG-2 TS(Transport Stream)에 실어 DAB의 스티림 모드를 통해 전송하는 방식을 사용한다. 본 논문에서는, 지상파 DMB 수신을 위한 프론트엔드(front-end)는 외장형 기기를 이용하고, 이로부터 USB 인터페이스를 통해 기저대역 다중화 스트림을 PC 상으로 업로드한 뒤, 소프트웨어에 의해 역다중화하고 압축을 푼 후, 오디오와 비디오를 재생하는 지상파 DMB 백엔드 시스템을 구현하고 이를 검증하였다.

• Journal of Broadcast Engineering
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• v.10 no.1 s.26
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• pp.22-30
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• 2005

We implement an interactive 3D audio broadcasting terminal which synthesizes an audio scene according to the request of a user. Audio scene structure is described by the MPEG-4 AudioBIFS specifications. The user updates scene attributes and the terminal synthesizes the corresponding sound images in the 3D space. The terminal supports the MPEG-4 Audio top nodes and some visual nodes. Instead of using sensor nodes and route elements, we predefine node type-specific user interfaces to support BIFS commands for field replacement. We employ sound spatialization, directivity/shape modeling, and reverberation effects for 3D audio rendering and realistic feedback to user inputs. We also introduce a virtual concert program as an application scenario of the interactive broadcasting terminal.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.589-598
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• 2009

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

• Journal of Broadcast Engineering
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• v.6 no.3
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• pp.215-224
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• 2001

In recent days, research activities on multimedia services mainly focus on the multiplexing system with timing synchromization for media components, such as video, audio and text. The MPEG-4 standard emphasizes object-based coding which includes analysis and understanding of the Image content. Since in MPEG-4 we can define objects and encode them independently, we can manipulate and display each object for different applications. This feature of MPEG-4 is also vero useful for multimedia services, such as video streaming cia different network channels, digital versatile disc, internet TV, video E-mail, and so on. In this Paper, we implement a multimedia streaming system which is compliant with the MPEG-4 system and the MP4 file format.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2019.05a
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• pp.456-458
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• 2019

In this paper, transmission performance of streaming audio as a railway communication service based on LTE-R is analyzed. Performance analysis is perfomed with computer simulation based on NS(Network Simulation)-3, audio frame of MPEG(Moving Picture Experts Group)-4 is used as target application service for straming audio traffic. Results of this paper can be used to implement LTE-R networks and develope application services over LTE-R network.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.3
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• pp.87-95
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• 2010

Portable multi-media products which can service the highest audio-quality by using lossless audio codec has been released and the international lossless codecs, MPEG-4 audio lossless coding(ALS) and MPEG-4 scalable lossless coding(SLS), were standardized by MPEG in 2006. The simple profile of MPEG-4 ALS, it supports up to stereo, was defined by MPEG in 2009. The lossless audio codec should have low-complexity in stereo to be widely used in portable multi-media products. But the previous researches of MPEG-4 ALS have focused on an improvement of compression ratio, a complexity reduction in multi-channels coding, and a selection of linear prediction coefficients(LPCs) order. In this paper, the complexity and compression ratio of MPEG-4 ALS encoder is analyzed in simple profile of MPEG-4 ALS, the method to reduce a complexity of MPEG-4 ALS encoder is proposed. Based on an analysis of complexity of MPEG-4 ALS encoder, the complexity of short-term prediction filter of MPEG-4 ALS encoder is reduced by using the low-complexity filter that is proposed in previous research to reduce the complexity of MPEG-4 ALS decoder. Also, we propose a joint coding decision method, it reduces the complexity and keeps the compression ratio of MPEG-4 ALS encoder. In proposed method, the operation of joint coding is decided based on the relation between cross-correlation of residual and compression ratio of joint coding. The performance of MPEG-4 ALS encoder that has the method and low-complexity filter is evaluated by using the MPEG-4 ALS conformance test file and normal music files. The complexity of MPEG-4 ALS encoder is reduced by about 24% by comparing with MPEG-4 ALS reference encoder, while the compression ratio by the proposed method is comparable to MPEG-4 ALS reference encoder.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.592-602
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• 2003

This paper aims to describe the basic structure of novel object-based 3D audio broadcasting system To overcome current uni-directional audio broadcasting services, the object-based 3D audio broadcasting system is designed for providing the ability to interact with important audio objects as well as realistic 3D effects based on the MPEG-4 standard. The system is composed of 6 sub-modules. The audio input module collects the background sound object, which is recored by 3D microphone, and audio objects, which are recorded by monaural microphone or extracted through source separation method. The sound scene authoring module edits the 3D information of audio objects such as acoustical characteristics, location, directivity and etc. It also defines the final sound scene with a 3D background sound, which is intended to be delievered to a receiving terminal by producer. The encoder module encodes scene descriptors and audio objects for effective transmission. The decoder module extracts scene descriptors and audio objects from decoding received bistreams. The sound scene composition module reconstructs the 3D sound scene with scene descriptors and audio objects. The 3D sound renderer module maximizes the 3D sound effects through adapting the final sound to the listner's acoustical environments. It also receives the user's controls on audio objects and sends them to the scene composition module for changing the sound scene.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.554-561
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• 2004

As the technologies of semiconductor and multimedia communication have been improved. the high-quality video and the multi-channel audio have been highlighted. MPEG Audio Layer 3 decoder has been implemented as a Processor using a standard. Since the synthesis filter of MPEG-1 Audio Layer 3 decoder requires the most outstanding operation in the entire decoder. the synthesis filter that can reduce the amount of operation is needed for the design of the high-speed processor. Therefore, in this paper, the synthesis filter. the most important part of MPEG Audio, is materialized in FPGA using the method of DAULT (distributed arithemetic look-up table). For the design of high-speed synthesis filter, the DAULT method is used instead of a multiplier and a Pipeline structure is used. The Performance improvement by 30% is obtained by additionally making the result of multiplication of data with cosine function into the table. All hardware design of this Paper are described using VHDL (VHIC Hardware Description Language) Active-HDL 6.1 of ALDEC is used for VHDL simulation and Synplify Pro 7.2V is used for Model-sim and synthesis. The corresponding library is materialized by XC4013E and XC4020EX. XC4052XL of XILINX and XACT M1.4 is used for P&R tool. The materialized processor operates from 20MHz to 70MHz.

• The Magazine of the IEIE
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• v.28 no.4
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• pp.103-104
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• 2001

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.1 no.2
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• pp.91-98
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• 2008

Domestic standard for Korean T-DMB includes MPEG-4 BSAC (Bit Sliced Arithmetic Coding) audio coding that has been established in 2003. This paper presents an implementation and optimization of MPEG-4 BSAC Audio Decoder on ARM926EJ-S processor. Tools and modules of the BSAC audio decoder were implemented with 32-bit fixed point operations. Further optimization was accomplished using ARM926EJ-S Inline Assembly. The optimization was based on the total number of multiplications and MAC (Multiply and Accumulation) operations causing most of core cycles of ARM926EJ-S, and also based on analysis of ARMv5 instructions. The result of optimization was evaluated on the basis of MIPS (Million Instruction per second). Implementation results show that BSAC bitstream at 96kbps can be decoded in real-time at 65MHz CPU clocks.