• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.3 s.309
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• pp.76-83
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• 2006

A main objective of audio equalizer is for user to tailor acoustic frequency response to increase sound comfort and example applications of audio equalizer includes large-scale audio system to portable audio such as mobile MP3 player. Up to now, all the audio equalizer requires manual setting to equalize frequency bands to create suitable sound quality for each genre of music. In this paper, we propose an intelligent audio graphic equalizer system that automatically classifies the music genre using music content analysis and then the music sound is boosted with the given frequency gains according to the classified musical genre when playback. In order to reproduce comfort sound, the musical genre is determined based on two-step hierarchical algorithm - coarse-level and fine-level classification. It can prevent annoying sound reproduction due to the sudden change of the equalizer gains at the beginning of the music playback. Each stage of the music classification experiments shows at least 80% of success with complete genre classification and equalizer operation within 2 sec. Simple S/W graphical user interface of 3-band automatic equalizer is implemented using visual C on personal computer.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.161-164
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• 2000

본 논문에서는 최근 휴대용 오디오 기기 등에서 활발하게 사용되고 있는 MP3 (MPEG-1,2 계충-III) 오디오 복호화 알고리듬을 실시간 구현하였다. 휴대용 기기에 적합한 저전력 설계를 위하여 16비트 고정 소수점 범용 DSP인 모토로라 DSP56654를 이용하였고, 연산량을 줄이기 위한 작업을 수행하였다. 또한 음질 열화를 최소화하고 CD 수준의 고음질을 얻기 위해서 각 복호화 과정에 대한 최적의 고정소수점 연산을 연구하였다. 구현된 복호화기는 약 40MIPS 정도의 연산량으로 90dB이상의 SNR을 갖는 최종 PCM 샘플을 생성한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.06a
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• pp.381-384
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• 2013

본 논문에서는 허밍기반 음원 검색 시스템을 위한 오디오 특징 시퀀스 데이터 색인 기법을 제안한다. 우선 Query-by-Singing/Humming (QbSH) 시스템의 특징 데이터베이스를 생성하기 위하여 MP3 와 같은 다성음원에서 주요 멜로디를 추출하여 시퀀스데이터를 생성하고, 고속 검색을 지원하기 위한 시퀀스데이터를 색인화한다. 본 논문에서는 최소 Dynamic Time Warping (DTW) 거리 기법, 시퀀스 추상화 기법, 상한 값 기반 DTW 기법과 같이 세 가지의 시퀀스 데이터의 색인화 기술을 제시하고 각각에 대한 문제점을 파악하고, 성능을 평가한다. 이를 통하여 향상된 검색 시간과 검색 정확도를 얻을 수 있다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2002.12a
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• pp.309-312
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• 2002

본 논문에서는 오디오 워터 마킹 기법을 제안한다. 이 방법은 5차 웨이브릿 변환을 이용한sync 탐지 기법을 제안한다. 이 원리를 Zuicker의 인간청각모델의 한계 밴드이론을 이용한다. 그리고 워터마킹 검출에는 정점 탐지 기법에서 많이 이용하는 에너지와 제로통과 비율을 이용하여 워터마크를 검출한다 실험을 통하여 본 알고리즘이 mp3압축에 강인할 뿐 아니라 디지털에서 아날로그신호로 바꾸고 다시 디지털 신호로 바꾸는 아날로그 공격에 시간영역이나 DCT영역에서 워터마킹을 행하는 것보다 본 알고리즘이 강인함을 보인다 본 오디오 알고리즘은 음악에 연동하는 전기기기를 구성할 때 유용한 알고리즘이 될 수 있다. 즉 음악에 워터마크를 삽입하여 이 워터마크를 전기기기 동작제어 비트열로 이용할 수 있을 것이다.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04a
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• pp.453-456
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• 2001

21세기에 새로운 분야로 대두되고 있는 가상현실은 많은 사람들의 흥미를 끌고 있다. 상상 속에서나 가능하던 일들을 현실감과 입체감을 통해 실제처럼 느낄 수 있게 해준다는 것이 가상현실의 가장 큰 매력일 것이다. 가상현실을 요하는 멀티미디어 기기들의 활발한 시장진출로 3D 효과를 가진 오디오/비디오의 하드웨어 구현이 불가피하다. 본 연구에서는 휴대용 기기들에서 실시간 3D 입체음향 효과를 얻을 수 있도록 하드웨어를 구성하였다. 기존의 입체음향 기술에서 사용되는 콘볼루션 방법은 계산량이 많기 때문에 실시간 구현이 어렵다. 그러나 제안된 방식은 FFT를 사용하여 주파수 영역에서 처리함으로써 계산량을 줄여 하나의 프로세서로도 실시간 처리가 가능하도록 하였다. 저가/저전력/소형화조건을 요구하는 휴대용 기기에서 3D 입체 음향 효과를 얻을 수 있는 것이다. DSP는 TI(Texas Instruments)사의 16비트 고정소수점(fixed-point) 프로세서인 TMS320C5416을 사용한다. 구현된 3D 입체음향 칩은 입체음향을 필요로 하는 휴대용 MP3 Player, 가전용 오디오/비디오 등에 응용될 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.10
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• pp.23-32
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• 2007

This paper proposes a reconfigurable parallel processor architecture which can efficiently implement various multimedia applications, such as 3D graphics, H.264/H.263/MPEG-4, JPEG/JPEG2000, and MP3. The proposed architecture directly connects memories and processors so that memory access time and power consumption are reduced. It supports floating-point operations needed in the geometry stage of 3D graphics. It adopts partitioned SIMD to reduce hardware costs. Conditional execution of instructions is used for easy development of parallel algorithms.

• Journal of Intelligence and Information Systems
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• v.24 no.2
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• pp.21-35
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• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1C
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• pp.12-18
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• 2008

This paper This paper proposes a new inverse-quantization(IQ) table interpolation algorithm, specialized Digital Signal Processor(DSP) instructions and hardware architecture for digital audio codecs. Non-linear inverse quantization algorithm is representatively used in both MPEG-1 Layer-3 and MPEG-2/4 Advanced Audio Coding(AAC). The proposed instructions are optimized for the non-linear inverse quantization. The proposed algorithm can minimize operational complexity which reduces total computational load. Performance comparisons show a significant improvement of average error. The proposed instructions and hardware architecture can reduce 20% of the instruction counts and minimize computational loads of IQ algorithms effectively compared with existing IQ table interpolation algorithms. Proposed algorithm can implement commercial DSPs.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.