• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.599-609
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• 2009

An overview of the recently finalized multi-channel audio coding standard MPEG Surround is provided. This audio coding standard downmixes multi-channel signals to mono or stereo signals and, simultaneously, extracts spatial parameters for its encoding process. In its decoding process, it reconstructs multi-channel signals based on the downmix signals and spatial parameters. Since the downmix signals are coded in conventional audio coding format such as AAC and MP3 and the spatial parameters require a small amount of information MPEG Surround guarantees high sound quality multi-channel audio at low bit rates. Besides, it is backward-compatible to conventional audio coding techniques because the downmix signals can be played on portable audio devices ignoring the spatial parameter information. In this paper, Part 1 presents an overview of the basic structure of MPEG Surround and Part 2 describes various modes and tools including the binaural mode which supports the virtual 5.1-channel playback via headphones or earphones. The listening test results by various companies and organizations are also presented.

• The Journal of the Acoustical Society of Korea
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• v.32 no.1
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• pp.56-63
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• 2013

This study surveyed MP3 player usage and evaluated its effect on hearing thresholds in high school students. It is reported an increase in risk factors for hearing loss in adolescents that paralleled the increase in sales of MP3 players. A total of 538 students in two high schools in Gwangju province participated in this study. The students completed questionnaires about their MP3 player use and their pure tone hearing thresholds were tested at several frequencies. The questionnaire results indicated that 61.1%, of the students listened to music on MP3 players for more than one hour per day, 71.6% listened for four or more days per week, and 36.7% had listened for more than 3 years. Listening at a high volume of 60% of the maximum level was reported by 51% and they recognized having ear pains. The pure tone hearing threshold was found to be raised for the adolescents who had a long history of MP3 listening at high volume (p< 0.0001).

• Proceedings of the Korea Information Processing Society Conference
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• 2001.10b
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• pp.965-968
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• 2001

최근 초고속 통신망을 이용한 인터넷의 대중화와 더불어 인터넷을 기반으로 하는 전자상거래가 활발해지고 있다. 특히 인터넷을 통한 MP3 데이터 등의 멀티미디어 컨텐츠의 유통은 많은 연구의 대상이 되고 있다. 하지만 기존의 AOD(Audio On Demand) 시스템들은 실질적인 불법복제방지와 저작권 보호에 미흡한 단점이 있다. 따라서 본 논문에서는 에이전트를 이용하여 실질적인 불법복제방지와 저작권을 보호하는 AOD 시스템을 제안한다. 제안된 기법은 사용자의 시스템에 에이전트를 설치하여 정식 사용자를 인증하고 해당 사용자키로 암호화되어 전송된 MP3 데이터를 재생함으로써 전송 도중 제 3 자로부터의 공격에 대응할 수 있고, 정당한 사용자 외에는 MP3 데이터를 사용하지 못하도록 함으로써 사용자와 판매자의 권리를 보장한다.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.12
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• pp.716-722
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• 2005

In this paper, we propose the audio watermarking algorithm based on the critical band of HAS(human auditory system) without audibly affecting the quality of the watermarked audio and implement the detecting algorithm on the BSS(broadcast synchronizing system) for testing the proposed algorithm. According to the audio quality test, the SNR(signal to noise ratio) of the watermarked audio objectively is 66dB above. In the robustness test, the proposed algorithm can detect the watermark more than $90\%$ from various compression(MP3, AAC), A/D and D/A conversions, sampling rate conversions and especially asynchronizing attacks. The BSS automatically switches the programs between the key station and the local station in broadcasting system. The result of reliability test of implemented system by using the real broadcasting audio has no false positive error during 30 days. Because of detecting once processing per 0.5 second, we can judge that the false positive error does not occur.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2132-2135
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• 2003

