• ETRI Journal
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• v.22 no.1
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• pp.20-29
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• 2000

In this paper, we present a design of video and audio single chip encoder/decoder for portable multimedia application. The single-chip called as video audio signal processor (VASP) consists of a video signal processing block and an audio single processing block. This chip has mixed hardware/software architecture to combine performance and flexibility. We designed the chip by partitioning between video and audio block. The video signal processing block was designed to implement hardware solution of pixel input/output, full pixel motion estimation, half pixel motion estimation, discrete cosine transform, quantization, run length coding, host interface, and 16 bits RISC type internal controller. The audio signal processing block is implemented with software solution using a 16 bits fixed point DSP. This chip contains 142,300 gates, 22 Kbits FIFO, 107 kbits SRAM, and 556 kbits ROM, and the chip size is $9.02mm{\times}9.06mm$ which is fabricated using 0.5 micron 3-layer metal CMOS technology.

• Journal of Broadcast Engineering
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• v.12 no.3
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• pp.222-229
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• 2007

According to the advancement of multimedia compression technologies, high quality multi-media services are easily found in common life. Along with this situation, 5.1-channel audio service also has expanded the application area to home theater system and car theater system and consumer can easily take a chance to experience the feeling of 5.1-channel audio. On the other hand, terrestrial DMB service has been launched in Korea from Dec. 2005 as a handhold multi-media broadcasting service. However, multi-channel audio was not considered due to the insufficiency of bandwidth and the handhold usage. Lately, MPEG is standardizing high efficiency multi-channel audio compression technology for handheld broadcasting service, and several trial for application is introduced in Europe. In this paper, we would like to explain multi-channel audio compression technology, describe the implementation of the verification system for the multi-channel audio service over T-DMB and investigate the possibility of further realization of the service.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.149-152
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• 2003

In this paper, we propose a scheme that applies to the VoIP application based on H.323 protocol to enhance the bandwidth efficiency. We multiplex the audio and video stream. In this scheme, audio frame is carried with video stream. And we applies not only multiplexing but also (in header compressing to the real audio/video stream to increase the bandwidth efficiency. With the multiplexing and RTP header compressing, we gain the bandwidth efficiency. In the finite network environment, We can assign bandwidth to other users who want to use other service. and other VoIP users. If we can apply the real time network situation to the our VoIP application, we can get more efficient performance.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.2
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• pp.461-468
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• 2000

Amounts of audio data used in several computer application have necessitated the development of audio database systems with newer features such as content-based queries and similarity searches to manage and use such data. Fast and accurate retrievals for content-based queries are crucial for such systems to be useful. Efficient content-based indexing and similarity searching schemes are keys to providing fast and relevant data retrievals. This paper present a method for indexing of Korean Traditional Music audio data based on wavelets. Also this paper present possibility of wavelet based audio indexing.

• Proceedings of the IEEK Conference
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• 2004.06b
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• pp.399-402
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• 2004

This paper presents a system-on-chip prototype implementing a full integration for a portable digital audio system. The chip is composed of a audio processor block to implements audio decoding and voice compression or decompression software, a system control block including 8-bit MCU core and Memory Management Unit (MMU) a low power 16-bit ${\Sigma}{\Delta}$ CODEC, two DC-to-BC converter, and a flash memory controller. In order to support other audio algorithms except Mask ROM type's fixed codes, a novel 16-bit fixed-point DSP core with the program-download architecture is proposed. Funker, an efficient power management technique such as task-based clock management is implemented to reduce power consumption for portable application. The proposed chip has been fabricated with a 4 metal 0.25um CMOS technology and the chip area is about 7.1 mm ${\times}$ 7.1mm with 100mW power dissipation at 2.5V power supply.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.74-78
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• 2004

As analog audio systems are changing to digital systems, the DAB (Digital Audio Broadcasting) is expected to provide CD quality audio, various data services with interactiveness and excellent mobile reception ability. The DMB (Digital Multimedia Broadcasting), as more advanced successor of the DAB, adds video capability on the audio and data services. The DAB system assures high quality audio services even when the reception is through portable and mobile receivers. In this paper, USB-DAB receiver and PCI-DMB receiver are designed and implemented. The DAB receiver and the DMB receiver incorporate with PC to make use of computational power and application software of Pc. This enables the developed system to be more flexible and to meet various applications easier.

