• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

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 contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th 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, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.610-617
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• 2009

An overview of various modes and tools of MPEG Surround is provided Because the binaural mode of MPEG Surround supports the virtual 5.1-channel playback based on HRTFs, it can be played via headphones and earphones for portable audio devices. MPEG Surround also supports the enhanced matrix mode which converts stereo signals to 5.1-channel signals without side information, the 3D stereo mode which deals with 3D-coded signals, the low power version which greatly reduces the computational load in the decoding process. Besides, MPEG Surround provides the arbitrary downmix gains (ADGs) tool which is applied to artistic downmix signals, the matrix compatibility tool which is applied to downmix signals by conventional matrix-based methods, the residual coding tool -which can be used at high bit rates, and the GES tool which is applied to specific sound such as applause. The listening test results by various companies and organizations are also presented for important modes and tools.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.4 s.334
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• pp.61-68
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• 2005

A chip, which can fast encoder the audio data to AAC (Advanced Audio Coding) LC(Low Complexity) that is MPEG-2 audio standard, has been designed on the basis of a 32 bits DSP core and fabricated with 0.25um CMOS technology. At first, the various optimization methods for implementing the algerian are devised to reduce the memory size and calculation cycles. FFT(Fast Fourier Transform) hardware block is added to the DSP core to get the more reduction of the calculation cycles. The chips has the size of $7.20\times7.20 mm^2$ and about 830,000 equivalent gates, can carry out AAC encoding under 70MIPS(Million Instructions per Second).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.12C
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• pp.1251-1264
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• 2002

Currently, audio streaming services such as on-demand service and live service support only a limited number of clients. They also suffer from a lack of stability and degradation of service quality due to their inefficient use of network resources. Futhermore, since the streaming services usually do not consider dynamic services, they are very inconvenience to use. In order to resolve these drawbacks, we propose a novel audio streaming system based on MPEG-2 AAC file format which are facilitated with the network bandwidths efficiently. The proposed system supports QoS for audio streaming as well as guarantees a stability while streaming service is undergoing. Moreover, the system provides a dynamic interface which enables us to use the streaming service more easily and to manage streaming servers with convenient manner. On the contrary, most of the current available static interface streaming services are mainly depending only on a single fixed web page between client and server, which in consequence lead us to use unflexible static service environment. Our implementation results show the proposed system improves the performance compared to those of the currently existing systems that use MP3 file format. It also provides some benefits such as a stability of service and a easy to management of streaming servers.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.8
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• pp.1069-1082
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• 1993

The International Organization for Standardization(ISO) has undertaken an effort to develop a standard for video and associated audio on digital storage media. This effort is known by the name of the expert group that started if : MPEG-Moving Picture Experts Group Is currently part of the ISO-I EC/J TC1/SC2/WG11. The promise of MPEG-2 is that a video signal and its associated audio can be compressed to a bit rate of about 10 Mbits/s with an acceptable quality. In this paper, the implementation of a video compression simulator based on MPEG-2 Video Test Model 2(TM2) is described and analyzed according to the simulation results. The implemented simulator is also applied to code HDTV sequences at the several bit rates. Some computer simulation results using the MPEG and the HDTV test sequences are given. In addition, some techniques which can improve the coding efficiency of the implemented video compression simulator are also suggested.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.261-269
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• 2004

In this paper, an efficient algorithm for computing in the MPEG/audio Layer Ⅲ (MP3) encoder is proposed. The proposed algerian performs a full-band 1024-point FFT by computing 32-point FFT's of 32 subband outputs. To reduce the aliasing caused by the analysis filter bank, an aliasing cancellation butterfly is developed. A major benefit of the proposed algorithm is the computational saving. By using the proposed algorithm, it is possible to save 40~50% of computations for FFT, which results in about 20% reduction of the PAM-2 complexity.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.215-218
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• 2003

본 논문에서는 MPEG-7에 정의된 오디오 서술자를 이용한 오디오 특징을 기반으로 한 음악 검색 알고리즘을 제안한다. 특히 timbral 특징들은 음색 구분을 용이하게 할 수 있어 음악 검색뿐만 아니라 음악 장르 분류 또는 Query by humming에 이용 될 수 있다. 이러한 연구를 통하여 오디오 신호의 대표적인 특성을 표현 할 수 있는 특징벡터를 구성 할 수 있다면 추후에 멀티모달 시스템을 이용한 검색 알고리즘에도 오디오 특징으로 이용 될 수 있을 것이다 본 논문에서는 방송 시스템에 적용 할 수 있도록 검색 범위를 특정 컨텐츠의 O.S.T 앨범으로 제한하였다. 즉, 사용자가 임의로 선택한 부분적인 오디오 클립만을 이용하여 그 컨텐츠 전체의 O.S.T 앨범 내에서 음악을 검색할 수 있도록 하였다. 오디오 특징벡터를 구성하기 위한 MPEG-7 오디오 서술자의 조합 방법을 제안하고 distance 또는 ratio 계산 방식을 통해 성능 향상을 추구하였다. 또한 reference 음악의 템플릿 구성 방식의 변화를 통해 성능 향상을 추구하였다. Classifier로 k-NN 방식을 사용하여 성능 평가를 수행한 결과 timbral spectral feature들의 비율을 이용한 IFCR(Intra-Feature Component Ratio) 방식이 Euclidean distance 방식보다 우수한 성능을 보였다.

• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.637-646
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• 2002

In this paper, an implementation example of Java based MPEG-4 system that follows MPEG-4 standard protocols in order to provide multimedia-messaging service is introduced. The multimedia-messaging service is a wireless LAN based wired and wireless service that delivers multimedia contents including video and audio information. Detailed Methods to develop a MPEG-4 system like technology of MPEG-4 system implementation, definition for wired and wireless multimedia service, DMIF implementation, and mp4 file Parsing are described.