• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.599-609
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• 2009

An overview of the recently finalized multi-channel audio coding standard MPEG Surround is provided. This audio coding standard downmixes multi-channel signals to mono or stereo signals and, simultaneously, extracts spatial parameters for its encoding process. In its decoding process, it reconstructs multi-channel signals based on the downmix signals and spatial parameters. Since the downmix signals are coded in conventional audio coding format such as AAC and MP3 and the spatial parameters require a small amount of information MPEG Surround guarantees high sound quality multi-channel audio at low bit rates. Besides, it is backward-compatible to conventional audio coding techniques because the downmix signals can be played on portable audio devices ignoring the spatial parameter information. In this paper, Part 1 presents an overview of the basic structure of MPEG Surround and Part 2 describes various modes and tools including the binaural mode which supports the virtual 5.1-channel playback via headphones or earphones. The listening test results by various companies and organizations are also presented.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.3
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• pp.87-95
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• 2010

Portable multi-media products which can service the highest audio-quality by using lossless audio codec has been released and the international lossless codecs, MPEG-4 audio lossless coding(ALS) and MPEG-4 scalable lossless coding(SLS), were standardized by MPEG in 2006. The simple profile of MPEG-4 ALS, it supports up to stereo, was defined by MPEG in 2009. The lossless audio codec should have low-complexity in stereo to be widely used in portable multi-media products. But the previous researches of MPEG-4 ALS have focused on an improvement of compression ratio, a complexity reduction in multi-channels coding, and a selection of linear prediction coefficients(LPCs) order. In this paper, the complexity and compression ratio of MPEG-4 ALS encoder is analyzed in simple profile of MPEG-4 ALS, the method to reduce a complexity of MPEG-4 ALS encoder is proposed. Based on an analysis of complexity of MPEG-4 ALS encoder, the complexity of short-term prediction filter of MPEG-4 ALS encoder is reduced by using the low-complexity filter that is proposed in previous research to reduce the complexity of MPEG-4 ALS decoder. Also, we propose a joint coding decision method, it reduces the complexity and keeps the compression ratio of MPEG-4 ALS encoder. In proposed method, the operation of joint coding is decided based on the relation between cross-correlation of residual and compression ratio of joint coding. The performance of MPEG-4 ALS encoder that has the method and low-complexity filter is evaluated by using the MPEG-4 ALS conformance test file and normal music files. The complexity of MPEG-4 ALS encoder is reduced by about 24% by comparing with MPEG-4 ALS reference encoder, while the compression ratio by the proposed method is comparable to MPEG-4 ALS reference encoder.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.

• 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 Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.74-80
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• 2010

Speech and audio codecs have been developed based on different type of coding technologies since they have different characteristics of signal and applications. In harmony with a convergence between broadcasting and telecommunication system, international organizations for standardization such as 3GPP and ISO/IEC MPEG have tried to compress and transmit multimedia signals using unified codecs. MPEG recently initiated an activity to standardize the USAC (Unified speech and audio coding). However, USAC RM (Reference model) software has been problematic since it has a complex hierarchy, many useless source codes and poor quality of the encoder. To solve these problems, this paper introduces a new RM software designed with an open source paradigm. It was presented at the MPEG meeting in April, 2010 and the source code was released in June.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.181-190
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• 2006

Technology for compressing low-bitrate multichannel audio coding should be developed owing to the increasing need of consumer for multichannel audio contents and services. To meet this requirement, MPEG has standardized MPEG Surround. In this paper, we introduce status on MPEG Surround standardization and analyze techniques adopted in the current MPEG Surround.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1567-1570
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• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1E
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• pp.29-34
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• 2009

MPEG4 High Efficiency Advanced Audio Coding (HE-AAC) and MPEG Surround are one of the most efficient combinations for low bit rate multi-channel audio coding. Based on the fact that these two codecs have identical quadrature mirror filter (QMF) analysis and synthesis structures, we propose a dual-domain connection scheme for the codecs. Specifically two time-domain connection methods are analyzed and compared to the QMF subband-domain connection method. Experimental results show that both the time-domain connection methods cause no subjective sound quality degradation compared to the QMF subband-domain connection method, which verifies that one can select either of them depending on application scenarios.

• 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.