• Title/Summary/Keyword: Audio system

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Implementation of Tone Control Module in Anchor System for Improved Audio Quality

  • Seungwon Lee;Soonchul Kwon;Seunghyun Lee
    • 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.

An Architecture for 3D Audio Core Algorithm Evaluation DB (3차원 입체 음향 핵심 알고리즘 평가를 위한 DB 설계)

  • Hwang, Jaemin;Kim, Jeonghyuk;Kang, Sanggil
    • Journal of Information Technology and Architecture
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    • v.11 no.2
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    • pp.225-233
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    • 2014
  • In this paper an architecture for 3D audio core algorithm evaluation database system. Due to increase of 3D audio system through multimedia device, an evaluation system is required for evaluating the 3D core algorithms for developing 3D audio system. Conventional evaluation systems have some problems. Researchers have to learn usage of evaluation system, in addition it is inefficient to use and search audio sources because audio sources are not indexed in general. To solve these problems, we design the architecture of 3D audio core algorithm evaluation database system enabling to automatically evaluate core algorithms using database management system. Also we define XML metadata scheme for information of saved audio source in database. This approach allows improving efficiency of search audio source and use of audio database.

A study on the audio/video integrated control system based on network

  • Lee, Seungwon;Kwon, Soonchul;Lee, Seunghyun
    • International journal of advanced smart convergence
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    • v.11 no.4
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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.

Audio Fingerprinting Based Spatial Audio Reproduction System (오디오 핑거프린팅기반 입체음향 재현 시스템)

  • Ryu, Sang Hyeon;Kim, Hyoung-Gook
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.12
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    • pp.217-223
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    • 2013
  • This paper proposes a spatial audio reproduction system based on audio fingerprinting that combines the audio fingerprinting and the spatial audio processing. In the proposed system, a salient audio peak pair fingerprint based on modulation spectrum improves the accuracy of the audio fingerprinting system in real noisy environments and spatial audio information as metadata gives a listener a sensation of being listening to the sound in the space, where the sound is actually recorded.

Design and Development of T-DMB Multichannel Audio Service System Based on Spatial Audio Coding

  • Lee, Yong-Ju;Seo, Jeong-Il;Beack, Seung-Kwon;Jang, Dae-Young;Kang, Kyeong-Ok;Kim, Jin-Woong;Hong, Jin-Woo
    • ETRI Journal
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    • v.31 no.4
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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.

A Proposal for High-Resolution Encoding System with Backward Compatibility in CDDA (상용 CDDA와 하위 호환성을 가지는 고해상도 부호화방석의 제안)

  • Moon, Dong-Wook;Kim, Lark-Kyo;Nam, Moon-Hyun
    • Proceedings of the KIEE Conference
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    • 2004.11c
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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.

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Development of a Digital Down-mixer to Convert 5.1 Channel Audio Signals to Stereo Signals (5.1 채널 오디오 신호를 스테레오 신호로 변환하는 디지털 다운믹서 개발)

  • Jeon, Kwang-Sub;Cheong, Ho-Yong;Lee, Seung-Yo
    • 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.

Implementation of Local Distribution Audio System Based on AoIP (AoIP 기반 지역분산형 오디오시스템의 구현)

  • Kang, Min-Soo;Lee, Sang-Wook;Park, Yeoun-Sik
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.12 no.12
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    • pp.2165-2170
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    • 2008
  • In this parer, it is implemented a Local distribution Audio System, based on AoIP(Audio over Internet Protocol) of a part of TCP/IP Network which belongs to Internet transmission technology. The system is controlled based on SNMP(Simple Network Management Protocol) and it is transferred to UDP as packet after changing from Analog audio sources to Digital audio sources. The implemented Local distribution Audio System have presented practical possibilities in PA system transmitting various audio sources to several areas, dispersedly and using multichannel audio like Home theaters in the limelight, recently.

Retrieval of Broadcast News Using Audio Content Analysis

  • Kim, Hyoung-Gook
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.3E
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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.

The Implemetation of Real-time Broadcast Synchronizing System Using Audio Watermark (오디오 워터마크를 이용한 실시간 방송동기화시스템의 구현)

  • Shin Dong-Hwan;Kim Jong-Weon
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.54 no.12
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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.