• 한국정보컨버전스학회:학술대회논문집
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• 2008.06a
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• pp.53-56
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• 2008

This paper reports on the ACQA(Animated agent for Conversational Question Answering) project conducted at LIMSI. The aim is to design an expressive animated conversational agent(ACA) for conducting research along two main lines: 1/ perceptual experiments(eg perception of expressivity and 3D movements in both audio and visual channels): 2/ design of human-computer interfaces requiring head models at different resolutions and the integration of the talking head in virtual scenes. The target application of this expressive ACA is a real-time question and answer speech based system developed at LIMSI(RITEL). The architecture of the system is based on distributed modules exchanging messages through a network protocol. The main components of the system are: RITEL a question and answer system searching raw text, which is able to produce a text(the answer) and attitudinal information; this attitudinal information is then processed for delivering expressive tags; the text is converted into phoneme, viseme, and prosodic descriptions. Audio speech is generated by the LIMSI selection-concatenation text-to-speech engine. Visual speech is using MPEG4 keypoint-based animation, and is rendered in real-time by Virtual Choreographer (VirChor), a GPU-based 3D engine. Finally, visual and audio speech is played in a 3D audio and visual scene. The project also puts a lot of effort for realistic visual and audio 3D rendering. A new model of phoneme-dependant human radiation patterns is included in the speech synthesis system, so that the ACA can move in the virtual scene with realistic 3D visual and audio rendering.

• JSTS:Journal of Semiconductor Technology and Science
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• v.10 no.4
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• pp.276-281
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• 2010

A 2.5 V feed-forward second-order deltasigma modulator for audio application is presented. A 9-level quantizer with a tree-structured dynamic element matching (DEM) was employed to improve the linearity by shaping the distortion resulted from the capacitor mismatch of the feedback digital-toanalog converter (DAC). A chopper stabilization technique (CHS) is used to reduce the flicker noise in the first integrator. The prototype delta-sigma analogto-digital converter (ADC) implemented in a 65 nm 1P8M CMOS process occupies 0.747 $mm^2$ and achieves 109.1 dB dynamic range (DR), 85.4 dB signal-to-noise ratio (SNR) in a 24 kHz audio signal bandwidth, while consuming 14.75 mW from a 2.5 V supply.

• Journal of Industrial Technology
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• v.34
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• pp.27-32
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• 2014

Infrared remote controllers are widely used in controlling electronic devices due to its simplicity and convenience. This paper presents a smart phone application unifying any infrared remote controllers. We can select a device from device list menu of the application. Any new device can be added easily by downloading the protocol file of the device from protocol file server. Remote controller protocol files are stored in audio file format, and the file related to a specific menu button is transmitted through the audio jack of smart phones. The protocol file server is a standard file server, and protocol files for various devices are collected by infrared receiver module. For smart phones without infrared facility, a small infrared module translating audio signal to infrared signal is applied.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.3-12
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• 2001

Over the last few years, digital audio watermarking has become an interesting issue In many application areas, including digital broadcasting. This is primarily motivated by a need to provide copyright protection of digital audio content. Digital watermarking is a technique to embed copyright or other information int? the underlying data. Several possible audio watermarking techniques have been developed including spread spectrum watermarking, echo watermarking, phase coding, and patchwork. In this paper, we describe some requirements of digital audio watermarking as a tool for copyright protection of broadcasting content, and compare popular audio watermarking algorithms in some significant aspects.