• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2004년도 학술대회지
• /
• pp.631-635
• /
• 2004

The rapid development of digital media and communication network urgently brings about the need of data certification technology to protect IPR (Intellectual property right). This paper proposed a new watermarking method for embedding contents owner's audio signal in order to protect color image IPR. Since this method evolves the existing static model and embeds audio signal of big data, it has the advantage of restoring signal transformed due to attacks. Three basic stages of watermarking include: 1) Encode analogue ID owner's audio signal using PCM and create new 3D audio watermark; 2) Interleave 3D audio watermark by linear bit expansion and 3) Transform Y signal of color image into wavelet and embed interleaved audio watermark in the low frequency band on the transform domain. The results demonstrated that the audio signal embedding in color image proposed in this paper enhanced robustness against lossy JPEG compression, standard image compression and image cropping and rotation which remove a part of image.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2008년도 하계종합학술대회
• /
• pp.875-876
• /
• 2008

Parametric stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus small amount of parametric overhead to describe the stereo image. The stereo properties are analyzed, encoded, and reinstated in a decoder according to spatial psycho-acoustical principles. However, coding of multichannel audio signal using parametric stereo still requires considerable bit-rate. In this paper, enhanced parametric stereo coding for multichannel audio signal is proposed.

• Transactions on Electrical and Electronic Materials
• /
• 제9권3호
• /
• pp.123-127
• /
• 2008

The paper is to solve the problem of existing audio signal transfer system which has a difficulties of system organization and the increase of additional install cost and unfriendly interior. To solve the existing system, we drew the new audio signal transfer system based on PLC and evaluated it. A transmitter and a receiver were designed using the PLC chip INT5500CS. An audio signal transfer system was configured with a CD player to which audio signals are sent from the transmitter and a speaker connected to the receiver. For performance evaluation of this system, a USBPre external sound card and Smaart Live 5 which is a PC-based sound measuring program were added. As a result of our experiment, the measured signal level is $2{\sim}3$ dB lower than reference signal, latency is 16.69 ms, and the specific character of coherency is bad in high frequency band. Otherwise, this system transmits and receives signals over 90 % in good condition as a result of measuring pink noise, frequency (1 kHz), and phase, magnitude. In view of the result so far achieved, the system designed this study has excellent performance, it resolves defect of existing audio signal transfer system.

• International Journal of Internet, Broadcasting and Communication
• /
• 제16권2호
• /
• pp.10-21
• /
• 2024

Recently, audio systems are changing the configuration of conventional sound reinforcement (SR) systems and public address (PA) systems by using audio over IP (AoIP), a technology that can transmit and receive audio signals based on internet protocol (IP). With the advancement of IP technology, AoIP technologies are leading the audio market and various technologies are being released. In particular, audio networks and control hierarchy over peer-to-peer (Anchor) technology based on AoIP is a system that transmits and receives audio signals over a wide bandwidth without an audio mixer, creating a novel paradigm for existing audio system configurations. Anchor technology forms an audio system by connecting audio sources and output equipment with On-site audio center (OAC), a device that can transmit and receive IP. Anchor's receiving OAC is capable of receiving and mixing audio signals transmitted from different IPs, making it possible to configure a novel audio system by replacing the conventional audio mixer. However, Anchor technology does not have the ability to provide audio effects to input devices such as microphones and instruments in the audio system configuration. Due to this, when individual control of each audio source is required, there is a problem of not being able to control the input signal, and it is impossible to individually affect a specific input signal. In this paper, we implemented a tone control module that can individually control the tone of the audio source of the input device using the audio processor core in the audio system based on Anchor technology, tone control for audio sources is possible through a tone control module connected to the transmitting OAC. As a result of the study, we confirmed that OAC receives the signal from the audio source, adjusts the tone and outputs it on the tone control module. Based on this, it was possible to solve problems that occurred in Anchor technology through transmitting OAC and tone control modules. In the future, we hope that the audio system configuration using Anchor technology will become established as the standard for audio equipment.

