• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1415-1419
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• 2008

In this paper, we investigate various Audio Transmission Systems to implement Audio Transmission System based on AoIP of Internet transmission technology TCP/IP Network and we design and implement a Audio transmission system based on AoIP by adopting the most efficient one. The implemented system can be applied for various professional audio systems with large-scale audio distribution network as well as small-scale PA systems. Moreover, this can be applied in various fields which need audio transmission.

• 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.

• IEMEK Journal of Embedded Systems and Applications
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• v.2 no.1
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• pp.31-42
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• 2007

A transform coding scheme for the transmission of audio data in ZigBee based wireless personal area networks (WPAN) is presented in this paper. Here, wavelet transform is exploited to encode the features of audio data included mainly in the low frequency region. As a result, it is confirmed that the presented scheme recovers the original audio signals much accurately while it transmits the binary data compressed as 37.5% of the entire data generated without coding scheme. Especially, the mean-squared error between the recovered and original audio data approaches $10^{-4}$ when the signal-to-noise power ratio is sufficiently high. Hence, the presented coding scheme which exploits the wavelet transform is possibly applied for high-quality audio data transmission services in a small-scale sensor network based on ZigBee. Such a result is considered to be applicable as a basic material to update the technical specifications and develop the applications of ZigBee in WPANs.

• 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.

• 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.

• Journal of information and communication convergence engineering
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• v.18 no.2
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• pp.115-122
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• 2020

Audiovisual (AV) facilities such as TVs and signage are installed in various public places. However, audio cannot be used to prevent noise and interference from individuals, which results in a loss of concentration and understanding of AV content. To address this problem, a total technique for remotely listening to audio from audiovisual facilities with clean sound quality while maintaining video and lip-syncing through personal smart mobile devices is proposed in this paper. Through the experimental results, the proposed scheme has been verified to reduce system power consumption by 8% to 16% and provide real-time processing with a low latency of 120 ms. The system described in this paper will contribute to the activation of audio telehearing services as it is possible to provide audio remote services in various places, such as express buses, trains, wide-area and intercity buses, public waiting rooms, and various application services.

• Journal of KIISE:Information Networking
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• v.37 no.2
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• pp.150-156
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• 2010

Audio watermarking is the process of embedding inaudible information in an audio signal and has been widely used for copyright protection. In this research, we use the audio watermarking not for copyright protection but for the transmission of side information for user convenience, presenting an algorithm suitable for this purpose. Our new algorithm includes watermark embedding/extraction methods during acoustic transmission through the air that employ energy modulation of the audio signal and an efficient two-step synchronization method. The proposed algorithm is evaluated in experiments of conveying information from audio speakers to a mobile phone. The results show that reliable transmission is possible within a 5m range, which is an improvement achieved by our algorithm as compared to the existing one.

• Proceedings of the KIEE Conference
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• 1987.07b
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• pp.1053-1057
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• 1987

In this paper, we have studied a communication protocol which may provide Audio/Video data transmission in real time. Auido/Video data have its own characteristics. A new transport protocol with realtime constraint has been designed and implemented which performs dynamic error control and flow control depending on the characteristics of transmitted Audio/Video data. Since the receiving data can be predicted from the previously received data using the prediction function in Auido/Video data transmission, these functions are introduced in our transport protocol that may possibly improve the speed of data transmission and give a real time response. We have tested our transport protocol and measured the performance by the simulation. We assume that our transport protocol would be used in LAN environment. Our prime purpose is to provide a reliable and real time Auido/Video data transmission service.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.6
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• pp.180-187
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• 2013

In this paper, we propose a LED communication based multi-hop audio data transmission network system. The main contribution and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit audio data signal with long distance via multi-hops. Second, the developed system has the following features: In transmitter, audio data is transmitted after encoding with S/PDIF format via a general LED. The relay receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality audio transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted audio quality according to the used LED colors.