• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.55-60
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• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.912-918
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• 1994

Poor speech intelligibility in an air traffic control room is frequently a result of many, quite different causes and occasionally leads to complaints of the controller personnel. The paper describes a sequence of successful tasks performed in a local control room. The initial measurements included an investigation of the background noise (caused by fans, air condition, computer and radar equipment) and performance checks of the electronic audio and communication equipment with respect to the audio transmission behavior. The spectral composition of the noise as well as the characteristics of the audio communication path between the controllers and the pilots(which showed a loss of spectral information in the audio band due to built-in notch filters for the suppression of control tones) required adaptations of the amplitude behavior of the amplifiers through user adjustable tone controls. The radar console fans, which contributed significantly to the overall noise floor of the room, underwent a substantial reconstruction by replacing the tight mounting with an elastic double suspension, reducing the noise level by 50%. Finally, a possible source of untimely fatigue of the controllers during their working hours has been found in strong spectral components of the noise above the audio band, radiated by numerous video monitors in the control through vibrating components excited by the line frequency of the video signal.

• 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.

• ETRI Journal
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• v.29 no.1
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• pp.99-102
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• 2007

The channel level difference (CLD) is a main parameter in the reference model 0 (RM0) for MPEG Surround. Nevertheless, the CLD quantization method in the RM0 has problems such as the lack of theoretical background and inappropriate quantization levels. In this letter, a new CLD quantization method is proposed based on the virtual source location information which has strength in the quantization process. From experimental results, it is confirmed that the proposed scheme greatly reduces the quantization distortions measured in dB and degrees without any additional complexity.

• Journal of Internet Computing and Services
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• v.18 no.1
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• pp.57-64
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• 2017

Recent automobile manufacturing technology has improved not only the function and performance of cars, but also the audio systems in cars so as to increase their marketability. Automobile manufacturers always have the option of simply installing an expensive acoustic system to help customers enjoy a high-level sound quality car audio system. However, this also tends to increase the MSRP (Manufacturer's Suggested Retail Price) of the car. Therefore, it is desirable, where possible, to enhance the sound quality of plainer, less expensive audio devices to help customers feel as if they have a high-quality and expensive audio device in their car. In order to make this happen, the manufacturer must develop an optimal interior environment and audio system at a relatively lower cost. To this end, features of the car audio system can be enhanced by analyzing audio frequency response and using performance metrics to figure out the characteristics of the human auditory system. This study analyzed the sound field of Korean Recreation Vehicles (RVs) using the Internet of Things (IoT) sensor for the measurement of car audio system. As a result, high energy of sensitive bandwidth, one of the human auditory characteristics often makes annoying sound. This study also found that increasing the frequency response flatness is required by taking human auditory field into account when designing the car audio system for the future.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.865-873
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• 2004

MPEG-2 Advanced Audio Coding(AAC) is widely used in the multi-channel audio compression standards. And it combines hi인-resolution filter bank prediction techniques, and Huffman coding algorithm to achieve the broadcast-quality audio level at very low data rates. The forward and inverse modified discrete transforms which are operated in the encoder and the decoder of the filter bank need many computations. In this paper, we propose suitable recursive structure at IMDCT processing for MPEG-2 AAC real-time decoder. We confirm the memory, the computation speed and complexity of the proposed structure.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.28 no.11
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• pp.748-751
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• 2015

Sound Masking System technology as by sound the same on all bands and artificially generates a constant sound shield People want to hear or recognize the people with the noise generated from the interior of the way. Prevent hearing or prevent recognition by using the technology to control the audible frequency band Continue to emit constant and uniform shielding sound audible frequency band Even the security content of speech (20 Hz~20 KHz). That interception laser eavesdropping, internal solicitations, during recording Or delay the decoding was a result of the effect of interference calculated Experience noise disturbance index is applied around the Stress Index is the average index is 10.16 was a luxury for the average index is then applied to the index 3.07 Noise is significantly lower stress level has improved noise conditions.

• JSTS:Journal of Semiconductor Technology and Science
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• v.10 no.4
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• pp.276-281
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• 2010

A 2.5 V feed-forward second-order deltasigma modulator for audio application is presented. A 9-level quantizer with a tree-structured dynamic element matching (DEM) was employed to improve the linearity by shaping the distortion resulted from the capacitor mismatch of the feedback digital-toanalog converter (DAC). A chopper stabilization technique (CHS) is used to reduce the flicker noise in the first integrator. The prototype delta-sigma analogto-digital converter (ADC) implemented in a 65 nm 1P8M CMOS process occupies 0.747 $mm^2$ and achieves 109.1 dB dynamic range (DR), 85.4 dB signal-to-noise ratio (SNR) in a 24 kHz audio signal bandwidth, while consuming 14.75 mW from a 2.5 V supply.

• Journal of KIISE:Computing Practices and Letters
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• v.9 no.4
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• pp.470-485
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• 2003

The steganography is one of methods that users can hide data. Some steganography softwares use audio data among multimedia data. However, these commercialized audio steganography softwares have disadvantages that the existence of hidden messages can or easily recognized visually and only certain-sized data can be hidden. To solve these problems, this study suggested, designed and implemented Dynamic Message Embedding (DME) algorithm. Also, to improve the security level of the secret message, the file encryption algorithm has been applied. Through these, StegoWaveK system that performs audio steganography was designed and implemented. Then, the suggested system and the commercialized audio steganography system were compared and analyzed on criteria of the Human Visilable System (HVS), Human Auditory System (HAS), Statistical Analysis (SA), and Audio Measurement (AM).

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.201-204
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• 2007

In this paper, we research the content-based analysis and classification according to the composition of the feature parameters pool for the auditory signals to implement the auditory indexing and searching system. Auditory data is classified to the primitive various auditory types. we described the analysis and feature extraction method for the feature parameters available to the auditory data classification. And we compose the feature parameters pool in the indexing group unit, then compare and analysis the auditory data centering around the including level and indexing criterion into the audio categories. Based on this result, we composit feature vectors of audio data according to the classification categories, then experiment the classification using discrimination function.