• Speech Sciences
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• v.7 no.2
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• pp.225-234
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• 2000

This paper describes the fixed-point DSP implementation of a CELP(Code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56362. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4E
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• pp.137-141
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• 2006

This paper presents HE-AAC decoder optimization on TMS320C55x fixed-point DSP core using a DSP-C like FFR code, which provides fast and flexible porting to a DSP core. Our optimization efforts are focused on methodologies that include general optimization methods of FFR code suitable for general DSP or RISC platform in high-level language and software optimization methods in assembly language level. The implementation result requires 48 MIPS and 135 Kbytes memory space to decode 48 Kbps stereo using real Korean S-DMB data.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.51-54
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• 2000

This paper describes the fixed-point DSP implementation of a CELP(code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically Parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56309. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.

• The Transactions of the Korean Institute of Power Electronics
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• v.17 no.3
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• pp.259-265
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• 2012

This paper introduces a method for design and implementation of a current controller for boost converter operating in continuous conduction mode (CCM) using a digital signal processor (DSP). A Proportional-Integral (PI) type current controller outputs an average voltage command for inductor, used in the input side of the boost converter, and the duty-ratio of PWM (pulse width modulation) signal for switching device is directly calculated from the average voltage command. The gains of the PI current controller are selected such that the current response characteristics are the same as those of a first-order low-pass filter. The proposed current control scheme is implemented using a DSP based on fixed-point math operations and an experimental study has been performed to validate the proposed method.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.253-256
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• 2000

This paper describes a DSP implementation of a real-time 3D sound localization algorithm with the use of a low power embedded DSP. A distinctive feature of this implementation is that the audible frequency band is divided into three, in accordance with the sound reflection and diffraction phenomena through different media from a certain sound source to human ears, and then in each subband a specific implementation procedure of the 3D sound localization is devised so as to operate real-time at a low frequency of 50MHz on a 16bit fixed-point DSP. Thus out DSP implementation can provide a listener with 3D sound effects through a headphone at low cost and low power consumption.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.19-23
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• 2001

DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

• ETRI Journal
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• v.19 no.3
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• pp.242-258
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• 1997

The Qualcomm code excited linear prediction (QCELP) speech coder was adopted to increase the capacity of the CDMA Mobile System (CMS). In this paper, we implemented the QCELP speech coding algorithm by using TMS320C50 fixed point DSP chip. Also the fixed point simulation was done with C language. The computation complexity of QCELP on TMS320C50 was 10k words and data memory was 4k words. In the normal call test on the CMS, where mobile to mobile call test was done in the bypass mode without double vocoding, mean opinion score for the speech quality was he Qualcomm code excited linear prediction (QCELP) speech quality was 3.11.

• Journal of Korea Multimedia Society
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• v.3 no.3
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• pp.263-271
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• 2000

In this paper, an acoustic echo canceller(AEC) is implemented by ADSP-2181. This AEC uses a noise robust adaptive algorithm and a postprocessing method which attenuates residual echo using cross-correlation between estimated error signal and microphone input signal. We propose new postprocessing method that uses two thresholds to prevent signal distortion after postprocessing and to improve the performance of AEC without extra computational burden. Through experiments using a 16 bit fixed-point DSP board (ADSP-2181 EZ-KIT Lite board), it is shown that the noise robust adaptive algorithm performs well in the double-talk situations and the convergence speed is comparable to NLMS. Using the postprocessor, ERLE is improved about 20 dB. As a result, the AEC with a postprocessor shows better performance than conventional ones.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.6
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• pp.147-155
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• 1989

In this paper, a real-time implementation of the TWS(track-while-scan) system using the high-speed DSP (digital signal processor) TMS320C25 is described. First, attempts have been made to investigate the FWL (finite word length) effect, which is caused by employing a fixed point arithmetic, of implementing the Kalman filter. The real-time TWS system consists of TWS arithmetic unit, scan converter, and system controller. In addition, the TWS system is in tegrated in the Multi-Bus. In experiment, it is observed that by employing the floating point arithmetic the computation time of 0.35sec is required for tracking 8 targets simultaneously, while 0.28sec is required for the fixed point arithmetic. Since the TWS system is designed to track up to 8 targets simultaneously, we conclude that the system is enough to process Kalman filter in a real-time.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04a
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• pp.453-456
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• 2001

21세기에 새로운 분야로 대두되고 있는 가상현실은 많은 사람들의 흥미를 끌고 있다. 상상 속에서나 가능하던 일들을 현실감과 입체감을 통해 실제처럼 느낄 수 있게 해준다는 것이 가상현실의 가장 큰 매력일 것이다. 가상현실을 요하는 멀티미디어 기기들의 활발한 시장진출로 3D 효과를 가진 오디오/비디오의 하드웨어 구현이 불가피하다. 본 연구에서는 휴대용 기기들에서 실시간 3D 입체음향 효과를 얻을 수 있도록 하드웨어를 구성하였다. 기존의 입체음향 기술에서 사용되는 콘볼루션 방법은 계산량이 많기 때문에 실시간 구현이 어렵다. 그러나 제안된 방식은 FFT를 사용하여 주파수 영역에서 처리함으로써 계산량을 줄여 하나의 프로세서로도 실시간 처리가 가능하도록 하였다. 저가/저전력/소형화조건을 요구하는 휴대용 기기에서 3D 입체 음향 효과를 얻을 수 있는 것이다. DSP는 TI(Texas Instruments)사의 16비트 고정소수점(fixed-point) 프로세서인 TMS320C5416을 사용한다. 구현된 3D 입체음향 칩은 입체음향을 필요로 하는 휴대용 MP3 Player, 가전용 오디오/비디오 등에 응용될 수 있다.