DSP Starter Kit Research Articles

Simulation and Implementation of Signal Processing for LFM Radar Using DSK 6713

This research aims to propose a comprehensive simulation and implementation methodology for LFM (Linear Frequency Modulated) Radar Signal Processing and its application, using digital signal processing techniques on the DSP Starter Kit (DSK) 6713 board. The motivation behind this study is to develop control software based on MATLAB R14 and SIMULINK to model various system software tasks, including detection, A/D conversion, Fast Fourier Transform (FFT), modulation, accumulation, decision-making, and target detection. The simulations are categorized into two groups: ideal beat frequency and parameterized beat frequency. We introduce several important terminologies for consideration, including pulse compression, SNR, matched filter, Doppler effect, and more. The use of real-time data exchange (RTDX) will facilitate the generation of input data and enable real-time calculations for outputs, leading to the creation of machine code for the DSP chip. This process aims to ensure data verification calculations and enhance the credibility and performance of the proposed methodology. By conducting thorough simulations, verification, and practical testing, the study demonstrates the satisfactory credibility and performance of the developed method. Using this research, we aim to contribute to the advancement of LFM Radar Signal Processing and enable its efficient implementation using digital signal processing techniques on the DSP Starter Kit (DSK) 6713 board.

Real‐time implementation and performance evaluation of speech classifiers in speech analysis‐synthesis

In this work, six voiced/unvoiced speech classifiers based on the autocorrelation function (ACF), average magnitude difference function (AMDF), cepstrum, weighted ACF (WACF), zero crossing rate and energy of the signal (ZCR-E), and neural networks (NNs) have been simulated and implemented in real time using the TMS320C6713 DSP starter kit. These speech classifiers have been integrated into a linear-predictive-coding-based speech analysis-synthesis system and their performance has been compared in terms of the percentage of the voiced/unvoiced classification accuracy, speech quality, and computation time. The results of the percentage of the voiced/unvoiced classification accuracy and speech quality show that the NN-based speech classifier performs better than the ACF-, AMDF-, cepstrum-, WACF- and ZCR-E-based speech classifiers for both clean and noisy environments. The computation time results show that the AMDF-based speech classifier is computationally simple, and thus its computation time is less than that of other speech classifiers, while that of the NN-based speech classifier is greater compared with other classifiers.

Utilization of wireless technology for sound communication using TMS320 C6713

The development of communication technology, both on algorithms and devices has been so rapid. The application of communication devices that previously used radio signal (wireless) would be more efficient if cable (on wire) is used instead. With a large number of buildings with wide locations, there are still many spaces that are not reachable by communication devices. Therefore, it is necessary to add a new communication tool that can be used to convey information in the campus environment intended for announcements, from one place to another or from one room to another simultaneously, both with point to multipoint as well as point to point characteristic which is suitable for internal communication and emergency situations. This wireless network can be expanded without using a cable backbone. In addition, with the equal sharing of access rights, the network will be more effective because it can be done anywhere as long as there is still a network connection. Audio effects can be applied in real time using DSP (Digital Signal Processing) Starter Kit TMS320C6713 by implementing in the equipment. In the audio input signal of 0.050 Vpp there is no change in the frequency of the original signal that is 1 Khz there is a change in the intensity of the sound produced 0.364 Vpp, 2 kHz to 0.280 Vpp while 3 kHz to 0.236 Vpp. There is a change in amplitude while the output frequency approaches the original signal frequency.

Comparative performance evaluation of MMSE-based speech enhancement techniques through simulation and real-time implementation

In this article, a generic model of minimum mean square error (MMSE) based speech enhancement technique has been presented and implemented on the hardware. Models for different MMSE-based methods were obtained by changing the gain function in the generic model. In all the methods, modified cascaded median-based noise estimation method has been used for noise estimation. Performances of all these MMSE-based methods were compared, among themselves and also with the spectral subtraction method for speech enhancement. The results have been evaluated using objective measures, subjective measure and composite objective measures for different noisy speech files. Results, in terms of objective evaluation parameters, indicated that the adaptive β-order MMSE method yielded better performance than the other methods. In subjective quality test (according to MOS listening test), β-order MMSE and adaptive β-order MMSE method yielded high scores. Real-time implementation has been carried out using TMS320C6416T DSP starter kit and code composer studio software. Estimation of memory consumption and execution time has been done for all the methods.

