• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.865-873
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• 2004

MPEG-2 Advanced Audio Coding(AAC) is widely used in the multi-channel audio compression standards. And it combines hi인-resolution filter bank prediction techniques, and Huffman coding algorithm to achieve the broadcast-quality audio level at very low data rates. The forward and inverse modified discrete transforms which are operated in the encoder and the decoder of the filter bank need many computations. In this paper, we propose suitable recursive structure at IMDCT processing for MPEG-2 AAC real-time decoder. We confirm the memory, the computation speed and complexity of the proposed structure.

• The Journal of the Acoustical Society of Korea
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• v.25 no.3E
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• pp.89-94
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• 2006

For audio indexing and targeted search of specific audio or corresponding visual contents, the MPEG-7 standard has adopted a sound classification framework, in which dimension-reduced Audio Spectrum Projection (ASP) features are used to train continuous hidden Markov models (HMMs) for classification of various sounds. The MPEG-7 employs Principal Component Analysis (PCA) or Independent Component Analysis (ICA) for the dimensional reduction. Other well-established techniques include Non-negative Matrix Factorization (NMF), Linear Discriminant Analysis (LDA) and Discrete Cosine Transformation (DCT). In this paper we compare the performance of different dimensional reduction methods with Gaussian mixture models (GMMs) and HMMs in the classifying video sound clips.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2019.05a
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• pp.456-458
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• 2019

In this paper, transmission performance of streaming audio as a railway communication service based on LTE-R is analyzed. Performance analysis is perfomed with computer simulation based on NS(Network Simulation)-3, audio frame of MPEG(Moving Picture Experts Group)-4 is used as target application service for straming audio traffic. Results of this paper can be used to implement LTE-R networks and develope application services over LTE-R network.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.153-156
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• 2004

Spatial Audio Coding (SAC)은 낮은 대역폭에서 다채널/다객체 오디오 신호를 전송하기 위해 제안된 기술이다. 본 논문에서는 MPEG 에서 SAC 기술의 평가 방법으로 채택된 Multi-Stimulus test with Hidden Reference and Anchor (MUSHRA) 실험 절차에 대해서 설명한다. 또한 제 69 차 MPEG 회의에서 제안된 4 개 기관의 SAC 기술에 대한 청취실험을 수행하고 그 결과를 분석한다.

• 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.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.241-242
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• 2006

MPEG Surround is an audio coding technology that represents multi-channel audio signal with downmixed audio signal(s) and very low bitrate side information based on Binaural Cue Coding. The side information consists of Inter-Channel Level Difference, Inter-Channel Correlation, and payloads. These two parameters are correspondent to the well-known spatial parameters in psycho-acoustics, Inter-aural Level Difference (ILD) and Inter-Aural Cross Correlation (IACC). Though ICLD is to provide perceptually equivalent ILD to the listener, however, the ILD of the original multi-channel audio signal and that of the MPEG Surround encoded signal was different. The difference between two ILD values is defined as ILD Distortion (ILDD). This paper provides how ILDD can be applied to enhance sound quality in MPEG Surround and how much ILDD is decreased.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.43-48
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• 2000

This paper introduces MPEG-2 AAC codec system fur digital audio broadcasting. This system consists of encoder and decoder, and this system provides MPEG-2 system multiplexing and demultiplexing functions. Four DSPs are adopted fur encoder and three DSPs fur decoder. Each DSP processes system control, I/O control, and audio signal processing, multiplexing and demultiplexing. This paper also discusses about some near future estimations related to DAB system and services. And at the end of this paper describes about future development plans.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1C
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• pp.12-18
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• 2008

This paper This paper proposes a new inverse-quantization(IQ) table interpolation algorithm, specialized Digital Signal Processor(DSP) instructions and hardware architecture for digital audio codecs. Non-linear inverse quantization algorithm is representatively used in both MPEG-1 Layer-3 and MPEG-2/4 Advanced Audio Coding(AAC). The proposed instructions are optimized for the non-linear inverse quantization. The proposed algorithm can minimize operational complexity which reduces total computational load. Performance comparisons show a significant improvement of average error. The proposed instructions and hardware architecture can reduce 20% of the instruction counts and minimize computational loads of IQ algorithms effectively compared with existing IQ table interpolation algorithms. Proposed algorithm can implement commercial DSPs.

• Journal of Broadcast Engineering
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• v.13 no.3
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• pp.415-418
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• 2008

Some embedded microprocessors do not have an FPU(Floating Point Unit) due to a circuit complexity and power consumption. The performance speed of MPEG-4 AAC decoder on this hardware environment would be slower than corresponding speed for playing back of the decoded results. Therefore, irritating and high-pitched noises are interleaved in the original the audio data. So, in order to play MPEG-4 AAC file on such PDA, a new algorithm that transforms floating-point arithmetic to one with integers, is needed. We have developed a transformation algorithm from floating-point operation to integer operation and implemented the PDA's AAC Player. We also show the efficiency of our proposed method with the experimental results.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.249-254
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• 1998

MPEG 오디오 그룹에서는 오디오 부호화 기술의 국제 표준으로 MPEG-1 오디오, MPEG-2 오디오 BC, MPEG-2 AAC의 규격 제정을 완료하였고, 현재 MPEG-4 오디오 및 MPEG-7 오디오의 국제 표준을 제정하고 있다. 본 논문에서는 이들 표준에 대한 요구 기능 및 기술 특징을 분석하고, 각각의 표준에 대한 응용분야와 향후의 계획에 대하여 기술한다.