• 한국음향학회:학술대회논문집
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• 한국음향학회 2004년도 추계학술발표대회논문집 제23권 2호
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• pp.283-286
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• 2004

Due to advances in electronics very high quality audio reproduction is today possible. But the listening environment causes deviation of the audio system from the expected behavior. Firstly the listening Room significantly changes the audio signal frequencies and their phase. Secondly the position of the user in the room affects the perceived sound. With existing DSP technology it is possible to adequately correct these effects. In our work we developed a room correction system, correcting up to 7.1 channels using dual Motorola 56367 fixed point DSP's, implementing position dependent room effects measurement, real time compensation filter design and equalization filtering procedures.

• ETRI Journal
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• 제31권6호
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• pp.732-740
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• 2009

This paper describes the implementation of a digital audio effect system-on-a-chip (SoC), which integrates an embedded digital signal processor (DSP) core, audio codec intellectual property, a number of peripheral blocks, and various audio effect algorithms. The audio effect SoC is developed using a software and hardware co-design method. In the design of the SoC, the embedded DSP and some dedicated hardware blocks are developed as a hardware design, while the audio effect algorithms are realized using a software centric method. Most of the audio effect algorithms are implemented using a C code with primitive functions that run on the embedded DSP, while the equalization effect, which requires a large amount of computation, is implemented using a dedicated hardware block with high flexibility. For the optimized implementation of audio effects, we exploit the primitive functions of the embedded DSP compiler, which is a very efficient way to reduce the code size and computation. The audio effect SoC was fabricated using a 0.18 ${\mu}m$ CMOS process and evaluated successfully on a real-time test board.

• ETRI Journal
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• 제28권2호
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• pp.219-222
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• 2006

Virtual source location information (VSLI) has been newly utilized as a spatial cue for compact representation of multichannel audio. This information is represented as the azimuth of the virtual source vector. The superiority of VSLI is confirmed by comparison of the spectral distances, average bit rates, and subjective assessment with a conventional cue.

• 한국정보보호학회:학술대회논문집
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• 한국정보보호학회 2001년도 종합학술발표회논문집
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• pp.464-468
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• 2001

In this paper, we propose a robust watermarking technique that accepts time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC, WMA. The technique is developed based on digital filtering. Being designed according to critical band of HAS (human auditory system), the digital filters nearly affect audio quality. Furthermore, before implementing digital filtering, wavelet transform decomposes the audio signal into several signals that is composed of specific frequencies. Designed digital filters scan the decomposed signal. The designed digital filter, band-stop filter, distorts and eliminates specific frequencies of audio signals. Watermarking detection can be accomplished by FFT (Fast Fourier Transform). Firstly, segments of audio signal are transformed by FFT. Then, the obtained amplitude spectrum by FFT is summed repeatedly. Finally the watermark detector can find filters used to watermark encoding based on eliminating frequencies. The suggested technique can embed 4bits/s in a robust manner.

• 전자공학회논문지
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• 제49권11호
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• pp.149-156
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• 2012

디지털 입력 D급 증폭기는 보청기에서 사용되고 있으며 D급 증폭기는 디지털 회로와 아날로그 회로로 구성되어진다. 아날로그 회로는 가청 주파수 대역에서 잡음을 억제하고 디지털 입력을 아날로그 신호로 변환한다. 본 논문에서 제안한 인터폴레이티드 디지털 델타-시그마 변조기는 디지털 신호 처리기의 출력 신호를 D/A 변조기 입력에 적합하도록 데이터를 변조시킨다. 디지털 필터는 16-bit, 25-kbps 펄스 코드 변조 신호를 16-bit, 50-kbps 신호로 보간 작업을 한다. 이 보간 필터 출력은 3차 디지털 델타-시그마 변조기를 통하여 노이즈 쉐이핑(noise shaping) 처리된다. 최종적으로, 1.5-bit, 3.2-Mbps 신호가 D/A 변조기 입력으로 인가된다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 하계종합학술대회 논문집
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• pp.530-533
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• 1999

During the last decade, high quality digital audio has essentially replaced analog audio. During this period, digital audio have applied many application areas of the info-industry. These applications have created a demand for high quality digital audio. In audio compression, the methods using human auditory nervous properties are used and introduced from psychoacoustical model utilized perceptual audio coding unable to code above the limitation of human perception. The discussion concentrates on architectures and applications of those techniques which utilize psychoacoustical models to exploit efficiently masking characteristics of the human receiver. In this paper, the designed MDCT/IMBCT as a standard of current MPEG is implemented onto FPGA.

