• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 하계종합학술대회 논문집(2)
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• pp.399-402
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• 2004

This paper presents a system-on-chip prototype implementing a full integration for a portable digital audio system. The chip is composed of a audio processor block to implements audio decoding and voice compression or decompression software, a system control block including 8-bit MCU core and Memory Management Unit (MMU) a low power 16-bit ${\Sigma}{\Delta}$ CODEC, two DC-to-BC converter, and a flash memory controller. In order to support other audio algorithms except Mask ROM type's fixed codes, a novel 16-bit fixed-point DSP core with the program-download architecture is proposed. Funker, an efficient power management technique such as task-based clock management is implemented to reduce power consumption for portable application. The proposed chip has been fabricated with a 4 metal 0.25um CMOS technology and the chip area is about 7.1 mm ${\times}$ 7.1mm with 100mW power dissipation at 2.5V power supply.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅲ
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• pp.1419-1422
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• 2003

Multimedia contents what we call Digital Items include various types of resources such as music, speech, text, video, graphics, and so on. The current Adaptation QoS described in the ISO/IEC 21000-7 CD-Part 7: Digital Item Adaptation, however, lacks adaptation methods for audio signals. The goal of this paper is to provide adaptation methods that are necessary to deal with audio signals. Two operations are introduced in order to adapt audio items. One method is to make use of the functionality of Fine Grain Scalability, and the other is intended to drop the channel of audio output channel. This paper provides a DIA description tool that associates the operators with the corresponding values of the constraint and the utility. Furthermore, the operations are evaluated and compared to alternative solutions.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권2호
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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.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1995년도 하계학술대회 논문집 B
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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.

• 방송공학회논문지
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• 제6권1호
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• pp.13-20
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• 2001

본 논문에서는 켑스트럼 영역에서의 오디오 워터마킹 방법을 제안한다. 광대역 채널에 협대역 신호를 숨기는 대역확산(spread spectrum) 통신과 유사한 방법을 이용하여 워터마크 신호를 오디오 신호의 켑스트럼 성분에 삽입한다. 의사잡음 시퀸스 (pseudo-random sequence)를 이용하여 오디오 신호에 삽입할 워터마크 신호를 발생시킨다 켑스트럼 계수들의 분포와 인간 청각 시스템(Human Auditory System, HAS)의 주파수 마스킹 특성을 고려하여 결정한 가중치를 곱해서 얻은 워터마크 신호는 사람 의 귀로 감지할 수 없을 정도의 크기를 갖는다. 본 논문의 실험 결과에서 보인 것처럼. 오디오 신호의 사용 권한이 없는 사용자들 은 삽입된 워터마크 신호를 검출하기가 통계적으로 거의 불가능하며, 다중 워터마크, MPEG 오디오 부호화. 부가잡음과 같은 다양한 공격에도 소유자에 의해서는 워터마크 신호의 검출이 가능하다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2000년도 정기총회 및 학술대회
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• pp.43-48
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• 2000

본 논문에서는 디지털 오디오 방송 시스템으로서 사용하기 위한 AAC (MPEG-2 Advanced Audio Coding) 코덱 시스템의 개발에 관하여 기술한다. 인코더 및 디코더는 ETRI가 제안한 디지털 오디오 방송 시스템에 접속하기 위해 MPEG-2 (Moving Picture Experts Group phase 2) 시스템의 TS(Transport Stream) 형식으로 입/출력한다. 내부 오디오 신호처리를 위한 DSP(Digital Signal Processor)로서는 Tl(Texas Instruments) 사의 TMS320C6701 (Floating point, 166 MHz)을 사용하였다. 인코더에서는 DSP 를 4 개까지, 디코더에서는 3 개까지 사용하여 구성할 수 있도록 설계하였다. DSP 에서는 시스템 제어, 오디오 신호 입력, 오디오 신호 처리, TS 신호 발생, 비트스트림 출력 등의 처리를 수행하며, 각 DSP 는 직렬 및 병렬 접속에 의해 서로 데이터를 전달한다. 본 시스템은 이후 다양한 디지털 방송 분야에 활용될 것이다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 컴퓨터소사이어티 추계학술대회논문집
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• pp.243-246
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• 2003

In this paper, we propose a new digital audio watermarking technique with the wavelet transform. The watermark is embedded by eliminating unnecessary information of audio signal based on human auditory system (HAS). This algorithm is an audio watermarking method, which does not require any original audio information in watermark extraction process. In this paper, the masking effect is used for audio watermarking, that is, post-tempera] masking effect. We construct the window with the synchronization signal and we extract the best frame in the window by using the zero-crossing rate (ZCR) and the energy of the audio signal. The watermark may be extracted by using the correlation of the watermark signal and the portion of the frame. Experimental results show good robustness against MPEG1-layer3 compression and other common signal processing manipulations. All the attacks are made after the D/A/D conversion.

• 방송공학회논문지
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• 제11권2호
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• pp.181-190
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• 2006

멀티채널 오디오 콘텐츠에 대한 사용자의 욕구가 증가함에 따라 해당 서비스를 위한 기술로써 압축 효율을 최대화 하는 저비트율 멀티채널 오디오 부호화 기술이 발전해야 한다. 이러한 요구에 부응하기 위하여 MPEG에서는 MPEG Surround 라는 이름으로 표준화가 진행 중이다. 본 논문에서는 MPEG Surround의 표준화 진행 현황을 소개하고 그 기술을 분석한다.

• JSTS:Journal of Semiconductor Technology and Science
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• 제14권1호
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• pp.109-116
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• 2014

This paper presents a high-efficiency digital class-D audio amplifier using a composite interpolation filter for portable audio devices. The proposed audio amplifier is composed of an interpolation filter, a delta-sigma modulator, and a class-D output stage. To reduce power consumption, the designed interpolation filter has an optimized composite structure that uses a direct-form symmetric and Lagrange FIR filters. Compared to the filters with homogeneous structures, the hardware cost and complexity are reduced by about half by the optimization. The coefficients of the digital delta-sigma modulator are also optimized for low power consumption. The class-D output stage has gate driver circuits to reduce shoot-through current. The implemented class-D audio amplifier exhibited a high efficiency of 87.8 % with an output power of 57 mW at a load impedance of $16{\Omega}$ and a power supply voltage of 1.8 V. An outstanding signal-to-noise ratio of 90 dB and a total harmonic distortion plus noise of 0.03 % are achieved for a single-tone input signal with a frequency of 1 kHz.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.110-113
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• 2000

In this paper, we consider the problem of digital audio watermarking to robust about compression without original audio data. We specifically address the audio watermarking using BPSK with variable carrier frequency. This technique make audio data embeded watermarking robust with compression attack, for example MPEG, AC-3, etc.