DSP

In this paper, a technique for design, simulation, and DSP based program realization of recursive digital filters is proposed. The Matlab Filter Design Toolbox is used to obtain the coefficients, frequency and time domain responses with finite word length and second-order section realization of bandpass recursive digital filter. With the Real Time Workshop the previously simulated Simulink digital filter model has been automatically translated into DSP source code for the Texas Instruments DSP C5000 series. The TI Code Compose Studio is used to create a project that has been compiled by Visual Linker and has been loaded on the TMS320C5416 DSK board.

Multicore embedded systems introduce new opportunities and challenges. Scaling of computational power is one of the main reasons for a transition to a multicore environment. Parallel design patterns, such as Map Reduce, Task Graph, Thread Pool, Task Parallelism assist to derive a parallel approach for calculating the Fast Fourier Transform. By combining these design patterns, a robust application can be obtained. The key issues for concurrent calculation of a Fast Fourier Transform are determined at a higher level avoiding low-level patch-ups.

This study involves the fuzzy logic controlling of brushless DC (BLDC) motor with rapid control prototyping method. Initially a MATLAB/Simulink model of the six pulse BLDC motor and its power electronic driver is established. A fuzzy controller is developed by using MATLAB Fuzzy-Logic Toolbox and then it is inserted into the Simulink model. This model is directly communicated with a DSP through DS2201 dSPACE digital signal kit. PWM pulses are produced on the simulation model and DSP C program is directly developed through dSPACE system. It is shown that, fuzzy controller is successfully controlling the motor and, the model based programming of DSPs is very simple and versatile comparing with that of conventional method.

Comprobar de manera práctica el cálculo, la implementación y el funcionamiento, en el DSP TMS320C5402, de un oscilador tipo IIR de una determinada frecuencia; así como también de los filtros digitales tipo IIR y FIR.
La señal de un oscilador diseñado a 10 Hz, actúa como una señal de ruido sobre una señal de electrocardiograma (ECG) previamente muestreada, el objetivo principal es diseñar dos diferentes filtros, IIR y FIR, para eliminar la señal de 10 Hz de la señal ECG. Observando ventajas y desventajas de cada uno de ellos.

— Fast low power SRAMs have become a critical component of many VLSI chips. This is especially true for microprocessors, where the on-chip cache sizes are growing with each generation to bridge the increasing divergence in the speeds of the processor and the main memory. Simultaneously, power dissipation has become an important consideration both due to the increased integration and operating speeds. Thus, a significant effort has been invested in reducing the power of CMOS RAM chips using circuit and architectural techniques. This paper presents a design using hierarchical divided bit-line approach for reducing active power in SRAMs by 40-50% and access time at the expense of 5-10% increase in the number of transistors when compared to Conventional SRAM. A Hierarchical divided bit line approach is chosen to implement a 1Kb SRAM memory on 0.18 micron CMOS technology using CADENCE design tool. Keywords— SRAM Cell, Power, Hierarchical divided bit line approach, bit-line capacitance.

This paper discusses a highly flexible and powerful implementation toolkit that allows new ideas in the field of multi-channel surround sound to be implemented with great ease and minimal development time. The key feature of the toolkit is that it enables complex surround sound encoding and decoding algorithms to be implemented in real-time with relative ease. Introduction The authors experience in designing and implementing encoding and decoding systems for multi-channel surround sound showed that although the design of such systems was not in itself excessively complex, the time consuming nature of designing circuit boards and software routines for each new system, often proved to be prohibitive. This was especially true if a quick test of a new idea or innovation was required. The availability of a rapid development and prototyping toolkit for the design and implementation of multi-channel surround sound processors soon became a clear requirement. This paper describes a highly fl...

