• The Journal of the Acoustical Society of Korea
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• v.17 no.3
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• pp.27-36
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• 1998

현재 상용화된 사운드 합성 기기에서 널리 쓰이고 있는 PCM 방식에서의 문제점은 고음질의 음을 얻기 위해서 많은 메모리 용량을 필요로 하는 것이다. 이 논문에서는 이 문 제를 해결하기 위해 MPEG 오디오 압축 방식을 적용하여 샘플된 음을 압축하고, 실시간으 로 이를 복호화 해서 음을 합성해내는 사운드 합성 시스템을 설계하였다. 사운드 합성 시스 템은 마이크로프로세서, 음원 DSP, MPEG 오디오 복호화기로 구성되며, 44.1Khz의 샘플링 주파수로 32개의 음을 동시에 합성할 수 있도록 설계되었다. 설계 과정에서 각각의 기능 요 소를 C언어로 기술하여 사운드 합성 시스템에 대한 소프트웨어 모델을 작성하였다. 이것을 통해 미리 전체 시스템의 동작을 시뮬레이션하고, 압축 방식을 적용함으로써 발생될 수 있 는 여러 가지 문제점에 대한 해결 방안을 제시하였다. 시뮬레이터로 시스템의 동작을 검증 한 후, DSP와 MPEG 복호화기를 포함하는 사운드 합성 시스템을 VHDL로 설계하여 시뮬 레이션을 통해 하드웨어가 정상적으로 동작함을 확인하였다. MPEG 오디오 압축 방식을 이 용함으로써 메모리 용량 측면에서는 약8:1의 감소 효과를 얻을 수 있다.

• Journal of the Korean Society for information Management
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• v.15 no.1
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• pp.149-164
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• 1998

The study aims to construct a system which can provide multimedia materials, specifically, digitalized video and audio materials on the internet. To accomplish this objective, it investigates technology on constructing a VOD/AOD system, current situations on video and audio data management in domestic and internatinal broadcasting institution and information centers. The study proposes a VOD/AOD system which can effectively manage and disseminate these materials on the internet.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.4
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• pp.477-482
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• 2007

This paper presents an audio decoder architecture which can decode AC-3 and MPEG-2 audio bit-streams efficiently. MPEG-2 synthesis filtering is modified by the 32-point FFT to share the common data path with the AC-3's. A programmable Audio DSP core and a hardwired common synthesis tilter are incorporated for effective decoding of two different formats.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.70-75
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• 2007

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2
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• pp.77-82
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• 1999

A sinusoidal model has been widely used in the analysis and synthesis of speech and audio signals, and becomes one of the efficient candidates for high quality low bit rate audio coders. One of the crucial steps in the analysis and synthesis using a sinusoidal model is the detection of tonal components. This paper proposes an efficient method for the analysis and synthesis of audio signals using a sinusoidal model, which uses psychoacoustic criteria such as masking effect, masking index, and JNDf(Just Noticeable Difference in Frequency). Simulation results show that the proposed method reduces the number of sinusoids significantly without degrading the quality of the synthesized audio signals.

• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.150-157
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• 2009

In this paper, we propose a person tracking based multi-channel audio panning system for multi-view broadcasting services. Multi-view broadcasting is to render the video sequences that are captured from a set of cameras based on different viewpoints, and multi-channel audio panning techniques are necessary for audio rendering in these services. In order to apply such a realistic audio technique to this multi-view broadcasting service, person tracking techniques which are to estimate the position of users are also necessary. For these reasons, proposed methods are composed of two parts. The first part is a person tracking method by using ultrasonic satellites and receiver. We could obtain user's coordinates of high resolution and short duration about 10 mm and 150 ms. The second part is MPEG Surround parameter-based multi-channel audio panning method. It is a method to obtain panned multi-channel audio by controlling the MPEG Surround spatial parameters. A MUSHRA test is conducted to objectively evaluate the perceptual quality and measure localization performance using a dummy head. From the experiments, it is shown that the proposed method provides better perceptual quality and localization performance than the conventional parameter-based audio panning method. In addition, we implement the prototype of person tracking based multi-view broadcasting system by integrating proposed methods with multi-view display system.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.11
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• pp.82-88
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• 2016

In this paper, we propose an intelligibility enhancement algorithm for multimedia contents using spectral shaping. The dialogue signals is essential to understand the plot of audio-visual media contents such as movie and TV. However, the non-dialogue components as like sound effects and background music often degrade the dialogue clarity. To overcome this problem, this paper tries to improves the dialogue clarity of audio soundtracks which contain important cues for the visual scenes. In the proposed method, the dialogue components are first detected by soft masker based on speech presence probability (SPP) which is widely used in speech enhancement field. Then, extracted dialogue signals are applied to the spectral shaping method. It reallocate the spectral-temporal energy of speech to enhanced the intelligibility. The total energy is maintained as unchanged via a loudness normalization process to prevent saturation. The algorithm was evaluated using the modeled and real movie soundtracks and it was shown that the proposed algorithm enhances the dialogue clarity while preserving the total audio power.

• 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.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.958-966
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• 2007

In this paper, we implement a sound-enhanced MPEG-1 audio decoder on embedded OS Platform. Low bit rate lossy audio codecs such as MP3, OGG, and AAC for mitigating the problems in storage space and network bandwidth suffer a major common problem such as a loss of high frequency fidelity of audio signal. This high frequency loss will reproduce only a band-limited low-frequency part of audio in the standard CD-quality audio. In order to overcome this problem, we embedded a sound enhancement algorithm into the MPEG-1 audio decoder and then the algorithms optimized according to the characteristic of the MPEG-1 audio layer I, II, III were implemented on an embedded OS platform. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed system compared to the standard MPEG-1 audio decoder.