본 논문에서는 고정소수점 DSP로 구현한 실시간 MP3 오디오 부호화기에 사용되는 초월함수용 하드웨어 가속기 구조를 제안한다. 구현된 하드웨어 가속기는 MP3 부호화 성능을 저하시키는 초월함수 연산오차에 강인하도록 설계되었다. 제안된 가속기의 연산오차는 Q1.23 고정소수점 출력에서 2비트, 즉 2/sup -21/ 까지의 연산오차를 가진다. LAME 부호화기[5]심리음향 모델의 SMR 오차는 테이블 보간법[4]을 사용할 경우에 비해 4dB이상 향상되었으며, 연산량은 총 4 MIPS 감소하였다. 제안한 하드웨어 가속기는 Verilog HDL로 기술되었으며, SYNOPSYS에서 0.18㎛ CMOS 표준 셀 라이브러리 공정으로 합성되었다. 합성 면적은 7514 게이트이며 초월함수 연산에 대한 동작속도는 3 사이클이다.

• Proceedings of the IEEK Conference
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• 2000.06c
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• pp.93-96
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• 2000

Currently, the most important technology is the compression methods in the multimedia society. Audio files are rapidly propagated through internet. MP-3 is offered to CD tone quality in 128Kbps, but 64Kbps below tone quality is abruptly down and high bitrate. 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 better than MP-3. Especially, AAC has max. 7.1 channel and 96KHz sampling rate. In this paper, the perceptual model is dealt with 44.1KHz sampling rate for SMR(Signal to Masking Ratio)

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.269-275
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• 2003

In this paper, we proposed the robust watermark diminishing distortion of the original data inserting the watermark in specific coefficients of the frequency domain. In case the alpha is more than 0.5. we found that proposed watermark is detected by experiment of MP3, FFT, Cropping and Echo attack. Our proposed method improved the Cox's method in the SNR aspect.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.708-711
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• 1999

MDCT( Modified Discrete Cosine Transform ) is one of the most compute-intensive operations in the MPEG audio coding standard. In this paper a fast algorithm to perform MDCT operation is presented. The algorithm presented in the MPEG audio coding standard requires (N/2) $\times$ N multiplications and (N/2) $\times$ (N-1) additions to generate the result, but the algorithm presented in this paper requires (N/2) $\times$ (N/2) multiplications and (N/2) $\times$ (N/2) additions to perform the same task. In this algorithm N should be multiple of 4. The algorithm was implemented using ARM processor and the processing time comparison between the original algorithm and the fast algorithm is presented.

• Proceedings of the Korea Multimedia Society Conference
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• 2004.05a
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• pp.191-194
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• 2004

본 논문에서는 무선망 개방을 맞이하여 각 이동통신사별로 상이한 오디오 형식들을 새로 제작하지 않고 자동 변환을 통해 기존 컨텐츠들을 그대로 재사용할 수 있는 오디오 변환 모듈에 대하여 연구하였다 본 연구에서 mid에서 mmf로의 변환, way에서 mmf로의 변환은 Yamaha사 툴들을 사용하였고, wav와 mp3간의 변환 작업은 Directshow 기술을 사용하였다. 본 연구 결과로 이동통신사간 상이한 벨소리의 자동 변환이 가능하게 되었고, 일반 오디오 형식인 way나 mp3의 mmf 변환이 가능하게 되었다.

• Journal of KIISE:Computing Practices and Letters
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• v.12 no.2
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• pp.102-110
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• 2006

This paper explains the design and development of the stand-alone high sound quality Internet Radio system, which is aimed for a small embedded type audio device rather than a general PC type. This device is designed to work with an Internet connection. This kind of system is not standardized so far, and also the related algorithm is not open to the public. So it is necessary to analyze several receiving algorithms of current radio receivers, and develop our own hardware in order to overcome these obstacles, finally to get the high quality of sound radio. The main electronic components of this Internet Radio are TCP/IP interfaces, an audio MP3 decoder, an I/O interface, and a Flash Memory Card with advanced audio multicasting for the next-generation Internet Radio. Basic structures and implementation issues of the next-generation most-versatile digital music player, and Internet Radio receivers, are discussed.