• Journal of Power Electronics
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• 제6권1호
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• pp.83-93
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• 2006

A new high efficiency and low profile on-board DC/DC converter for digital car audio amplifiers is proposed. The proposed converter shows low conduction loss due to the low voltage stress of the secondary diodes, a lack of DC magnetizing current for the transformer, and a lack of stored energy in the transformer. Moreover, since the primary MOSFETs are turned-on under zero-voltage-switching (ZVS) conditions and the secondary diodes are turned-off under zero-current-switching (ZCS) conditions, the proposed converter has minimized switching losses. In addition, the input filter can be minimized due to a continuous input current, and an output inductor is absent in the proposed converter. Therefore, the proposed converter has the desired features, high efficiency and low profile, for a viable power supply for digital car audio amplifiers. A 60W industrial sample of the proposed converter has been implemented for digital car audio amplifiers with a measured efficiency of $88.3\%$ at nominal input voltage.

• IEIE Transactions on Smart Processing and Computing
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• 제2권1호
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• 방송공학회논문지
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• 제6권1호
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• pp.3-12
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• 2001

디지털 오디오 워터마킹 기술은 최근 들어 많은 응용 분야에서 관심을 가지고 있는 새로운 연구분야이다. 디지털 방송의 경우도 컨텐츠에 대한 저작권 보호 필요성이 요구됨에 따라 오디오 워터마킹에 대한 관심이 고조되고 있다. 디지털 워터마킹이란 영상, 오디오 등과 같은 디지털 데이터에 보이거나 들리지 않는 정보를 은닉시키는 기술을 말한다. 대표적인 오디오 워터마킹 방법에는 대역확산 기반의 워터마킹, 반향 워터마킹, 위상 부호화 워터마킹 패치워크 워터마킹 등이 있으며, 계속해서 새로운 워터마킹 기법들이 개발되고 있다. 본 논문에서는 디지털 방송 컨텐츠의 보호를 위한 오디오 워터마킹의 적용 방법에 따른 기술적 요구사항을 알아보고, 현재 개발된 대표적인 오디오 워터마킹 방법들의 특징을 살펴본 뒤 몇 가지 항목에 대해 장단점을 비교 평가한다.

• 전자공학회논문지CI
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• 제43권5호
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• pp.48-56
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• 2006

본 논문에서는 심리음향 모델과 MDCT/IMDCT(Modified DCT/Inverse Modified DCT)를 이용하여 고음질 오디오 워터마킹 알고리즘을 제안하였다. 일반적으로 디지털 오디오 워터마크는 디지털 오디오 신호를 주파수 영역으로 변환 한 다음 주파수 영역에 워터마크를 삽입하지만 삽입된 워터마크에 의해 디지털 오디오 음질이 영향을 받게 된다. 제안된 알고리즘에서는 디지털 오디오 데이터를 PN(Pseudo Noise) 코드를 사용하여 확산시킨 다음 심리음향 모델을 참조하여 MDCT 과정을 통하여 오디오 워터마크를 삽입시킨다. MDCT 과정에서 고음질의 오디오를 얻기 위해 필터뱅크 출력의 첨예도에 따라 256, 1,024 또는 2,048 포인트의 윈도우가 선택되어진다. 본 논문에서 워터마크 계수 ${\alpha}$가 2.5 이하일 때, 워터마크의 검출률이 SDMI(Secure Digital Music Initiative)의 제안 조건을 50% 이상 상회 하며, SNR은 4종류의 공격(압축, 절단, FFT, 에코)에 대해 $50{\sim}68dB$ 값을 가짐을 확인하였다.