• 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.

• 정보화연구
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• 제11권2호
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• pp.225-233
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• 2014

오디오 산업은 프리미엄 산업으로써 나날이 발전 하고 있다. 입체 음향 시스템에 관한 연구는 많이 진행 되고 있다. 하지만 Audio database, algorithm, evaluation, metadata scheme 이 모두 각각 이루어지고 있다. 하나의 시스템에서 만들어진 audio 알고리즘을 평가 하고, 저장 할 수 있다면 입체 음향 오디오 연구 발전에 도움이 될 것이다. 그래서 이 논문 에서는 실감형 3D 오디오의 알고리즘을 시스템 적으로 평가 할 수 있는 Database Architecture 제안 하고, 이 Database system 구현을 위하여 XML metadata scheme를 정의 하였다. 본 논문에서는 새로운 오디오 평가 DB를 제시하고, 이를 체계적으로 구현하기 위한 설계를 제시하고자 한다.

• International journal of advanced smart convergence
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• 제11권4호
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• pp.1-9
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• 2022

The recent development of information and communication technology is also affecting audio/video systems used in industry. The audio/video device configuration system changes from analog to digital, and the network-based audio/video system control has the advantage of reducing costs in accordance with system operation. However, audio/video systems released on the market have limitations in that they can only control their own products or can only be performed on specific platforms (Windows, Mac, Linux). This paper is a study on a device (Network Audio Video Integrated Control: NAVICS) that can integrate and control multiple audio / video devices with different functions, and can control digitalized audio / video devices through network and serial communication. As a result of the study, it was confirmed that individual control and integrated control were possible through the protocol provided by each audio/video device by NAVICS, and that even non-experts could easily control the audio/video system. In the future, it is expected that network-based audio/video integrated control technology will become the technical standard for complex audio/video system control.

• 전자공학회논문지
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• 제50권12호
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• pp.217-223
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• 2013

본 논문에서는 오디오 핑거프린팅 방식과 스파셜 오디오 처리 방식을 결합한 오디오 핑거프린팅 기반 입체음향 재현 시스템을 제안한다. 제안된 시스템에서는 변조스펙트럼 기반의 명확한 오디오 정점 핑거프린트를 이용하여 잡음환경에서 오디오 핑거프린팅 시스템의 검색정확도를 향상시켰으며, 메타데이터로 제공되는 스파셜 오디오 정보는 청취자에게 소리가 실제로 녹음된 공간에서 소리를 듣는 것 같은 느낌을 준다.

• ETRI Journal
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• 제31권4호
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• pp.365-375
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• 2009

In this paper, a terrestrial digital multimedia broadcasting (T-DMB) multichannel audio broadcasting system based on spatial audio coding is presented. The proposed system provides realistic multichannel audio service via T-DMB with a small increase of data rate as well as backward compatibility with the conventional stereo-based T-DMB player. To reduce the data rate for additional multichannel audio signals, we compress the multichannel audio signals using the sound source location cue coding algorithm, which is an efficient parametric multichannel audio compression technique. For compatibility, we use the dependent property of an elementary stream descriptor, and this property should be ignored in a conventional T-DMB player. To verify the feasibility of the proposed system, we implement the T-DMB multichannel audio encoder and a prototype player. We perform a compatibility test using the T-DMB multichannel audio encoder and conventional T-DMB players. The test demonstrates that the proposed system is compatible with a conventional T-DMB player and that it can provide a promisingly rich audio service.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2004년도 학술대회 논문집 정보 및 제어부문
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• pp.150-152
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• 2004

Conventional CDDA (Compact Disc Digital Audio) system has limitations come from sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. So, new medium is developed for high-resolution audio recording, like as DVD-audio etc. But CDDA is a widely used medium for high fidelity audio yet, because new medium has complexity and difficulty in manufacturing system. In this paper, we design a new encoding system for high-resolution audio signal. The system is backward compatible with conventional CDDA. By evaluation for encoding and decoding process, we describe practicability of our proposal system.

• 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.

• 한국정보통신학회논문지
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• 제12권12호
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• pp.2165-2170
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• 2008

본 논문에서는 인터넷 기반 전송기술인 TCP/IP 네트워크의 한 분야인 AoIP(Audio over Internet Protocol)를 기반으로 한 지역분산형 오디오시스템을 구현하였다. 이 시스템은 SNMP 프로토콜을 기반으로 제어하며 입력된 아날로그 음원을 디지털로 변환하여 패킷으로 만들어 UDP로 전송하게 된다. 구현된 지역분산형 오디오 시스템은 최근 들어 각광받고 있는 홈시어터 시스템과 같이 다중 채널의 음향을 이용하는 경우와 다양한 음원 소스를 여러 지역에 분산하여 전송하는 PA(Public Address)시스템 등의 실용 가능성을 제시하였다.

• The Journal of the Acoustical Society of Korea
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• 제26권3E호
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• pp.74-79
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• 2007

In this paper, we report our recent work on a indexing and retrieval system of broadcast news using audio content analysis. Key issues addressed in this work are two major parts of the audio indexing system: anchorperson detection based on audio segmentation, and phone-based spoken document retrieval, developed in the framework of the emerging MPEG-7 standard. Experiments are conducted on a database of Britisch broadcast news videos. We discuss the development of the retrieval system, and the evaluation of each part and the retrieval system.

• 대한전기학회논문지:시스템및제어부문D
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• 제54권12호
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• pp.716-722
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• 2005

In this paper, we propose the audio watermarking algorithm based on the critical band of HAS(human auditory system) without audibly affecting the quality of the watermarked audio and implement the detecting algorithm on the BSS(broadcast synchronizing system) for testing the proposed algorithm. According to the audio quality test, the SNR(signal to noise ratio) of the watermarked audio objectively is 66dB above. In the robustness test, the proposed algorithm can detect the watermark more than $90\%$ from various compression(MP3, AAC), A/D and D/A conversions, sampling rate conversions and especially asynchronizing attacks. The BSS automatically switches the programs between the key station and the local station in broadcasting system. The result of reliability test of implemented system by using the real broadcasting audio has no false positive error during 30 days. Because of detecting once processing per 0.5 second, we can judge that the false positive error does not occur.