• 제어로봇시스템학회논문지
• /
• 제13권4호
• /
• pp.385-391
• /
• 2007

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era. Since the digital audio amplifier is based on the PWM signal processing, it is improper to analyze the principle of signal generation using linear system theories. In this paper, a class D digital audio amplifier based ADSM (Advanced Delta-Sigma Modulation) is considered. We first model the digital audio amplifier and then explain the operation principle using variable structure control algorithm. Moreover, the ripple signal generated by the hysteresis in the comparator has a significant effect on the system performance. Thus, we present a method to find the magnitude and the frequency of the ripple signal using describing function. Finally, simulations and experiments are provided to show the validity of the proposed methods.

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

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2000년도 하계종합학술대회 논문집(3)
• /
• pp.125-128
• /
• 2000

In this paper, we proposed the high quality digital audio watermarking algorithm in the frequency domain. The spread spectrum technique is used to encrypted a stream of information by spreading the data as much of the frequency spectrum as possible. It's technique adapt well to data hiding in audio signal. We have used the perceptual model and MDCT/IMDCT for the high qualify digital audio watermarking. The proposed watermark algorithm preserved high quality audio data from watermark signal.

• 한국음향학회지
• /
• 제24권4호
• /
• pp.202-209
• /
• 2005

디지털 미디어와 통신 네트워크의 급속한 발전으로 지적소유권 보호 기술이 절실히 요구된다. 본 논문에서 영상컨텐츠에 소유자의 음성신호를 삽입할 수 있는 새로운 영상 워터마킹 방법을 제안한다. 제안한 방법은 삽입할 워터마크로 음성신호를 이용하기 때문에 청각적으로 소유권을 주장하는데 매우 유리하다 그리고 LBX (Linear Bit expansion)에 의해 인터리빙하여 음성 워터마크를 삽입하기 때문에 공격에 의해 변형되거나 특히 제거된 음성신호를 복구할 수 있는 이점을 가진다. 본 방법은 다음세가지 기본 절차를 포함한다. 첫째, 아날그 소유자음성 신호를 PCM에 의해 부호화하고 새로운 디지털 워터마크를 생성한다. 둘째, 제안한 LBX에 의해 음성 워터마크를 인터리빙한다. 마지막으로 영상 컨텐츠를 이산 Haar 웨이브렛변환 (DHWT) 하고 저주파 영역을 마킹공간으로 하여 인터리빙 된 음성워터마크를 삽입한다. 실험 결과 본 연구에서 제안한 소유자 음성신호의 워터마크 삽입방법은 기존 워터마크 정보보다 눈에 보이지 않게 많은 정보량을 삽입할 수 있고 표준영상압축방식인 JPEG 손실압축과 특히 영상의 일부가 제거되는 영상절단과 영상회전 공격에 대해 강인성을 강건히 할 수 있었다.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2002년도 ITC-CSCC -2
• /
• pp.872-875
• /
• 2002

A digital watermarking technique applying psychoacoustic model for audio signal is proposed in this paper. In the watermarking scheme, the pseudo-random bit stream used as a watermark signal is embedded into the audio signal in both speech and music. The strength of the embedded signal is subject to the human auditory system in such a way that the disturbances on host audio signal are beyond the sensing of human ears. The experimental results show that the quality of the watermarked audio signal, in term of signal to noise ratio, can be improved up to 3.2 dB.

• 한국음향학회지
• /
• 제24권7호
• /
• pp.387-394
• /
• 2005

본 논문에서는 다채널 오디오 시스템의 스피커들을 직렬로 연결하기 위한 새로운 오디오 신호 전송 기법을 제시한다. 다채널 오디오 본체로부터의 아날로그 신호는 디지털 신호로 변환되고 신호 처리 과정을 거쳐서 직렬로 연결된 각 스피커에 전달된다. 여기서 신호 처리 과정은 오디오 신호의 특성을 고려한 데이터 압축과 전송을 위한 패킷 생성을 포함한다. 각 스피커는 전달된 패킷으로부터 해당하는 디지털 신호만을 검출하여 아날로그 신호로 다시 변환하여 음향을 재생한다. 제시된 모든 기능은 VHDL을 사용하여 모델링되었으며 FPGA 칩으로 구현하였고 실제 다채널 오디오 시스템에서 테스트하였다.