• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.171-173
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• 2012

본 논문에서는 최근 MPEG 오디오 서브그룹에서 활발히 논의 중인 3D Audio 표준화 동향에 대해서 소개하고, 관련한 국내외 기관들의 기술개발 현황에 대해서 알아본다. MPEG 3D Audio 는 NHK 22.2 채널방송과 같은 실감 오디오 서비스를 고다채널(High-Order Multichannel)로 특징짓고, 이러한 서비스를 위한 다채널 오디오 부호화 및 복호화 기술과 다양한 출력채널 환경에 적응할 수 있는 렌더링(rendering) 기술을 표준화 대상으로 규정하고 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1433-1442
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• 1998

The thesis presents a dual audio decoder which can decode both AC-3 and MPEG-2 bitstream. The MPEG-2 synthesis processi s optimized via FFT to establish the common data path with AC-'3s. A dual audio decoder consists of a DSP core which performs the control-intensive part of each algorithm and a common synthesis filter which perfomrs the computation-intensive part. All the components of the dual audio decoder have been described in VHDL and simulated with a SYNOPSYS tool. The software modeling of the DSP core was used for functional validation. After being synthesized using 0.6 .mu.m-3ML technology standard cell, the dual audio decoder was simulated at gate-level with a COMPASS tool for hardware validation.

• Proceedings of the KIEE Conference
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• 1995.07b
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• pp.977-979
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• 1995

MPEG-II is the international standard of compression for digital image and digital audio that is the most important in the multimedia environment. Now many researchers are developing relevant systems. MPEG-II consists of video, audio, system and the other part. In this paper, we have designed and demonstrated two channel audio encoder system that processes the audio compression part, and excutes layer II for complexity and psychoacoustic model II, with ADSP-21020 of Analog Device.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.4
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• pp.477-482
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• 2007

This paper presents an audio decoder architecture which can decode AC-3 and MPEG-2 audio bit-streams efficiently. MPEG-2 synthesis filtering is modified by the 32-point FFT to share the common data path with the AC-3's. A programmable Audio DSP core and a hardwired common synthesis tilter are incorporated for effective decoding of two different formats.

• 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.

• Journal of Korea Society of Industrial Information Systems
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• v.13 no.5
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• pp.117-123
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• 2008

In this paper, we develop MPEG-4 AAC streaming player on embedded Linux processor such as mobile terminals. Moreover we show the experimental results that the player preforms the decoding processes of MPEG-4 AAC data effectively. MPEG-4 AAC technology supports a wide range encoding rates and high sound quality so it is appropriate to adopt various applications. In particular, the need in the development of the application of audio data increases according to significantly increase in devices used in mobile environments, such as cell phones and PDAs. In this environment, it is necessary to optimize the decoding processes to the ability of the terminal hardware in order to play audio data without delays. We also implement the decoding module to optimize the processor capabilities and make the player to decode and play streaming audio data from streaming server.

• Convergence Security Journal
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• v.8 no.4
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• pp.91-96
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• 2008

An increasing in quantity of multimedia data brought a new problem that expected data should be retrieved fast and exactly. The adequate representation is a key element for the efficient retrieval. For this reason, MPEG-7 standard was established for description of multimedia data in 2001. In this paper, we designed a Audio/Image Retrieval System based on MPEG-7 that can retrieve multimedia data like audio, image efficiently. And we integrated high-level and low-level schemas to retrieve datas for users.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.12C
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• pp.1617-1622
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• 2004

MPEG/Audio decoder for mobile multimedia systems requires low power consumption. Implementations of AV decoder using a single RISC processor often need high power consumption owing to cash-miss in case of insufficient cash memory. In this paper, we present a MPEG/Audio decoder for mobile handset applications and implement it on a RISC processor embedding a minimized DSP accelerator. Audio decoding algorithm is splined into two parts; computation intensive and control intensive parts. Those parts we, respectively, allocated to DSP and RISC core, which are designed to run in parallel to increase the processing efficiency. The proposed system implements MP3 and AAC decoders at l7MHz and 24MHz clocks, which are reductions of 48% and 40% of complexities in comparison with implementations on a single RISC processor. The proposed method is adequate for mobile multimedia applications with insufficient cash memory.

• Proceedings of the IEEK Conference
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• 2000.11c
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• pp.97-100
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• 2000

The most important technology is the compression methods in the multimedia society. Audio files are rapidly propagated through internet. MP-3(MPEG-1 Layer3) is offered to CD tone quality in 128kbps, but 64kbps below tone-quality is abruptly down. On the other hand, MPEG-II AAC (Advanced Audio Coding) is not compatible with MPEG-I, but AAC has a high compression ratio 1.4 times better than MP-3 and it has max. 7.1 channel and 96KHz sampling rate. In this paper, we designed the optimized MDCT (Modified Discrete Cosine Transform) that could decrease the capacity of enormous computation and could increase the processing speed in the MPEG-2 AAC encoder.