• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.477-481
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• 2004

High Quality CD, OAT audio requires that large amount of data. Currently, multi channel preference has been rapidly propagated among latest users. The MPEG(Moving Picture Expert Group) is provides data compression technology of sound and image system. The MPEG standard provides multi channel and 5.1 sounds, using the same audio algorithm as MPEG-l. And MPEG-2 audio is forward and backward compatible. The MDCT (Modified Discrete Cosine Transform) is a linear orthogonal lapped transform based on the idea of TDAC(Time Domain Aliasing Cancellation). In this paper, we proposed the micro-program type vector processor architecture a benefit in MDCT/IMDCT of MPEG-II AAC. And it's reduced operating coefficient by overlapped area to bind. To compare original algorithm with optimized algorithm that cosine coefficient reduced $0.5\%$multiply operating $0.098\%$ and add operating 80.58\%$. Algorithm test is used C-language then we designed hardware architecture of micro-programmed method that applied to optimized algorithm. This processor is 20MHz operation 5V.

• Journal of Digital Convergence
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• v.12 no.7
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• pp.77-83
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• 2014

Currently, a lot of colleges and universities offer Acoustics and audio engineering courses. In this paper, we analyze and classify the current state of the graduate level curriculums of the area. For the purposes, we focus on graduate school courses of the U.S. and Europe where audio engineering is highly advanced. They were classified into three different types depending on the educational objectives. In addition, the representative cases of each type are presented to examine the characteristics of the subjects.

• Journal of Broadcast Engineering
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• v.10 no.1 s.26
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• pp.22-30
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• 2005

We implement an interactive 3D audio broadcasting terminal which synthesizes an audio scene according to the request of a user. Audio scene structure is described by the MPEG-4 AudioBIFS specifications. The user updates scene attributes and the terminal synthesizes the corresponding sound images in the 3D space. The terminal supports the MPEG-4 Audio top nodes and some visual nodes. Instead of using sensor nodes and route elements, we predefine node type-specific user interfaces to support BIFS commands for field replacement. We employ sound spatialization, directivity/shape modeling, and reverberation effects for 3D audio rendering and realistic feedback to user inputs. We also introduce a virtual concert program as an application scenario of the interactive broadcasting terminal.

• Journal of Broadcast Engineering
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• v.12 no.3
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• pp.222-229
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• 2007

According to the advancement of multimedia compression technologies, high quality multi-media services are easily found in common life. Along with this situation, 5.1-channel audio service also has expanded the application area to home theater system and car theater system and consumer can easily take a chance to experience the feeling of 5.1-channel audio. On the other hand, terrestrial DMB service has been launched in Korea from Dec. 2005 as a handhold multi-media broadcasting service. However, multi-channel audio was not considered due to the insufficiency of bandwidth and the handhold usage. Lately, MPEG is standardizing high efficiency multi-channel audio compression technology for handheld broadcasting service, and several trial for application is introduced in Europe. In this paper, we would like to explain multi-channel audio compression technology, describe the implementation of the verification system for the multi-channel audio service over T-DMB and investigate the possibility of further realization of the service.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• Journal of Broadcast Engineering
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• v.21 no.6
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• pp.977-985
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• 2016

In this paper, we propose a new method for on-line genre classification using spectrogram and deep neural network. For on-line processing, the proposed method inputs an audio signal for a time period of 1sec and classifies its genre among 3 genres of speech, music, and effect. In order to provide the generality of processing, it uses the spectrogram as a feature vector, instead of MFCC which has been widely used for audio analysis. We measure the performance of genre classification using real TV audio signals, and confirm that the proposed method has better performance than the conventional method for all genres. In particular, it decreases the rate of classification error between music and effect, which often occurs in the conventional method.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.12
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• pp.108-116
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• 2009

In this paper, we propose a Ethernet based audio distribution system with a new L1/L2 hybrid switching scheme, and evaluate its performance. The proposed scheme not only offers guaranteed low latency and jitter characteristics that are essentially required for the distribution of high-quality uncompressed audio traffic, and but also provide an efficient transmission of data traffic on the Ethernet environment. The audio distribution system with a proposed scheme consists of a master node and a number of relay nodes, and all nodes are mutually connected as a daisy-chain topology through up and downlinks. The master node generates an audio frame for each cycle of 125us, and the audio frame has 24 time slotted audio channels for carrying stereo 24 channels of 16-bit PCM sampled audio. On receiving the audio frame from its upstream node via the downlink, each intermediate node inserts its audio traffic to the reserved time slot for itself, then relays again to next node through its physical layer(L1) transmission - repeating. After reaching the end node, the audio frame is loopbacked through the uplink. On repeating through the uplink, each node makes a copy of audio slot that node has to receive, then play the audio. When the audio transmission is completed, each node works as a normal L2 switch, thus data frames are switched during the remaining period. For supporting this L1/L2 hybrid switching capability, we insert a glue logic for parsing and multiplexing audio and data frames at MII(Media Independent Interlace) between the physical and data link layers. The proposed scheme can provide a good delay performance and transmission efficiency than legacy Ethernet based audio distribution systems. For verifying the feasibility of the proposed L1/L2 hybrid switching scheme, we use OMNeT++ as a simulation tool with various parameters. From the simulation results, one can find that the proposed scheme can provides outstanding characteristics in terms of both jitter characteristic for audio traffic and transmission efficiency of data traffics.

• Proceedings of the Korean Information Science Society Conference
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• 2006.06a
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• pp.274-276
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• 2006

임베디드 시스템은 특정 임무를 수행하도록 설계된 전용 컴퓨팅 시스템으로 그 용도에 따라 다양한 하드웨어 구성요소를 가지며, 그 쓰임새에 따라 특정 하드웨어 중심으로 시스템을 구현할 수 있다. 이런 하드웨어 시스템의 자원을 효율적으로 관리하기 위해서는 그 시스템에서 요구하는 기능을 만족시키는 특정 운영체제가 필요하다. 본 논문에서는 적은 크기의 메모리에 실시간 운영체제와 파일시스템, 애플리케이션이 모두 탑재되어야 하는 Audio Player 시스템을 위한 경량 실시간 운영체제를 설계 및 구현하였다.

• Proceedings of the IEEK Conference
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• summer
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• pp.114-117
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• 2004

This paper proposes a novel algorithm for hybrid lossless audio coding, which employs integer discrete cosine transform. The proposed algorithm divides the input signal into frames of a proper length, decorrelates the framed data using the integer DCT and finally entropy-codes the frame data. In particular, the adaptive Golomb-Rice coding method used for the entropy coding selects an optimal option which gives the best compression efficiency. Since the proposed algorithm uses integer operations, it significantly improves the computation speed in comparison with an algorithm using real or floating-point operations. When the coding algorithm is implemented in hardware, the system complexity as well as the power consumption is remarkably reduced. Finally, because each frame is independently coded and is byte-aligned with respect to the frame header, it is convenient to move, search, and edit the coded data.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2063-2066
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• 2002

In this paper, we propose a structure far lossless audio coding method. Prediction model is used in the wavelet transform domain. After DWT, wavelet coefficients is quantized and decorrelated by prediction modeling. The DWT can be constructed to critical bands. We can get a lower data rate representation of audio signal which has a good quality like the result of perceptual coding. Then the prediction errors are efficiently coded by the Golomb-coding method. The prediction coefficients are fixed for reducing the computational burden when we find prediction coefficients.