• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.387-394
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• 2005

In this paper, we propose a new transmission technique of audio signals for the serial connection of the speakers of multiple-channel audio systems. Analog audio signals from a multi-channel audio system are converted into digital signals with signal processing steps and transferred to each speaker through a serial line. The signal processing steps contain data compression and packet generation in association with audio signal characteristics. Each speaker gets its corresponding digital audio signals from the transmitted packets and converts the signals into analog audio signals to make sounds with the speaker All the proposed functions in this paper are modeled in VHDL. implemented with FPGA chips, and tested for actual multi-channel audio systems.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.592-602
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• 2003

This paper aims to describe the basic structure of novel object-based 3D audio broadcasting system To overcome current uni-directional audio broadcasting services, the object-based 3D audio broadcasting system is designed for providing the ability to interact with important audio objects as well as realistic 3D effects based on the MPEG-4 standard. The system is composed of 6 sub-modules. The audio input module collects the background sound object, which is recored by 3D microphone, and audio objects, which are recorded by monaural microphone or extracted through source separation method. The sound scene authoring module edits the 3D information of audio objects such as acoustical characteristics, location, directivity and etc. It also defines the final sound scene with a 3D background sound, which is intended to be delievered to a receiving terminal by producer. The encoder module encodes scene descriptors and audio objects for effective transmission. The decoder module extracts scene descriptors and audio objects from decoding received bistreams. The sound scene composition module reconstructs the 3D sound scene with scene descriptors and audio objects. The 3D sound renderer module maximizes the 3D sound effects through adapting the final sound to the listner's acoustical environments. It also receives the user's controls on audio objects and sends them to the scene composition module for changing the sound scene.

• ETRI Journal
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• v.36 no.6
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• pp.999-1007
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• 2014

The robustness of an audio fingerprinting system in an actual noisy environment is a major challenge for audio-based content identification. This paper proposes a high-performance audio fingerprint extraction method for use in portable consumer devices. In the proposed method, a salient audio peak-pair fingerprint, based on a modulated complex lapped transform, improves the accuracy of the audio fingerprinting system in actual noisy environments with low computational complexity. Experimental results confirm that the proposed method is quite robust in different noise conditions and achieves promising preliminary accuracy results.

• Journal of Advanced Navigation Technology
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• v.26 no.5
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• pp.358-364
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• 2022

The Inter Communication System for avionics is in charge of processing all voice signals that internal calls between Pilot and Co-pilot, internal calls between Pilots and Crews, external calls through communication equipment such as Ultra/Very High Frequency Receiver/Transmitter(U/VHF RT), audio signal monitoring for navigation and mission equipment such as VHF Omnidirectional Range/Instrument Landing System(VOR/ILS), Tactical Air Navigation(TACAN), audio signal output for voice recording to Flight Data Recorder(FDR) and Data Transfer System(DTS), and warning/caution audio signal generate about the status and threat of aircraft. Because Inter Communication System for avionics is sensitive to noise in the case of analog audio signals, a redundant design that can protect audio signal from electromagnetic noise inside/outside of aircraft is required for the mission of pilots and crews. In this paper, Normal/Back-up operation mode and redundancy design plan based on digital method for the redundancy of the digital Inter Communication System for avionics and manufacturing, verification results are described.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.186-193
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• 2005

The Terrestrial Digital Multimedia Broadcasting (T-DMB) system supplies high quality audio comparable with VCD in 7 inch display and high quality audio comparable CD at the mobile reception environment T-DMB will launch commercial service at the middle of 2005. However the bandwidth for audio data and the number of channels are restricted to 128 kbps and 2 respectively in the current T-DMB standard because of the limitation of available bandwidth for multimedia data. This Paper Proposes a novel media processor structure for providing multi-channel audio contents oyer T-DMB system allowing backward compatibility with the legacy T-DMB receiver. Furthermore. we also Propose an adaptive receiver structure to supply optimal audio contents on various speaker configuration in T-DMB receiver. To provide multi-channel audio contents allowing backward comaptilbity with the legacy T-DMB receiver, the additional data for multi-channel audio are defined as a dependent stream of main audio stream. The OD strucure for control an additional multi-channel audio elementary stream is proposed without changing the BIFS of the legacy T-DMB system.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.4 s.310
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• pp.123-132
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• 2006

In this paper, the digital video and audio system is improved so that various digital video data formats in DVD disc, and digital audio data formats through the S/PDIF ports may be decoded. It is not easy to implement all decoding functions of video and audio by a DVD processor. The special structure in audio decoding circuit is proposed in this system so as to have simultaneously almost same video and audio performance in quality. By dividing the decoding circuit separately into video and audio part, the audio quality can be dramatically improved together with supporting several audio formats and with several effects. In order to satisfy the perfect audio system to support to audio decoding formats, it is just enough to get the expensive, complicated decoder. However, it may be not easy to get this expensive decoder in near future. Therefore it is rather to adopt the downloading method by which the host should download the appropriate code into memory by detecting the corresponding audio bit streams. It is proved that this method may be efficient in the point of sharing resource of audio data for video decoding.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.91-97
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• 2004

In this paper, we propose a classification system of TV program scenes based on audio information. The system classifies the video scene into six categories of commercials, basketball games, football games, news reports, weather forecasts and music videos. Two type of audio feature set are extracted from each audio frame-timbral features and coefficient domain features which result in 58-dimensional feature vector. In order to reduce the computational complexity of the system, 58-dimensional feature set is further optimized to yield l0-dimensional features through Sequential Forward Selection (SFS) method. This down-sized feature set is finally used to train and classify the given TV program scenes using κ -NN, Gaussian pattern matching algorithm. The classification result of 91.6% reported here shows the promising performance of the video scene classification based on the audio information. Finally, the system stability problem corresponding to different query length is investigated.

• International journal of advanced smart convergence
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• v.6 no.2
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• pp.51-58
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• 2017

Recently, rapid advances of Ethernet and IP technology have brought many changes in the sound industry. In addition, due to AoIP-based audio transmission technology, various problems of the acoustic system (sound quality deterioration due to long distance transmission, complicated wiring) have improved dramatically. However, when many distributed audio systems are connected with AoIP equipment, if there is a problem in the equipment, it is impossible to operate the connected system. AoIP equipment only can transmit audio signals but cannot adjust the system for acoustic environment. In this paper, AoIP equipment is to be installed with sound equipment on a one-to-one basis, so that various existing problems can be solved and adjustment of sound quality (reverberation, echo, delay and EQ) can be possible by AoIP-based OAC (On-site Audio Center) with built-in DSP function. As a result, uncompressed real-time transmission by distributed transmission/receipt module in OAC (On-site Audio Center) and high quality sound by adjustment of sound quality with built-in DSP can be acquired. It is expected that OAC based sound system will be the industry standard in ubiquitous environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.2
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• pp.225-232
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• 2017

Recently, the development of digital transmission technology of audio signals and the introduction of audio network equipment using digital transmission technology have been made. As a result, audio network technology and equipment are actively applied to the design and construction of audio systems. The meeting room is a place where a large number of participants exchange opinions and communicate with each other. In addition to using an electric acoustic device such as a microphone and a speaker, it improves the intelligibility of the conference room through an example using an audio network.

• 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.