• 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 Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.160-163
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• 2002

This paper presents a statistical technique for audio watermarking. We describe the application of the promising public key watermarking method to the patchwork algorithm. Its detection process does not need the original content nor the secret key used in the embedding process. Special attention is given to statistical method working in the frequency domain. We will present a solution of robust watermarking of audio data. In this scheme, an extension of patchwork audio watermarking is presented which enables public detection of the watermark. Experimental results show good robustness of the approach against MP3 compression and other common signal processing manipulations.

• Journal of Korea Game Society
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• v.2 no.2
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• pp.72-77
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• 2002

Sound contents of these days are in the trend of 3D surround audio which has much higher quality then CD audio with the aid of DVD, Development environments of sound contents should be advanced and the efficient application of this system is quite necessary. In this paper, we proposed an improving method for utilizing sound/audio card systems which are necessory for developing game and multimedia sound contents as well as relate know-hows of producing 3D sounds with sound production tools for their efficient usages and applications.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.4
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• pp.519-525
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• 2007

With the high interest on robot technology and application, the research on artificial auditory systems for robot is very active. In this paper we discuss sound source localization and tracing based on audio-visual information. For video signals we use face detection based on skin color model. Also, binaural-based DOA is used as audio information. We integrate both informations using Kalman filter. The experimental results show that audio-visual person tracking Is useful, specially in the case that some informations are not observed.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.157-159
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• 2014

In this paper, we introduce SlimSRP, an ultra-low-power digital signal processor (DSP) solution for mobile audio and voice applications. So far, application processors (APs) have taken charge of all the tasks in mobile devices. However, they have suffered from short battery life problems to deal with complex usage scenarios, such as always-on voice trigger with continuous audio playback. From extensive analysis of audio and voice application characteristics, SlimSRP is designed to relive the performance and power burden of APs. It employs three-issue VLIW architecture, and the major low-power and high-performance techniques include: (1) an optimized register-file architecture friendly for constants generation, (2) a powerful instruction set to reduce the number of register file accesses and (3) a unique instruction compression scheme that contributes to saved memory size and reduced cache miss. An implementation of SlimSRP runs at up to 200MHz and the logic occupies 95K NAND2 gates in Samsung 28LPP process. The experimental results demonstrate that a MP3 decoder application with a 128kbps 44.1kHz input can run at 5.1MHz and the logic consumes only 22uW/MHz.

• Journal of Broadcast Engineering
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• v.8 no.3
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• pp.217-227
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• 2003

This paper describes an implementation of the slide-show functionality, which is one of the services that can be provided by the Digital Multimedia Broadcasting (DMB). While the existing analog radio broadcasting services provide audio only, DMB slide-show is the functionality that can deliver still images associated with the audio. For example, it can deliver the photographs of the singer, album cover images, or the lyrics of the song that correspond to the audio. There are two modes for the transmission of the slide-show. Firstly. the program-associated data (PAD) field within the DMB audio frame can be utilized and secondly, the slide-show data can be transmitted, after being multiplexed, with other service data as individual data stream separated from the audio. This paper describes PC-based implementations of a transmitter-side module that inserts slide-show data into the PAD area within audio bitstream and a receiver-side application module that plays the slide-show through decoding the PAD within the received audio bitstream and demonstrates their validity through experiments.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.633-634
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• 2008

In this paper, we describe implementation of a network Speaker for easily read streaming audio data from the network. The Network Speaker uses MAXIM company's DS80C400 for network control and MAX542 for audio data play. The DS80C400 network microcontroller offers TCP IPv4/6 network stack with the TINI-OS provided in ROM. The TINI-OS is adopted as an embedded operating system. Application programs are implemented by using JAVA language.

• 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.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.35C no.3
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• pp.59-68
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• 1998

This paper propose a hardware architecure of multimediasysgem for integrated processing of the multimedia data such as audio and video, and describes on the design and implementation of multimedia integrated processing Unit. The unit comprises most commonly needed multimedia processing function for computer-based video conference: audio-visual datacapture, playback, compression, decompression as well as interleaving/disinterleaving of compressed audio-visual data. The proposed architecture minimizes the CPU overhead that might be caused by multimedia data processing and assures the fluent data flow among system components. Also, this unit is tested and analyzed under the computer-based video conference to confirm the multimedia unit of proposed architecture using communication protocol and application software through Ethernet and FDDI (Fiber Distributed Data Interface) networks.