• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• The KIPS Transactions:PartB
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• v.14B no.4
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• pp.273-280
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• 2007

This paper presents a new 3D sound toolkit called $EVE-Sound^{TM}$ that consists of pre-processing tool for environment simplification preserving sound effect and 3D sound API for real-time rendering. It is designed so that it can allow users to interact with complex 3D virtual environments by audio-visual modalities. $EVE-Sound^{TM}$ toolkit would serve two different types of users: high-level programmers who need an easy-to-use sound API for developing realistic 3D audio-visually rendered applications, and the researchers in 3D sound field who need to experiment with or develop new algorithms while not wanting to re-write all the required code from scratch. An interactive virtual environment application is created with the sound engine constructed using $EVE-Sound^{TM}$ toolkit, and it shows the real-time audio-visual rendering performance and the applicability of proposed $EVE-Sound^{TM}$ for building interactive applications with complex 3D environments.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4
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• pp.151-158
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• 2006

In this paper, we proposed a new audio fingerprint extraction method, based on Philips' music retrieval algorithm, which uses the energy difference of neighboring filter-bank and probabilistic characteristics of music. Since Philips method uses too many filter-banks in limited frequency band, it may cause audio fingerprints to be highly sensitive to additive noises and to have too high correlation between neighboring bands. The proposed method improves robustness to noises by reducing the number of filter-banks while it maintains the discriminative power by representing the energy difference of bands with 2 bits where the quantization levels are determined by probabilistic characteristics. The correlation which exists among 4 different levels in 2 bits is not only utilized in similarity measurement. but also in efficient reduction of searching area. Experiments show that the proposed method is not only more robust to various environmental noises (street, department, car, office, and restaurant), but also takes less time for database search than Philips in the case where music is highly degraded.

• Proceedings of the Korean Institute of IIIuminating and Electrical Installation Engineers Conference
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• 2007.05a
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• pp.400-403
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• 2007

This research describes how the class D amplifiers with power efficiency are designed and implemented for the PA audio systems. The configuration that makes use of the class D amplifier properties depends strongly on their applications. Thus in this paper the characteristics of the 2-level and 3-level PWM are analysed and the circuit implementation for them is presented. Using the proposed methods, they are designed and simulated for the further investigation. Test(Simulation) results present the improved performance that shows the satisfactory operations in controlling the PWM to the input signals.

• Proceedings of the KSR Conference
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• 2011.05a
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• pp.1709-1715
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• 2011

A track-circuit device exploits two rails as a conductor of an electrical circuit. It detects the presence or absence of the trains with variation of received level when wheel sets of the train shunt rails together. So, it plays an important role in train operation. Non-insulated audio frequency(AF) track-circuit device can separate the rails into tracks by insulated joints without physically cutting the rails. And it transmits AF signals at track boundary, of which frequency is different between adjacent tracks. It has been employed at station-to-station section of domestic high speed rail, general rail, and subway. But It was not used at station yard of complicated general railway where lots of branches exist. In this paper, we consider current development status of non-insulated AF track-circuit device for the use of station yard.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.251-253
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• 2008

최근 유럽에서 DVB-T HDTV 방송 표준이 정하지면서 오디오 포맷으로 HE-AAC가 채택되었다. HE-AAC는 압축효율은 높지만 연산량이 높아 낮은 성능의 DSP에서 수행하기에는 어려움이 있다. DVB-T에서는 5.1채널을 사용하고 있어 더욱더 많은 연산을 필요로 한다. 본 논문은 ISO/DEC 14496-3 MPEG4 HE(High Efficiency)-AAC의 Level4에 해당하는 Multichannel Decoder를 최적화하여 구현하고. 가장 많은 연산을 필요로 하는 Synthesis Filter Bank에 제안된 알고리즘을 적용하여 연산량을 줄였고 대부분의 연산부를 어셈블리로 코드 최적화를 하여 작은 성능의 DSP를 사용하여 실시간 Multichannel HE-AAC Audio Decoder의 구현이 가능하게 하였다. DVB-T 오디오 시스템에 필수로 필요한 Audio Description, Dynamic Range Control, Downmix 등을 함께 구현하여 실제 수신기에 사용이 가능하도록 하였다. DSP는 Samsung의 CalmRISC16 + MAC24 core 를 사용하였다.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.459-461
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• 2004

