• 전기학회논문지
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• 제62권12호
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• pp.1764-1770
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• 2013

Use of the 5.1 channel audio signals suitable for the television system is improper for the radio broadcasting system, which uses the stereo audio system. Therefore, it is necessary to develop an audio down-mixer to convert 5.1 multi-channel audio signals to stereo signals for radio broadcasting. In this paper, a development of an audio down-mixer was carried out to convert 5.1 multi-channel audio signals to stereo signals. The down-mixer which was developed can use the audio signals separated from video signals, including sound signals or individual signals provided from 3-channel AES/EBU signals including Left(L), Right(R), Left Surround(Ls), Right Surround(Rs), Center(C) and Low Frequency Effect(Lfe) sounds as mixer inputs.

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

• ETRI Journal
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• 제37권4호
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• pp.780-786
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• 2015

This paper describes an audio source separation that is based on nonnegative matrix factorization (NMF) and expectation maximization (EM). For stable and highperformance separation, an effective auxiliary source separation that extracts source residuals and reprojects them onto proper sources is proposed by taking into account an ambiguous region among sources and a source's refinement. Specifically, an additional NMF (model) is designed for the ambiguous region - whose elements are not easily represented by any existing or predefined NMFs of the sources. The residual signal can be extracted by inserting the aforementioned model into the NMF-EM-based audio separation. Then, it is refined by the weighted parameters of the separation and reprojected onto the separated sources. Experimental results demonstrate that the proposed scheme (outlined above) is more stable and outperforms existing algorithms by, on average, 4.4 dB in terms of the source distortion ratio.

• 대한임베디드공학회논문지
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• 제6권3호
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• pp.174-182
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• 2011

Conventional home audio system (e.g. AV Receiver, CD Player etc) has a various functionality of audio play, channel mixing, but the remote controller of these audio players is too complex, difficult for user to manage them effectively. Users want to use these functionalities with more easy, comprehensible way. In this study, "intelligent real-sound presentation technology" that support high quality, realistic audio and the "design of complex information and controller of real sound using intelligent real sound play and control interface" will be introduced. So user can actively, realistically enjoy and play real sound based on user's preference, emotion and circumstance, instead of user's passive service.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권12호
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• pp.5936-5954
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• 2018

Eight out of the top ten the largest technology companies in the world are involved in some way with the coming mobile VR revolution since Facebook acquired Oculus. This trend has allowed the technology related with mobile VR to achieve remarkable growth in both academic and industry. Therefore, the importance of reproducing the acoustic expression for users to experience more realistic is increasing because auditory cues can enhance the perception of the complicated surrounding environment without the visual system in VR. This paper presents a audio downmixing system for auralization based on hardware, a stage of sound rendering pipelines that can reproduce realiy-like sound but requires high computation costs. The proposed system is verified through an FPGA platform with the special focus on hardware architectural designs for low power and real-time. The results show that the proposed system on an FPGA can downmix maximum 5 sources in real-time rate (52 FPS), with 382 mW low power consumptions. Furthermore, the generated 3D sound with the proposed system was verified with satisfactory results of sound quality via the user evaluation.

• 한국인터넷방송통신학회논문지
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• 제16권5호
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• pp.63-68
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• 2016

오디오 믹서(audio mixer)는 다수의 오디오 신호의 믹싱을 수행하는 전자장치이다, DSP기술과 IT기술의 발전으로 다양한 기능과 확장성을 가진 디지털 믹서가 시장에 보급되고 있다. 그러나 디지털믹서는 제어를 SW를 통해 수행하기 때문에 디지털 오류 또는 소프트웨어 오류에 대해 기존 아날로그 믹서에 비해 안정성에서 매우 취약하다. 본 논문에서는 이러한 문제점에 대응하기 위해 다채널 디지털 아날로그 하이브리드 믹서 구조와 디지털믹서 기능 고장감지 메커니즘과 고장절체 기술을 제안하고 디지털-아날로그 이중화 구조를 가진 오디오 믹서를 개발하였다. 실험결과 전원상실을 제외한 디지털 믹서의 고장을 120ms내에 감지하고 자동으로 아날로그 믹서 모드로 전환하여 믹서 기능 상실 없이 끊임없는 방송 기능을 제공하였다.

• 한국음향학회지
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• 제37권6호
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• pp.475-482
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• 2018

본 논문에서는 DNN(Deep Neural Network) 기반 오디오 표식을 위한 데이터 증강 방법을 연구한다. 본 시스템에서는 오디오 신호를 멜-스펙트로그램으로 변환하여 오디오 표식을 위한 심층신경망의 입력으로 사용한다. 적은 수의 훈련 데이터를 사용하는 경우 발생하는 문제를 해결하기 위해, 타임 스트레칭, 피치 변화, 동적 영역 압축, 블록 혼합 등의 방법을 사용하여 훈련 데이터를 증강시켰다. 사용된 데이터 증강 기법의 최적 파라미터와 최적 조합을 오디오 표식 시뮬레이션을 통해 확인하였다.

• Tribology and Lubricants
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• 제26권5호
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• pp.277-282
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• 2010

The purpose of this paper is to study and analysis the failure examples for parasitic current leakage produced in starting system on gasoline engine. It verified the discharge of battery by electric leakage because of internal wiring damage problem for CD auto changer installed in car. Also, it verified the no-stating phenomenon because of deposit forming by chemical reaction of battery fluid between battery post and cable fixing parts. It verified the damage for brush holder and commutator mixing by internal short phenomenon because of brush carbon a particle and engine oil that was flowed into internal of starting motor. It verified the working phenomenon of audio by a point of contact even if the driver turn to "LOCK" position the key.

• 디지털융복합연구
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• 제9권5호
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• pp.225-233
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• 2011

인터넷 전화는 IP 기반 인터넷에서 VoIP(Voice over IP) 기술을 이용해서 음성 전화를 지원하는 인터넷 서비스다. 인터넷 전화는 영상통화, 메시징과 같은 인터넷 멀티미디어 서비스를 융합한 음성전화 서비스를 지원할 수 있는 장점을 갖고 있다. 특히 스마트폰을 통한 인터넷 소셜 네트워크 서비스가 보급되면서 기존의 전화망을 대체하는 서비스로써 인터넷 전화에 대한 연구와 개발이 활발히 진행되고 있다. 본 논문에서는 SIP(Session Initiation Protocol) 기반 인터넷 전화의 음성 통화 내용을 녹취하는 시스템의 설계 및 구현에 대해 설명한다. 인터넷 전화 녹취 시스템은 리눅스 기반으로 양방향 음성 스트림을 믹싱하는 기능, 라이브 패킷 스니핑 기능, 녹취 음성 파일 송신 기능은 공개 소프트웨어를 사용해서 구현하였다. 향후 개발된 시스템은 VoIP 기반 콜센터 시스템 등과 같은 복합 시스템을 구축하는데 있어 기반 기술로 활용될 계획이다.

• 한국정보처리학회논문지
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• 제3권3호
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• pp.591-607
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• 1996

MuX서버는 다양한 멀티미디어 응용 프로그램 개발을 지원하기 위하여 멀티미디어 스트림의 생성 , 복제, 분할, 합성, 접합등의 멀티미디어 입출력을 위한 객체지향적이고 일관적인 인터페이스를 제공한다. 본 논문에서는 MuX의 요소객체를 이용하여 컴퓨터 영상회의 프로그램을 개발할 경우에 가능한 분산형과 집중형연결 구조에 대하여 지연, 동기 등 멀티미디어관련 성능에 따른 장단점을 비교 분석한다.