• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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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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• v.16 no.2
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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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• v.37 no.4
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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.

• IEMEK Journal of Embedded Systems and Applications
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• v.6 no.3
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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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• v.12 no.12
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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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.63-68
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• 2016

Audio mixer is an electronic device which performs a mixing of multiple audio signals. Digital mixer having various functions and scalability is spreaded thanks to advanced DSP and IT technology. However, digital mixer is more vulnerable to stability comparing to conventional analog mixer in the digital error or software error sense because its control is executed by SW. To solve this problem, in this paper, we propose a multi-channel digital analog hybrid mixer scheme, digital mixer error detection mechanism and malfunctioning switching technique. Also we develop the audio mixer having digital-analog hybrid structure. By simulation, we can sense the error of digital mixer except power loss in a 120ms, change into analog mixer mode automatically and provide continuous broadcasting function without mixer function loss.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.475-482
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• 2018

In this paper, we present a study on data augmentation methods for DNN (Deep Neural Network)-based audio tagging. In this system, an audio signal is converted into a mel-spectrogram and used as an input to the DNN for audio tagging. To cope with the problem associated with a small number of training data, we augment the training samples using time stretching, pitch shifting, dynamic range compression, and block mixing. In this paper, we derive optimal parameters and combinations for the augmentation methods through audio tagging simulations.

• Tribology and Lubricants
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• v.26 no.5
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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.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• The Transactions of the Korea Information Processing Society
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• v.3 no.3
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• pp.591-607
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• 1996

To support various multimedia applications. MuX server produces object-oriented and consistent interfaces for creation, copying, splitting, mixing and interleaving of streams. In this paper, we describes distributed connection structures and centralized connection structures which can be used in building a teleconferencing system using basic objects of MuX and compares merits and demerits of each structure from the viewpoint of multimedia related performance like delay and synchronization.