Kinerja Modulasi BPSK Modem Software Defined Radio pada DSK TMS320C6713

Abstract— Software Defined Radio (SDR) is a signal processing technology that optimizes the use of PC as a device supporting. With the application of SDR in wireless communication system so it will provide possibility and flexibility on manipulating DSP without the need of hardware changes. SDR modems are designed to take advantage of Digital Signal Processor Starter Kit (DSK) TMS320C6713 for baseband signal processing. In the implementation phase, the DSK is programmed directly using Matlab Simulink integration to build a system of modulation and demodulation Binary Phase Shift Keying (BPSK). Sampling frequency 8000 kHz is used with a variety of bit rate of 1 kbps and 2 kbps.

Fractal Communication System Using Digital Signal Processing Starter Kit (DSK) TMS320c6713

In 1992, Wornell and Oppenheim did research on a modulation which is formed by using wavelet theory. In some other studies, proved that this modulation can survive on a few channels and has reliability in some applications. Because of this modulation using the concept of fractal, then it is called as fractalmodulation. Fractal modulation is formed by inserting information signal into fractal signals that are selffractal similary. This modulation technique has the potential to replace the OFDM (Orthogonal Frequency Division Multiplexing), which is currently used on some of the latest telecommunication technologies. The purpose of this research is to implement the fractal communication system using Digital Signal Processing Starter Kit (DSK) TMS320C6713 without using AWGN and Rayleigh channel in order to obtain the ideal performance of the system. From the simulation results using MATLAB7.4. it appears that this communication system has good performance on some channels than any other communication systems. While in terms of implementation by using (DSK) via TMS320C6713 Code Composer Studio (CCS), it can be concluded that thefractal communication system has a better execution time on some tests.

Curriculum Development of an Audio Processing Laboratory Course

This paper describes the development of an audio processing laboratory curriculum at the graduate level. A real-time speech and audio signal-processing laboratory is set up to enhance speech and multi-media signal processing courses to conduct design projects. The recent fixed-point TMS320C5510 DSP Starter Kit (DSK) from Texas Instruments (TI) is used; a set of courseware is developed. In addition, this paper discusses the instructor's and students' assessments and recommendations in this real-time signal-processing laboratory course.

Detection and Correction of Saturated Current Transformer Measurements Using Decaying DC Components

When fault currents contain decaying dc components, current transformers (CTs) face the risk of becoming saturated. Traditionally, fault current waveform or CT model analyses were used to detect CT saturation. This paper analyzes the decaying dc components in fault currents to detect CT saturation. The decaying dc component in fault currents is first estimated using phasor-based computations. Such a component is then used to define a detection index. The proposed detection index can be easily used to detect CT saturation because its value in an unsaturated period is within a pre-known range. After CT saturation has been detected, the current samples and phasors in the latest unsaturated period are used to correct the saturated current samples. The proposed algorithm was tested using MATLAB/SIMULINK simulator and realized on a DSP Starter Kit to demonstrate its effectiveness and applicability.

DSP implementation of underwater communication using SSB modulation with random carrier frequencies

Underwater acoustic communication is a rapidly growing field of research and applied engineering. It is especially necessary for submarine-ship, submarine-submarine and ship-submarine communications. Recent years have seen rapid advances in DSP (Digital signal processor) architecture technologies, which advances have made it possible for DSP's to be used for applications previously considered unsuitable such as underwater communication. Underwater acoustic communication applications had been performed in an analog environment several times before; however, in this work, it is performed digitally by using DSP. Alternative solutions for underwater communication are presented as well. The presented solution is a more reliable prototype design of underwater acoustic communication that was developed via the transmitter and receiver sides on two identical DSK (DSP Starter Kit) boards by using upper sideband - suppressed carrier (SSB - SC) and random carrier frequencies (8 - 16 kHz). Key words: Underwater acoustic communication, underwater telephone (uwt), digital signal processor.