This paper presents an implementation of real-time speaker localization using audio-visual information. Four channels of microphone signals are processed to detect vertical as well as horizontal speaker positions. At first short-time average magnitude difference function(AMDF) signals are used to determine whether the microphone signals are human voices or not. And then the orientation and distance information of the sound sources can be obtained through interaural time difference and interaual level differences. Finally visual information by a camera helps get finer tuning of the speaker orientation. Experimental results of the real-time localization system show that the performance improves to 99.6% compared to the rate of 88.8% when only the audio information is used.

• Journal of Computing Science and Engineering
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• v.6 no.1
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• pp.40-50
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• 2012

Increase in the number of older people due to demographic changes poses great challenges to the social healthcare systems both in the Western and as well as in the Eastern countries. Support for older people by formal care givers leads to enormous temporal and personal efforts. Therefore, one of the most important goals is to increase the efficiency and effectiveness of today's care. This can be achieved by the use of assistive technologies. These technologies are able to increase the safety of patients or to reduce the time needed for tasks that do not relate to direct interaction between the care giver and the patient. Motivated by this goal, this contribution focuses on applications of acoustic technologies to support users and care givers in ambient assisted living (AAL) scenarios. Acoustic sensors are small, unobtrusive and can be added to already existing care or living environments easily. The information gathered by the acoustic sensors can be analyzed to calculate the position of the user by localization and the context by detection and classification of acoustic events in the captured acoustic signal. By doing this, possibly dangerous situations like falls, screams or an increased amount of coughs can be detected and appropriate actions can be initialized by an intelligent autonomous system for the acoustic monitoring of older persons. The proposed system is able to reduce the false alarm rate compared to other existing and commercially available approaches that basically rely only on the acoustic level. This is due to the fact that it explicitly distinguishes between the various acoustic events and provides information on the type of emergency that has taken place. Furthermore, the position of the acoustic event can be determined as contextual information by the system that uses only the acoustic signal. By this, the position of the user is known even if she or he does not wear a localization device such as a radio-frequency identification (RFID) tag.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6A
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• pp.488-498
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• 2005

The DAB(Digital Audio Broadcasting) service which is based on the Eureka-147 of Europe is developed to DMB(Digital Multimedia Broadcasting) service that is divided into Terrestrial DMB and Satellite DMB. The Satellite DMB is a new broadcasting service, which will service multi-channel multimedia broadcasting by the portable receiver or the vehicle receiver. In this paper, we consider that link level performance of satellite DMB system A which is based on the COFDM(Coded Orthogonal Division Multiplexing). It uses the OFDM method which is sensitive to nonlinearity, so we analyze the effect of the HPA(High Power Amplifier) nonlinearity. And then we define the appropriate back-off value by performing the link level simulation considering back-off effect. Also we consider the effect of frequency and time offset, and then confirm the overall link level performance by analyzing and verifying a suitable synchronization method for satellite DMB system A.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.5
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• pp.41-49
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• 2010

Being popularized the use of portable entertainment/information devices, the demand on flash memory has been also increased radically. In general, flash memory reveals various error patterns by the devices it is mounted, and thus the memory makers are trying to minimize error ratio in the final process through not only the electric test but also the data integrity test under the same condition as real application devices. This process is called an application-level memory test. Though currently various flash memory testing devices have been used in the production lines, most of the works related to memory test depend on the sensual abilities of human testers. In case of testing the flash memory for MP3 devices, the human testers are checking if the memory has some errors by hearing the audio played on the memory testing device. The memory testing process like this has become a bottleneck in the flash memory production line. In this paper, we propose an audio comparison technique to support the efficient flash memory test for MP3 devices. The technique proposed in this paper compares the variance change rate between the source binary file and the decoded analog signal and checks automatically if the memory errors are occurred or not.