• 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 Digital Contents Society
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• v.8 no.3
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• pp.341-346
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• 2007

The purpose of this paper is to propose the design of VADSM(Value-Added Digital Speaker Module) which tunes up the speaker unit by measuring the speaker's frequency responses and controlling EQ band. This module can reduce audible distortions at particular frequency band and improve some flatness in the speaker's frequency response. VADSM is composed of DSP AMP and speaker unit. When a speaker transforms electrical signal to sound, the magnitude response at some frequencies are more or less than normal level. So, DSP AMP can be used to adjust those magnitudes up or down by controlling its EQ bands.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1409-1418
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• 1998

The recent audio CODEC(Coding/Decoding) algorithms are complex of several coding techniques, and can be divided into DSP tasks, controller tasks and mixed tasks. The traditional DSP processor has been designed for fast processing of DSP tasks only, but not for controller and mixed tasks. This paper presents a new architecture that achieves high throughput on both controller and mixed tasks of such algorithms while maintaining high performance for DSP tasks. The proposed processor, YSP-3, operates four algorithms while maintaining high performance for DSP tasks. The proposed processor, YSP-3, operates functional units (Multiplier, two ALUs, Load/Store Unit) in parallel via 4-issue super-scalar instruction structure. The performance evaluation of YSP-3 has been done through the implementation of the several DSP algorithms and the part of the AC-3 decoding algorithms.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.340-343
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• 2005

A realization for digital multimedia receiving system using Conditional Access System is presented in this paper. The key word for descrambling is make from smart card and Conditional Access System, a Stabilization is grow up in the method. It is possible to decoding that of average 15 fame/second of H.264 video format and that 24Khz${\sim}$48Khz audio sample rate using dual processor that of high performance DSP and RISC. This system is evaluated correct descrambling procedure in test stream added that signed user data.

• The KIPS Transactions:PartA
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• v.13A no.4 s.101
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• pp.275-288
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• 2006

Rapid design space exploration is crucial to customizing embedded system design for exploiting the application behavior. As the time-to-market becomes a key concern of the design, the approach based on an application specific instruction-set processor (ASIP) is considered more seriously as one alternative design methodology. In this approach, the instruction set architecture (ISA) for a target processor is frequently modified to best fit the application with regard to code size and speed. Two goals of this paper is to introduce our new retargetable compiler and how it has been used in ASIP-based design space exploration for a popular digital signal processing (DSP) application. Newly developed retargetable compiler provides not only the functionality of previous retargetable compilers but also visualizes the features of the application program and profiles it so that it can help architecture designers and application programmers to insert new application specific instructions into target architecture for performance increase. Given an initial RISC-style ISA for the target processor, we characterized the application code and incrementally updated the ISA with more application specific instructions to give the compiler a better chance to optimize assembly code for the application. We get 32% performance increase and 20% program size reduction using 6 audio codec specific instructions from retargetable compiler. Our experimental results manifest a glimpse of evidence that a higgly retargetable compiler is essential to rapidly prototype a new ASIP for a specific application.

• 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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• 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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.254-256
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• 2012

본 논문에서는 MPEG-D 오디오 서브그룹에서 진행 중인 Unified Speech and Audio Coding (USAC) 표준의 Linear Prediction Domain (LPD) 복호화기 모듈을 고정소수점 알고리즘으로 제안한다. USAC 부호화기는 두 개의 최신 음성-오디오 부호화기가 융합된 형태로, 음성 및 오디오 신호에 대하여 우수한 성능을 갖는 부호화기이다. USAC의 표준 완료와 본격적인 서비스화에 앞서서 USAC LPD 복호화기의 구조적인 특성을 분석하고, Digital Signal Processor (DSP)구현을 위한 LPD 복호화기의 고정소수점 알고리즘을 구축하는 동시에 모듈의 복잡도를 측정하고자 한다. 또한 고정소수점 알고리즘으로 구현된 LPD 복호화기와 기존의 부동소수점 복호화기의 성능을 비교하고, LPD 복호화기의 두 가지 부호화 모드에 따른 복잡도 이슈를 다루도록 한다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.6
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• pp.944-949
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• 1986

By improving the ability of arithmatic processing with an arithmatic processor in a microcomputer and realizing the data input system for real time analysis, an FFT analyzer that is usable within the range of audio frequency is designed and constructed. The input signal passes through a gain programmable pre-amplifier and anti-aliasing lowpass filter into an analogditital converter to be converted into digital form. The converted input data is processed by an Apple II microcomputer. The results of the processing are displayed using a microcomputer display unit and can be copied on a printer or stored in a floppy disk.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.1
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• pp.49-55
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• 2003

The use of digital filters in the signal process field is increasing rapidly with development of the modern industrial society. Especially, detail processors, Y/C separators, ghost removing filters, standard converters (NTSC to PAL or PAL to NTSC) and noise reducers, all of which use digital filters, tend to be used in digital video and audio processing, CATV and various communication fields. Generally, there are two different digital filters, the Rf (infinite impulse response) filter and the FIR (finite impulse response) filter in digital filter. In this paper, we have designed FIR filter which has the phase linearity and the easiness of creation. In the design of the FIR digital filter, the window function is used to alleviate the ripples caused by Gibbs Phenomenon around the cut off frequency of the band pass. But there're some problems to choose proper window function for the design destination due to its fixed values. Therefore, in this paper, we designed a modified Hanning window with new parameter which is adaptively chosen corresponding to design objectives. The digital filter was simulated to prove the validity of the model and it was compared with the Hamming, the Manning, the Blacknan and the Kaiser window function. And we have used peak side-lobe and transient characteristics as standard of judgement.