• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.70-75
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• 2007

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

• The KIPS Transactions:PartB
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• v.9B no.1
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• pp.67-76
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• 2002

MP3 audio format has good quality and high compression rate ; therefore, the use of MP3 format increases. The requirement of keeping the extra information such as author and lyrics in MP3 files increases. And the tagging systems designed to meet this requirement are suggested. ID3 vl tag and Lyrics3 v2 tag are two most widely used tagging systems. But ID3 vl tag and Lyrics3 v2 tag are the last things to arrive when the file is being streamed. Therefore, users cannot get the tag information until the entire audio file is downloaded. Moreover information synchronized with audio stream may lose its feature. In this paper, a system searching and playing audio files based on tag information in MP3 files is implemented. This system solves the problem that the tag information is ignored when an MP3 files is played on internet. An audio object is described in an XML document, and timing and synchronization between elements in that In document is provided in HTML+TIME style using XSL.

• Proceedings of the KAIS Fall Conference
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• 2000.10a
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• pp.194-198
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• 2000

최근 저장 장치로 많이 사용되고 있는 IBM PC의 CD-ROM(Compact Disc Read Only Memory)과 MP3(MPEG-1 Layer Ⅲ Audio) 디코더 칩을 이용하여 PC 작동과 관계없이 독립적으로 작동하는 CD-ROM 기반의 MP3 Player를 구현하였다. 여기서는 CD-ROM의 규격과 CD-ROM에 사용되는 ATAPI(AT Attachment Packet Interface) Format, MPEG-1 Layer Ⅲ의 오디오 부분에 대하여 알아보고 MP3 디코더 칩을 사용하여 CD-ROM에서 읽은 MP3 데이터즐 재생하는 방법을 취하였다. 이리하여 PC 자체로 MP3를 작동시키는 부하를 경감시키는 효과를 얻었다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.662-675
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• 2018

Compression history detection plays an important role in digital multimedia forensics. Most existing works, however, mainly focus on digital image and video. Additionally, the existed audio compression detection algorithms aim to detect the trace of double compression. In real forgery scenario, multiple compression is more likely to happen. In this paper, we proposed a detection algorithm to reveal the compression history for MP3 audio. The statistics of the scale factor and Huffman table index which are the parameters of MP3 codec have been extracted as the detecting features. The experimental results have shown that the proposed method can effectively identify whether the testing audio has been previously treated with single/double/triple compression.

• The Journal of the Acoustical Society of Korea
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• v.24 no.8
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• pp.492-497
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• 2005

In this paper, we suggest that the new volume of robust audio watermarking method protects form the MP3 compression attack, mostly used in audio field. Watermarks were inserted at the coefficients repeatedly from low frequencies to high frequencies after DCT transform in commonly used Cox's spread spectrum method. However, our method could be investigated losing frequency when MP3 attacks and make order by small amount of losing frequency and then insert watermarks rightly through statistical analysis. In the result about various sound clips, the suggested method have better performances than Cox's method by preserving watermarks and reducing distortions of the original sounds.

• Journal of Digital Contents Society
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• v.15 no.3
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• pp.357-363
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• 2014

In this paper, we propose a new audio forensic marking system for protecting the copyright of the Internet-based music streaming services. In the proposed method, in order to guarantee the QoS of the streaming service, high speed, and generates a forensic mark inserted MP3 file. We make pre-marking process and generate a new forensic marked MP3 file, a combination of the pre-marked MP3 frame, the inserted user information. Experimental results show that the proposed method satisfactory results robustness and imperceptibility, and real-time properties. In addition, we were confirmed that the real-time embedding and detection from the streaming-based audio forensic marking system that has been implemented on the server/client is possible.

• Proceedings of the Korean Information Science Society Conference
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• 1999.10c
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• pp.709-711
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• 1999

오늘날에 이르러 이동(mobile) 컴퓨팅의 중요성은 점차로 증가하는 추세에 있다. 하지만 이를 지원하기 위한 어플리케이션의 개발은 미진한 상태에 머물러 있으며, 특히 이동 컴퓨터를 위한 멀티미디어 서비스는 몇몇 기술적인 문제로 인하여 거의 제공도지 못하고 있는 것이 현실이다. 본 논문에서는 무선 LAN으로 구성된 이동 네트워크상에서 실시간 오디오 서비스를 제공하기 위한 MP3 Audio Decoder를 설계, 구현함으로써 이동 컴퓨팅 응용의 폭을 넓힌다. 그리고 이를 위한 몇 가지 고려사항과 관련 기술들을 서술하고, 구현된 프로그램의 실행 결과를 제시한다.

• Proceedings of the IEEK Conference
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• summer
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• pp.233-236
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• 2004

This paper proposes new method for the enhancement of nonexistent high frequency spectral contents from low sample rate audio signal. For example, Due to the protocol constraint, the audio bandwidth of MP3 is restricted to 16Khz. Although band-restricted MP3 audio provide savings of storage space and network bandwidth, it suffers a major problem of a loss in high frequency fidelity such as localization, ambient information, and bright nature of audio. This paper provides a new mathematical analysis for the adaptive estimation of the high frequency contents based on the nature of the input low sample rate audio. Proposed method can be worked globally to any kind of audio such as speech and music that are restricted by sampling rate and bandwidth.

• 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.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.958-966
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• 2007

In this paper, we implement a sound-enhanced MPEG-1 audio decoder on embedded OS Platform. Low bit rate lossy audio codecs such as MP3, OGG, and AAC for mitigating the problems in storage space and network bandwidth suffer a major common problem such as a loss of high frequency fidelity of audio signal. This high frequency loss will reproduce only a band-limited low-frequency part of audio in the standard CD-quality audio. In order to overcome this problem, we embedded a sound enhancement algorithm into the MPEG-1 audio decoder and then the algorithms optimized according to the characteristic of the MPEG-1 audio layer I, II, III were implemented on an embedded OS platform. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed system compared to the standard MPEG-1 audio decoder.