• The KIPS Transactions:PartC
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• v.11C no.4
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• pp.539-544
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• 2004

This paper presents the implementation of the bluetooth headset and the audio gateway connected to the mobile Phone in the embedded environment. The bluetooth module includes the BC02 processor chip, the BCSP02 firmware and the bluelab software Including bluetooth protocol stack. The above components in the bluetooth module developed at CSR company are used as the development environment. The application program using API functions supported by bluelab is coded by C language and loaded on the flash ROM of the bluetooth module. The cail processing capacity measuring the call setup time and the clearing time between the audio gateway and the headset is considered as the performance parameter of the developed systems. As a call setup and clearing time between the audio gateway and the headset is about 88.8ms, the call processing capacity is about 11 calls per second. Therefore the performance result is satisfied in the aspect of the call processing time.

• Journal of Korea Multimedia Society
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• v.6 no.5
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• pp.816-823
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• 2003

In this paper, we propose a blind audio watermarking using the tonal characteristic. First, we explain the perceptional effect of tonal on the existed researches and shout the experimental result that tonal characteristic is more stable than other characteristics used in previous watermarking studies against several signal processing. On the base of the result, we propose the blind audio watermarking using the relation among the signals on the frequency domain which compose a tonal masker. To evaluate the sound quality of our watermarked audios, we used the SDG(Subjective Diff-Grades) and got the average SDG 0.27. This result says the watermarking using the perceptional effect of tonal is available from the viewpoint of non-perception. And we detected the watermark hits from the watermarked audios which were changed by several signal processing and the detection ratios with exception of the time shift processing were over 98%. About the time shift processing, we applied the new method that searched the most proper position on the time domain and then detected the watermark bits by the ratio of 90%.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.80-88
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• 1997

In this paper, the implementation of audio/data processor using ETRI DSP to support analog mode in digital/analog dual mode mobile phone is presented. Audio/data processor performs the wideband data processing, audio signal processing, demodulation function, and data rate conversion when it is operated in analog mode. These functions are programmed in assembly language, and then loaded to ETRI DSP together with vocoder program for the digital mode operation. This is a very efficient implementation of the dual mode cellular phone ASIC since the vocoder for the digital mode and audio/data processor for the analog mode are programmed together in the same hardware.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.605-607
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• 2004

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era made of home theater system and the digital audio broadcasting (DAB). It is impossible to analyze the stability of the digital audio amplifier, which is based on the PWM signal processing. To solve this problem, the digital audio amplifier is analyzed using variable structure control theory which is one of nonlinear system theories. Moreover, the magnitude and the frequency of ripple signal, which generated by hysteresis in the comparator, is obtained using describing function which is useful to represent the input-output relation of nonlinear system.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2101-2102
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• 2006

In this paper, a new digital audio watermarking algorithm is presented. The proposed algorithm embeds watermark into audio signal based on human auditory system (HAS). This algorithm is a blind audio watermarking method, which does not require any prior information during watermark extraction process. This algorithm finds watermarking position using time-domain masking effect. First we insert the watermark into wavelet domain, and then we use a back-propagation neural network (BPN) to learn the characteristics of relationship between the watermark and the watermarked audio. Due to the teaming and adaptive capabilities of the BPN, the false recovery of the watermark can be greatly reduced by the trained BPN. Experimental results show that the proposed method has good inaudibility and high robustness to common audio processing attacks.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.4
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• pp.385-391
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• 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.

• Broadcasting and Media Magazine
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• v.13 no.1
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• pp.82-96
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• 2008

3D Audio Technologies include whole signal processing steps from acquisition to reproduction through encoding and transmitting technologies. However, there is a certain difference on adapted technologies according to audio presentation environments, because the presentation environment is the last step to provide 3D audio th listeners. In this paper, we describe variable 3D audio technologies to adapt variable audio presentation environments for consuming music contents.

• Journal of KIISE:Computing Practices and Letters
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• v.5 no.4
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• pp.440-448
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• 1999

본 논문은 객체지향형 분산처리 환경 하에서 오디오나 비디오 등과 같은 실시간(real-time) 스트림(stream) 데이타를 처리하는 데 필요한 소프트웨어 기반구조를 설계하고 구현한 내용을 기술한다. 본 논문에서 제시한 DAViS(Distributed Object Framework supporting Audio/Video Streaming)는, 오디오/비디오 데이타의 처리와 관련된 여러 소프트웨어 구성요소들을 분산객체로 추상화하고, 그 객체들간의 제어정보 교환경로와 오디오/비디오 데이타 전송경로를 서로 분리하여 처리한다. 분산응용프로그램 작성자는 DAViS에서 제공하는 서비스들을 이용하여, 기존의 분산프로그래밍 환경이 제공하는 것과 동일한 수준에서 오디오/비디오 데이타에 대한 처리를 표현할 수 있다. DAViS는, 새로운 형식의 오디오/비디오 데이타를 처리하는 부분을 손쉽게 통합하고, 하부 네트워크의 전송기술이나 컴퓨터시스템 관련 기술의 진보를 신속하고 자연스럽게 수용할 수 있도록 하는 유연한 구조를 가지고 있다. Abstract This paper describes the design and implementation of software framework which supports the processing of real-time stream data like audio and video in distributed object-oriented computing environment. DAViS(Distributed Object Framework supporting Audio/Video Streaming), proposed in this paper, abstracts software components concerning the processing of audio/video data as distributed objects and separates the transmission path of data between them from that of control information. Based on DAViS, distributed applications can be written in the same abstract level as is provided by the existing distributed environment in handling audio/video data. DAViS has a flexible internal structure enough to easily incorporate new types of audio/video data and to rapidly accommodate the progress of underlying network and computer system technology with very little modifications.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.367-374
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• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.