• Journal of Broadcast Engineering
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• v.2 no.2
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• pp.86-93
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• 1997

Fast growth of multimedia applications requires new functions for video data processing. such as obj;cted-based video representation and manipulation. which are not supported by 11PEG-l and 11PEG-2. To support these requirements. 11PEG-4 video coding allows users to manipulate every video object easily by decomposing a scene into several video objects and coding each of them independently. However. the large amount of computations and flexible structure of 11PEG-4 video CODEC make it difficult to be implemented by either the general purpose DSP or a dedicated VLSI. In this paper, we propose a hardware architecture using a hybrid of a high performance programmable DSP and an application specific IC to implement a flexible 11PEG-4 video codec requiring the large amount of computations. The application specific IC has the functions of motion estimation and compensation.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.4
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• pp.30-39
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• 1994

A 1-chip high quality binarizing VLSI image processor (which has 8 bit ADC. 6 bit flash ADC, 15K standard cell, and 1K word ROM) based on 10 MIPS 16 bit DSP is implemented for FAX. This image processor(IP) performs image pre-processing. image quality improvement in copying and sending mode, and mixed image processing based on the fuzzy theory. And smoothing in sub-scan direction is applied for normal receiving mode data so the received data is enhanced like fine mode data. Each algorithm is processed with the same type of image processing window and 2-D image processing is implemented with a 1-D line buffer. The fabricated chip is applied to a FAX machine and image quality improvement is verified.

• Speech Sciences
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• v.6
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• pp.219-233
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• 1999

This paper presents a real-time implementation of a speaker independent speech recognition system based on a discrete hidden markov model(DHMM). This system is developed for a car navigation system to design on-chip VLSI system of speech recognition which is used by fixed point Oak DSP core of DSP GROUP LTD. We analyze recognition procedure with C language to implement fixed point real-time algorithms. Based on the analyses, we improve the algorithms which are possible to operate in real-time, and can verify the recognition result at the same time as speech ends, by processing all recognition routines within a frame. A car noise is the colored noise concentrated heavily on the low frequency segment under 400 Hz. For the noise robust processing, the high pass filtering and the liftering on the distance measure of feature vectors are applied to the recognition system. Recognition experiments on the twelve isolated command words were performed. The recognition rates of the baseline recognizer were 98.68% in a stopping situation and 80.7% in a running situation. Using the noise processing methods, the recognition rates were enhanced to 89.04% in a running situation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2303-2307
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• 2006

The round off noise properties of wave digital filters have known and desirable properties in respect to their realization with short coefficient wordlengths. This paper presents the optimal implementation of wave digital filters by employing multiprocessor archtectures in the sense of input sampling rate, the number of processors, and input-output delay. The implementation will be specified by complete circuit diagrams including control signals, and can be applied to an existing silicon complier for VLSI layout generation.

• Journal of the Korea Society of Computer and Information
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• v.11 no.2 s.40
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• pp.159-167
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• 2006

Lipreading has been suggested as one of the methods to improve the performance of speech recognition in noisy environment. However, existing methods are developed and implemented only in software. This paper suggests a hardware design for real-time lipreading. For real-time processing and feasible implementation, we decompose the lipreading system into three parts; image acquisition module, feature vector extraction module, and recognition module. Image acquisition module capture input image by using CMOS image sensor. The feature vector extraction module extracts feature vector from the input image by using parallel block matching algorithm. The parallel block matching algorithm is coded and simulated for FPGA circuit. Recognition module uses HMM based recognition algorithm. The recognition algorithm is coded and simulated by using DSP chip. The simulation results show that a real-time lipreading system can be implemented in hardware.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1296-1299
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• 2002

We implemented a CD signal processor operated on a CAV 48-speed CD-ROM drive into a VLSI. The CD signal processor is a mixed mode monolithic IC including servo-processor, data recovery, data-processor, and I-bit DAC. For servo signal processing, we included a DSP core, while, for CAV mode playback, we adopted a PLL with a wide recovery range. Data processor (DP) was designed to meet the yellow book specification.［2］So, the DP block consists of EFM demodulator, C1/C2 ECC block, audio processor and a block transferring data to an ATAPI chip. A modified Euclid's algorithm was used as a key equation solver for the ECC block To achieve the high-speed decoding, the RS decoder is operated by a pipelined method. Audio playability is increased by playing a CD-DA disc at the speed of 12X or 16X. For this, subcode sync and data are processed in the same way as main data processing. The overall performance of IC is verified by measuring a transfer rate from the innermost area of disc to the outermost area. At 48-speed, the operating frequency is 210 ㎒, and this chip is fabricated by 0.35 um STD90 cell library of Samsung Electronics.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.19 no.2
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• pp.61-67
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• 2000

Recent progress in digital and VLSI technology has offered new possibility fer noticeable advance of hearing aids. Yet, environmental noise remains one of the major problems to hearing aid users. This paper describes results which speech recognition performance and speech discrimination performance was measured for listeners with sensorineural hearing loss, while listeners in speech-band noise. In addition, to ameliorate hearing-aided environments of hearing impaired listeners, environmental noise reduction using speech enhancement techniques are investigated as a front-end of conventional hearing aids. Speech enhancement techniques are implemented in a realtime system equipped with DSP board. The clinical test results suggest that the speech enhancement technique may work in synergy with gain functions fer the greater SNR improvement as the preprocessing algorithm of digital hearing aids.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.6 s.336
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• pp.39-48
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• 2005

This paper proposes a reconfigurable coprocessor for communication systems, which can perform high speed computations and various functions. The proposed reconfigurable coprocessor can easily implement communication operations, such as scrambling, interleaving, convolutional encoding, Viterbi decoding, FFT, etc. The proposed architecture has been modeled by VHDL and synthesized using the SEC 0.18$\mu$m standard cell library. The gate count is about 35,000 gates and the critical path is 3.84ns. The proposed coprocessor can reduced about $33\%$ for FFT operations and complex MAC, $37\%$ for Viterbi operations, and $48\%\~84\%$ for scrambling and convolutional encoding for the IEEE 802.11a WLAN standard compared with existing DSPs. The proposed coprocessor shows Performance improvements compared with existing DSP chips for communication algorithms.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.8 no.1
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• pp.75-81
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• 2008

This paper presents a top-down approach to implement an MPEG-2 audio encoder in VLSI. As the algorithm of an MPEG-2 audio encoder is heavy-weighted and heterogeneous(to be mixture of several strategies), the encoder design process is undertaken carefully from the algorithmic level to the architectural level. Firstly, the encoding algorithm is analyzed and divided into sub-algorithms, called tasks, and the tasks are partitioned in the way of reusing the same designs. Secondly, the partitioned tasks are scheduled and synthesized to make the most efficient use of time and space. In the end, a real-time 5 channel MPEG-2 audio encoder is designed which is a heterogeneous multiprocessor system; two hardwired logic blocks and one specialized DSP processor.