A real time fingerprint identification system based on the AFS8600 sensor and the C6713 DSP processor

AbstractIn this paper, a fingerprint pre‐process and identification algorithm is implemented using the TMS320C6713 DSP starter kit (DSK) module, along with the Authentec AFS 8600 fingerprint sensor. Database: Acquisition is performed using a solid state sensor. The sensor postulates emerge from the theory of the Electric Field technology. Method: Images are first subjected to a frequency and orientation processing. This is achieved using Gabor‐based filters. This processing has been optimized and implemented on the DSP system. A new method has been developed in order to extract the feature vector of the fingerprint image. A grid, centered at the corepoint of the image, is applied to the fingerprint image in order to derive local information. Classification algorithms are developed, which include training as well as evaluating phase. The types of classifiers used were based on the Bayesian approach along with the K‐nearest neighbour. Results: An identification accuracy of 90% was achieved which is comparable to other procedures described in the literature. Conclusion: The combination of a fingerprint processing and identification algorithm using a low cost sensor and DSP module has been presented. (© 2008 WILEY‐VCH Verlag GmbH & Co. KGaA, Weinheim)

Adaptive Active Noise Control Schemes for Headset Applications

Abstract This paper presents the design and implementation of adaptive feedback active noise control system (ANC) for head phone applications. Active Noise Control is a technique of acoustic noise reduction using a secondary source of sound which produces “antinoise” to cancel the primary noise. In this paper, narrow band single channel feedback ANC for headsets application is focused upon. The filter weights are updated by using the feedback form of Filtered-X LMS (FXLMS) algorithm. Performances for the IIR-based filter, FIR-based filter are compared with those of the algorithm by using ADALINE. The real time implementation of the system is performed using TMS320C6713 DSP Starter Kit (DSK) and the performance is analyzed for two tone sinusoidal noise.

Digital-signal-processor-based waveform generator

This note describes a digital signal processor (DSP) based waveform generator, which can generate a sine wave, up to 24 kHz, a square wave, up to 5 kHz, and a triangular wave, up to 12 kHz. A DSP starter kit with Code Composer Studio has been used in the design of a waveform generator. The waveform generators can also produce periodic arbitrary waveforms and amplitude modulated signals. Two synchronized signals can be obtained by using the waveform generator too. The spectral components of the signals generated are found comparable with a commercially available signal generator. The total harmonic distortion of the sine wave generated is less than 0.6%.

A two-channel impulse response measurement system based on a DSP starter kit

A two-channel impulse response measurement system based on a DSP starter kit

Teaching and learning the hows and whys of real-time digital signal processing

This paper discusses the approach taken in conducting a real-time digital signal processing (DSP) design class for undergraduates. The most recent TMS320C5402 DSP Starter Kit (DSK) from Texas Instruments (TI) is used, and a set of courseware that exposes students to both theoretical understanding and hands-on exercises is developed. It leads the students from the conceptual stage in DSP design to the actual implementation stage. This paper discusses the lecturer and students' perspective in this real-time DSP course.

CONTROL OF A MAGNETIC LEVITATION SYSTEM USING A DIGITAL SIGNAL PROCESSOR AS A DIGITAL PHASE-LEAD CONTROLLER

This paper addresses real-time magnetic levitation control using the TMS320C31 Digital Signal Processing Starter Kit (DSK). A phase-lead controller is designed. The controller coefficients are then included in an assembly language program that implements the phase-lead controller. A MATLAB program is used to activate the phase-lead controller, calculate and plot the root locus of the control system. When the MATLAB program is run, it plots the root locus of the system on the PC screen, assembles the phase-lead controller assembly language program, and loads/runs the resulting executable file on the C31 DSK to achieve real-time control. Results show the system outputs are as expected with a phase-lead controller, with actual system outputs matching theoretical calculations.

An academic DSP workstation

The University of Florida is developing a technology-based comprehensive engineering educational delivery system for the study of digital processing (DSP). The DSP curriculum, which covers both the undergraduate and graduate programs, exhibits a commitment to cooperative learning and personal computer-based instructional (CBI) tools. The article discusses the technological elements of this mix, which includes low-cost DSP software, a low-cost TI TMS320C5x DSP starter kit, and a generic soundboard.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>