• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.612-621
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• 2003

As the use of MP3 audio compression increased, the demand for the insertion of additive data about copyright or information on music contents has been groved and the related research has been progressed actively. When an additive data is inserted into MP3 bitstream, it should not to happen any distortion of music quality or the change of file size, due to the modification of MP3 bitstream structure. In our study, to make these conditions satisfied, we inserted some additive data to bitstream by modifying some bits of linbits among the quantized integer coefficients having big values. At this time, we consider the characteristics of linbits and their distributions. As a result of subjective sound quality test through MOS test, we confirmed that the quality of MOS 4.6 can be achieved at the data insertion rate of 60 bytes/sec. Using the proposed method, it is possible to effectively insert an additive data such as copyright information or information about media itself, so that various applications like audio database management can be realized.

• 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.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.5
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• pp.41-49
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• 2010

Being popularized the use of portable entertainment/information devices, the demand on flash memory has been also increased radically. In general, flash memory reveals various error patterns by the devices it is mounted, and thus the memory makers are trying to minimize error ratio in the final process through not only the electric test but also the data integrity test under the same condition as real application devices. This process is called an application-level memory test. Though currently various flash memory testing devices have been used in the production lines, most of the works related to memory test depend on the sensual abilities of human testers. In case of testing the flash memory for MP3 devices, the human testers are checking if the memory has some errors by hearing the audio played on the memory testing device. The memory testing process like this has become a bottleneck in the flash memory production line. In this paper, we propose an audio comparison technique to support the efficient flash memory test for MP3 devices. The technique proposed in this paper compares the variance change rate between the source binary file and the decoded analog signal and checks automatically if the memory errors are occurred or not.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.251-254
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• 2001

본 논문에서는 ETRI와 연세대가 공동 개발한 AAC/MP3 복합 복호화기 ASIC 칩을 이용한 AAC/MP3 오디오 플레이어의 설계 및 구현에 대해 기술한다. 본 논문에서 사용한 AAC/MP3 복합 복호화 ASIC Chip은 20비트 고정소수점 DSP 코어를 이용하여 MP3와 MPEG-2 AAC LC 프로파일을 복호화하며, MPEG-2 AAC 메인 프로파일을 실시간으로 복호화하기 위하여 허프만 복호화 과정과 예측 과정은 전용 하드웨어 모듈을 이용하였다 이를 이용한 오디오 플레이어는 AAC/MP3 파일 재생 기능, USB를 이용한 호스트 PC와의 인터페이스 기능, Flash 메모리와의 인터페이스 기능 등의 특성을 갖는다.

• 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.

• Journal of the Institute of Convergence Signal Processing
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• v.19 no.2
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• pp.47-53
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• 2018

This paper proposes a method to express an MP3 audio file by a series of color QR codes which can be printed on the paper. Moreover, the method can compress the data considerably. Firstly, an MP3 file is divided into many small files which have maximum capacity of binary file of a QR code. Secondly, the multiple files are converted to multiple black-and-white QR codes. Lastly, every three QR codes are combined into color QR codes. When combining, each of three black-and-white QR codes are regarded as red, green, blue components respectively. In this method, the areas of a color QR code where two QR codes are overlapped are expressed by the colors Cyan, Magenta and Yellow. And the areas where three components are overlapped are expressed by white color. Contrarily, the areas that no components are overlapped are expressed by white color. Experimentation result shows that an MP3 file with 8.5MB the original MP3 files are compressed with the compression rate around 15.7. This method has the advantage that can be used in the environments that the internet access is impossible.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04a
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• pp.213-216
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• 2001

본 논문은 심리음향모델과 주파수변환을 이용하여 MP3 와 AAC 의 압축에서 강인하게 살아남을 수 있는 디지털 오디오 워터마킹 알고리즘에 관한 것이다. 워터마크를 의사난수열이나 이미지 등 외부 정보를 이용하지 않고 원본음악으로부터 생성시킨다는 것이 본 논문의 가장 큰 특징으로 원본 오디오로부터 생성된 워터마크는 음악과 융합되어 워터마크의 삽입여부를 일반인의 청각으로는 인식할 수 없다.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.54-60
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• 2004

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio, current digital audio is always restricted by sampling rate and bandwidth. This restriction normally results in low sample rate audio or calls for the data compression scheme such as MP3. However, they can only reproduce a lower frequency range than a regular CD quality because of the Nyquist sampling theory. Consequently they lose rich spatial information embedded in high frequency. The propose of this paper is to propose efficient high frequency enhancement of low sample rate audio using n adaptive filtering and DWT analysis and synthesis. The proposed algorithm uses the LMS adaptive algorithm to estimate the missing high frequency contents in DWT domain and it then reconstructs the spectrally enhanced audio by using the DWT synthesis procedure. Several experiments with real speech and audio are performed and compared with other algorithm. From the experimental results of spectrogram and sonic test, we confirm that the proposed algorithm outperforms the other algorithm and reasonably works well for the most of audio cases.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.338-341
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• 2003

최근의 인터넷 음악방송은 MP3 오디오를 기반으로 하는 TCP 프로토콜의 전송방식이 일반적이다. TCP방식의 전송은 HTTP 프로토콜을 이용한 파일 전송방식으로 네트워크의 부하가 급증할 경우 TCP의 특성으로 인해 음악의 끊김 현상이 발생하여 QoS 문제가 발생한다. 본 논문은 실시간 전송방식의 RTP(Real-time Transfer Protocol) 프로토콜을 이용하여 MP3 오디오 기반의 생방송 시스템 개발에 대한 연구로서, 기존의 TCP 방식의 음악의 끊김 현상을 개선하기 위한 모듈 구현을 목적으로 한다. 본 연구에서는 MP3 오디오 전송에 따른 QoS(Quality Of Service) 개선을 위하여 인터리빙 기법을 이용한 SPH/RPH(Send Payload Handler/ Receive Payload Handler) 모듈을 구현하였다.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.