• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.1956-1959
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• 2003

본 논문에서는 HDTV(High Definition TV) 방송수신을 위한 DSP(Digital Signal Processor)기반의 HD급 비디오/오디오 디코더 시스템을 개발하고 그 성능을 확인하였다. DSP 플랫폼은 TI(Texas Instrument)사의 TMS320C6415를 대상으로 하였으며 TI의 DSP RTOS인 DSP/ BIOS를 이용하여 방송스트림인 TS(Transport Stream)을 분리하기 위한 TS Demuxer, MPEG-2 비디오 디코더 및 AC-3 오디오디코더 알고리즘을 통합하였으며, 각각의 알고리즘은 대상 DSP플랫폼인 TMS320C64x에 맞게 고정소수점 구조화 및 최적화를 실시하였다. 테스트를 위한 시스템은 스트리밍을 위한 호스트 PC와 PCI(Peripheral Component Interconnect)버스를 통해 연결된 DSP보드로 구성하였으며 실제 HDTV당송용 스트림과 SD(Standard Definition)급 스트림을 이용하여 성능을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.770-776
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• 2004

The goal of perceptual audio coder is to reduce redundancy and irrelevancy of audio signal based on the concept of masking. Several studies on masking effect reveal that the masking threshold varies as a function of the noise-like or tone-like nature of audio signals. Therefore, tonality of audio signal influences significantly the quality and efficiency of perceptual audio coder In this paper, we propose a new effective algorithm for tonality measure using spectrum energy. Since the proposed algorithm consists of a few transcendental functions and simple operations, it has lower complexity than MPEG psychoacoustic model-II. The proposed algorithm was tested with some audio signals, and DSP implementation showed that the proposed algorithm could be implemented with 3 MIPS. These results illustrate the efficiency of proposed algorithm in both performance and complexity.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.257-258
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• 2006

In this paper, we address several issues in the real time implementation of MPEG-1 Layer II decoder on a fixed-point digital signal processor (DSP), especially TMS320C6416. There is a trade-off between processing speed and the size of program/data memory for the optimal implementation. In a view of the speed optimization, we first convert the floating point operations into fixed point ones with little degradation in audio quality, and then the look-up tables used for the inverse quantization of the audio codec are forced to be located into the internal memory of the DSP. And then, window functions and filter coefficients in the decoder are precalculated and stored as constant, which makes the decoder faster even larger memory size is required. It is shown from the real-time experiments that the fixed-point implementation enables us to make the decoder with a sampling rate of 48 kHz operate with 3 times faster than real-time on TMS320C6416 at a clock rate of 600 MHz.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.4 s.334
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• pp.61-68
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• 2005

A chip, which can fast encoder the audio data to AAC (Advanced Audio Coding) LC(Low Complexity) that is MPEG-2 audio standard, has been designed on the basis of a 32 bits DSP core and fabricated with 0.25um CMOS technology. At first, the various optimization methods for implementing the algerian are devised to reduce the memory size and calculation cycles. FFT(Fast Fourier Transform) hardware block is added to the DSP core to get the more reduction of the calculation cycles. The chips has the size of $7.20\times7.20 mm^2$ and about 830,000 equivalent gates, can carry out AAC encoding under 70MIPS(Million Instructions per Second).

• 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.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1990.06a
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• pp.648-654
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• 1990

• Journal of Broadcast Engineering
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• v.13 no.4
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• pp.452-462
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• 2008

In this paper, we address issues associated with the real-time implementation of the MPEG-1/2 Layer-II (or MUSICAM) and MPEG-4 ER-BSAC decoders for Digital Multimedia Broadcasting (DMB) on TMS320C64x+ that is a fixed-point DSP processor with a clock speed of 330 MHz. To achieve the real-time requirement, they should be optimized in different steps as follows. First of all, a C-code level optimization is performed by sharing the memory, adjusting data types, and unrolling loops. Next, an algorithm level optimization is carried out such as the reconfiguration of bitstream reading, the modification of synthesis filtering, and the rearrangement of the window coefficients for synthesis filtering. In addition, the C-code of a synthesis filtering module of the MPEG-1/2 Layer-II decoder is rewritten by using the linear assembly programming technique. This is because the synthesis filtering module requires the most processing time among all processing modules of the decoder. In order to show how the real-time implementation works, we obtain the percentage of the processing time for decoding and calculate a RMS value between the decoded audio signals by the reference MPEG decoder and its DSP version implemented in this paper. As a result, it is shown that the percentages of the processing time for the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders occupy less than 3% and 11% of the DSP clock cycles, respectively, and the RMS values of the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders implemented in this paper all satisfy the criterion of -77.01 dB which is defined by the MPEG standards.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.165-168
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• 1998

제안하고 있는 CSD17C00 chip은 C&S technology에서 개발한 것으로, 음성 신호 처리를 위해 범용으로 구현되었으며, 16 bit 40 MIPS DSP group OAK DSP Core를 포함, 이에 Miscellaneous Logic, Serial Port, Host Interface, Timer, Compander 의 5가지 Peripherals 과 범용 I/O Ports 로 설계되었다. 1차적으로 CSD17C00 Chip 의 성능을 점검하였다. 그 결과, 응용 프로그램은 28MIPS의 계산속도를 갖으며, 프로그램 ROM 크기는 8.85KWords 이고, 10KWords 의 데이터 ROM 과 4KWords 데이터 RAM을 필요로 한다. CSD17C00 CHIP은 멀티미디어 통신용 VOCODER 개발을 위한 범용성을 갖추고 있으며, VOCODER 용 S/W 개발 환경 및 H/W 구조가 여타 범용 DSP에 비해편의성고 K합리성을 제공하도록 설계되어 있다. 따라서, 이를 이용한다면, 멀티 미디어 통신용 VOCODER, INTERNET PHONE CO-PROCESSOR, DIGITAL RECODER, MPEG AUDIO ENCODER & DECODER 등 다양한 제품으로의 응용이 가능할 것으로 전망된다.

• 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.