• ETRI Journal
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• v.30 no.4
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• pp.606-608
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• 2008

MPEG Surround has a potential clipping problem since its encoding is based on downmixing a multichannel signal. We propose a clipping prevention scheme for MPEG Surround, which is composed of modification and recovery processes of a downmix signal with recovery information conveyed in arbitrary downmix gains of an MPEG Surround bitstream. Experiments show that the proposed scheme effectively prevents sound quality degradation caused by clipping problems with negligible additional bit rates.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.610-617
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• 2009

An overview of various modes and tools of MPEG Surround is provided Because the binaural mode of MPEG Surround supports the virtual 5.1-channel playback based on HRTFs, it can be played via headphones and earphones for portable audio devices. MPEG Surround also supports the enhanced matrix mode which converts stereo signals to 5.1-channel signals without side information, the 3D stereo mode which deals with 3D-coded signals, the low power version which greatly reduces the computational load in the decoding process. Besides, MPEG Surround provides the arbitrary downmix gains (ADGs) tool which is applied to artistic downmix signals, the matrix compatibility tool which is applied to downmix signals by conventional matrix-based methods, the residual coding tool -which can be used at high bit rates, and the GES tool which is applied to specific sound such as applause. The listening test results by various companies and organizations are also presented for important modes and tools.

• ETRI Journal
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• v.32 no.5
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• pp.831-834
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• 2010

This paper proposes a new joint channel coding algorithm based on principal component analysis. A conventional joint channel coder using passive downmixing undergoes a reduction of both the primary-to-ambient energy ratio (PAR) of the downmix signal and the panning gain ratio of the primary source. The proposed system preserves the PAR of the downmix signal by using active downmixing which reflects spatial characteristic. The proposed system also improves the accuracy of the panning gain ratio estimation. Computer simulations and subjective listening tests verify the performance of the proposed system.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.184-190
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• 2013

Parametric stereo(PS) audio coding is a specific version of spatial audio coding. In this paper, the problem due to the conventional synthesis of phase differences. In the conventional upmix matrix, phase differences are synthesized not only on downmix signal but also ambient signal, which violates the assumption that the ambient signals are anti-phased. Deterioration due to the phase synthesis is analyzed, especially, for low interchannel correlation. To solve this problem, new upmix matrix is proposed, which synthesizes phase differences only on downmix signal. The performance of the proposed upmix matrix is verified by the subjective listening tests.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.251-253
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• 2008

최근 유럽에서 DVB-T HDTV 방송 표준이 정하지면서 오디오 포맷으로 HE-AAC가 채택되었다. HE-AAC는 압축효율은 높지만 연산량이 높아 낮은 성능의 DSP에서 수행하기에는 어려움이 있다. DVB-T에서는 5.1채널을 사용하고 있어 더욱더 많은 연산을 필요로 한다. 본 논문은 ISO/DEC 14496-3 MPEG4 HE(High Efficiency)-AAC의 Level4에 해당하는 Multichannel Decoder를 최적화하여 구현하고. 가장 많은 연산을 필요로 하는 Synthesis Filter Bank에 제안된 알고리즘을 적용하여 연산량을 줄였고 대부분의 연산부를 어셈블리로 코드 최적화를 하여 작은 성능의 DSP를 사용하여 실시간 Multichannel HE-AAC Audio Decoder의 구현이 가능하게 하였다. DVB-T 오디오 시스템에 필수로 필요한 Audio Description, Dynamic Range Control, Downmix 등을 함께 구현하여 실제 수신기에 사용이 가능하도록 하였다. DSP는 Samsung의 CalmRISC16 + MAC24 core 를 사용하였다.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1215-1218
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• 1998

본 논문은 Digital Audio Compression(AC-3) Standard 인 A-52를 기반으로 하였으며 Borland C++3.1 Compiler를 사용하여 AC-3 Decoding Algorithm 구현하였다. Input Stream은 DVD VOB File에서 AC-3 Stream만을 분리하여 사용하며 최종 출력은 16 Bit PCM File이다. AC-3의 Frame구조는 Synchronization Information, Bit Stream Information, Audio Block, Auxiliary Data, Error Check로 구성된다. Aduio Block 은 모두 6개의 Block으로 나뉘어져 있다. BSI와 Side Information을 참조하여 Exponent를 추출하여 Exponent Strategy에 따라 Exponent를 복원한다. 복원된 Exponent 정보를 이용하여 Bit Allocation을 수행하여 각각의 Mantissa에 할당된 Bit수를 계산하고 Stream으로부터 Mantissa를 추출한다. Coupling Parameter를 참조하ㅕ Coupling Channel을 Original Channel로 복원시킨다. Stereo Mode에 대해서는 Rematrixing을 수행한다. Dynamic Range는 Mantissa와 Exponent의 Magnitude를 바꾸는 것으로 선택적으로 사용할 수 있다. Mantissa와 Exponent를 결합하여 Floating Point coefficient로 만든 후 Inverse Transform을 수행하면 PCM Data를 얻을 수 있다. PC에서 듣기 위해서는 Multi Channel을 Stereo나 Mono로 Downmix를 수행한다. 이렇게 만들어진 PCM data는 PCM Data를 재생하는 프로그램으로 재생할 수 있다.

• Proceedings of the KIEE Conference
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• 2008.10b
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• pp.304-305
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• 2008

DVD/BD 및 HDTV의 보급으로 인해 다수의 오디오 컨텐츠들이 멀티채널(5.1채널 이상) 형식으로 제작되고 있다. 오디오 정보를 담고 있는 물리적인 채널의 수가 증가하면, 이에 따라 정보량도 선형적으로 증가하게 된다. 선형적으로 증가된 정보를 기존의 오디오 코덱을 이용해 큰 압축할 경우, 압축에 필요한 비트레이트의 선형적인 증가를 피할 수 없다. 최근 채널 수 증가로 야기되는 비트레이트의 증가를 최소화하고 효율적으로 멀티채널 오디오 신호를 압축할 수 있는 방법으로 MPEG surround와 같은 파라메트릭 멀티채널 오디오 코딩 방식이 제안되었다. 파라메트릭 멀티채널 오디오 코딩 방식의 경우, 멀티채널 오디오 신호를 채널 수가 감소된 다운믹스 신호와 다운믹스 신호로부터 다시 멀티채널 오디오 업믹스 하는데 필요한 파라미터로 표현하는 방식이다. 따라서 다운믹스 방식 및 업믹스에 필요한 파라미터에 따라 업믹스된 멀티채널 오디오 신호의 품질이 달라진다. 본 논문에서는 MPEG surround에서 사용하고 있는 기존의 ITU-R 다운믹스 방식의 문제점을 실제 멀티채널 오디오 신호의 사례를 통해 제시하고 이 문제점을 해결하기 위한 새로운 다운믹스 방식과 파라미터를 제안하고자 한다.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.340-351
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• 2012

In this paper, we proposed the stereo-10.2channel blind upmix algorithm for the enhanced 3D sound. Recently, consumers want to enjoy better sound and the use of a various of multichannel configuration has been steadily improved. Thus, upmix algorithms have been researched. However, conventional upmix algorithms have the problem that distorts the spatial information of original source. To solve this problem and enhance the spatial sound quality, we proposed front and rear channel gain adjustment and 10.2 channel upmix algorithm for each additional channel. The listening test results show that it maintains spatial information of stereo input and enhances 3D sound effects unlike other conventional upmix algorithms.

• 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.