• 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.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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.10a
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• pp.301-306
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• 2011

In this paper, active noise barriers for attenuation of road noise are proposed. multi-channel audio systems, DAQ part and high performance DSP H/W were designed. Active noise control firmware programs were implemented for multi-channel off-line/on-line estimation methods for secondary path transfer functions and FIR/IIR filter structure are used main noise control algorithms. To evaluate performance of proposed systems, the experiments were performed in an active noise barrier test bed for various noise cases.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.18-24
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• 2010

In this paper, a new matrix decoding system using vector based Virtual Source Location Information (VSLI) is proposed as an alternative to the conventional Dolby Pro logic II/IIx system for reconstructing multi-channel output signals from matrix encoded two channel signals, Lt/Rt. This new matrix decoding system is composed of passive decoding part and active part. The passive part makes crude multi-channel signals using linear combination of the two encoded signals(Lt/Rt) and the active part enhances each channel regarding to the virtual source which is emergent in each inter channel. Since the virtual sources are related to the perceptual sound images in virtual sound field, the reconstructed multi-channel sound results in good dynamic perception and stable image localization. Moreover, the good channel separation is maintained with nonlinear trigonometric enhancing function.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.05b
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• pp.96-102
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• 2003

Today, stage technology is developing highly by application of digital computer. Performance is composed of audio/video and acoustic technology takes very important position in field of stage technology. Generally speaking, four factors of sound are loudness, pitch, sound timbre and duration. Loudness depends on sound pressure level, yet partly related with spectrum and dulation. Pitch depends mainly on frequence and have a relation with sound pressure and duration. sound timbre depends strongly on spectrum and have a relation with frequence. In this paper, I designed a multi-microphone system which can used in broadcasting and performance stage with vicboss 200MHz-VHF wireless microphone and vicboss 900MHz-VHF wireless microphone. I also studied about multi-microphone which can use conveniently in the super play that needs many microphones. If this multi-microphone is prodused, we could expect better sound quality and a big progress in stereo recording technology.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.8
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• pp.1612-1627
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• 1994

There is a synchronization function which deals with only single media of text in the OSI Session Layer. So new synchronization schem and synchronization protocol are required for multimedia communications which include audio, video and graphic as well as text information. In this paper, conceptional Multmedia Synchronization Layer(MS layer) environment is composed and its service primitives and protocols based on 'multi-channel, base media scheme' are designed and proposed for multimedia synchronization services. This MS layer Manager (MSM) establishes the MS layer connection to the peer MS layer and manages each media channel which is created in MS layer media by media. The MSM also finds the synch-position through the media frame number by utilizing it like the time stamp to provide inter-media synchronization services as well as intra-media synchronization services.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.600-605
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• 2020

In this paper, we propose a sound event detection method using a multi-channel multi-scale neural networks for sound sensing home monitoring for the hearing impaired. In the proposed system, two channels with high signal quality are selected from several wireless microphone sensors in home. The three features (time difference of arrival, pitch range, and outputs obtained by applying multi-scale convolutional neural network to log mel spectrogram) extracted from the sensor signals are applied to a classifier based on a bidirectional gated recurrent neural network to further improve the performance of sound event detection. The detected sound event result is converted into text along with the sensor position of the selected channel and provided to the hearing impaired. The experimental results show that the sound event detection method of the proposed system is superior to the existing method and can effectively deliver sound information to the hearing impaired.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.125-128
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• 2002

In this paper, a new interactive TV service model is proposed. The MPEG-4 system is specified for composing and managing various object streams including user interactions. The data broadcasting model supporting user interactions is designed using MPEG-4 system in our proposal. We evaluate possibility of proposed service model using simulation player. This player supports MPEG-2 TS which contains MPEG-2 video and AC-3 audio streams as a main service and MPEC-4 system data as interactive services as well as user specific EPG information, and XML data, etc as supplemetary services. The player also supports a multi-channel environment. The synchronization between audio and visual data is achieved by DTS and PTS in TS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.2
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• pp.432-443
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• 2017

The primary goal of the MPEG-H 3D Audio standard is to provide immersive audio environments for high-resolution broadcasting services such as UHDTV. This standard incorporates a wide range of technologies such as encoding/decoding technology for multi-channel/object/scene-based signal, rendering technology for providing 3D audio in various playback environments, and post-processing technology. The reference software decoder of this standard is a structure combining several modules and can operate in various modes. Each module is composed of independent executable files and executed sequentially, real time decoding is impossible. In this paper, we make DLL library of the core decoder, format converter, object renderer, and binaural renderer of the standard and integrate them to enable frame-based decoding. In addition, by measuring the computation complexity of each mode of the MPEG-H 3D-Audio decoder, this paper also provides a reference for selecting the appropriate decoding mode for various hardware platforms. As a result of the computational complexity measurement, the low complexity profiles included in Korean broadcasting standard has a computation complexity of 2.8 times to 12.4 times that of the QMF synthesis operation in case of rendering as a channel signals, and it has a computation complexity of 4.1 times to 15.3 times of the QMF synthesis operation in case of rendering as a binaural signals.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.4
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• pp.142-152
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• 2011

In general, existing emergency evacuation guiding systems depend on visual techniques like emergency lights or LEDs. Actually people in the case of fire emergency condition may not obtain a range of view because of smoke from the fire. This paper introduces a technique to design an emergency guiding system using directivity sound to cope with this problem. In this case all speakers are serially connected for audio signal transmission in a serial fashion to achieve convenient speaker installation. Floyd algorithm is used to find shortest evacuation paths. Because serially connected multi-channel speakers are weak in case of disconnection, this paper uses a technique to solve the diagnostic problem. In the proposed system, a PC based on the USB protocol is used for control and observation. The system has achievements, such as increasing evacuation rate under emergency conditions, and serial transmission of audio signal for easy maintenance and low installation cost.