The focus of this thesis is on the signal processing techniques used to increase the au-dio quality of the most common digital audio effects employed in electronic musical instruments also taking into account the feasibility of the proposed algorithms’ implementation in accordance with the design constraints and the available computational limits. More in detail four different issues have been analyzed throughout this dissertation: artificial reverberation, analysis and emu-lation of nonlinear devices, audio morphing, and room response equalization.
Among the audio effects, one of the most used is definitely artificial reverberation. A great deal of research has been devoted in the last decades to improve the performance of digital artificial reverberators. Thanks to the progress of technology the traditional techniques based on recursive structures (i.e., IIR filters) are accompanied by new approaches based on fast convolution tech-niques and hybrid reverberator structures. On this basis, an efficient real-time implementation of a fast convolution algorithm has been proposed taking into account an embedded system. More-over, a technique for reducing the computational load required by this operation using psycho-acoustic expedients has been presented considering a joint assessment of the energy decay relief and the absolute threshold of hearing. Finally, some techniques for the approximation of the convolution operation with recursive structures at low computational cost have been suggested.
Although the convolution operation allows the exact reproduction of a linear system, it is im-portant to consider that most of the audio effects are nonlinear systems (i.e., compressors, distor-tion, amplifiers). For this reason, the most commonly used techniques for the emulation of non-linear systems based on a black box approach have been studied and analyzed. In particular, a technique for the approximation of the dynamic convolution operation by exploiting the princi-pal component analysis has been proposed. Using this procedure it is possible to reduce the cost of dynamic convolution without lowering the perceived audio quality. An adaptive algorithm for the identification of nonlinear systems using orthogonal functions has also been presented.
In order to provide greater flexibility and major artistic expression to musicians, several audio morphing techniques have been analyzed. In particular, this procedure makes possible to com-bine two or more audio signals in order to create new sounds that are acoustically interesting. This study has led to the development of an audio morphing algorithm for percussive hybrid sound generation. The main features of the presented approach are preprocessing of the audio references performed in the frequency domain and time domain linear interpolation to execute the morphing.
Finally, equalization techniques for improving the quality of sound reproduction systems by compensating the room transfer function have been taken into account. In particular, two algo-rithms for adaptive minimum-phase equalization and a mixed-phase equalization technique have been proposed. In order to verify the suitability of the proposed systems, experiments on a realis-tic scenario have been carried out.

This paper explains the implementation of a small-signal model for a new five-level series resonant inverter structure. The proposed circuit combines the advantages of multilevel inverters and the characteristics of the physical phenomena, resonance. The small-signal model offers the possibility to design a linearized linear time-invariant (LTI) model around an equilibrium point based on the first-order Taylor series. After that, the performance of the proposed five-level series resonant inverter and its small-signal model are evaluated in MATLAB/Simulink environment. Compared with a MATLAB model generated using the system identification toolbox, the developed small-signal model exhibited a good accuracy in frequency and time domains. In addition, a hardware test bench is developed to validate the proposed model. Both simulation and experimental results show that the proposed multilevel resonant inverter is very interesting for high-voltage high-frequency applications. As a perspective, from the results obtained, we suggest the use of small-signal AC analysis for multilevel series resonant inverter modeling. The paper contains rich information on the recently used tools for dynamic systems modeling and analysis of nonlinear processes that can be applied to modeling and analyzing other power electronics inverters.

El objetivo del siguiente documento es presentar un análisis cinemático de un robot hexápodo con el propósito de determinar su espacio de trabajo, contemplando sus restricciones articulares. Para ello se utilizarán simuladores, como el CodeWarrior y el MATLAB, que permiten operar realizar cálculos rápidamente y observar lo propuesto. Para ello, se han analizado las extremidades del robot como una cadena cinemática abierta con 3 grados de libertad cada una.
Para hacer un estudio más detallado del mismo se evidencia el método de cálculo de la cinemática inversa y del algoritmo de marcha del robot en su totalidad.

There is a fundamental conflict between algorithms which must operate in real time and algorithms which may operate out of real time. For real time operation, the compute time must be approximately constant for each sample, and the overall delay introduced by the computation must be small. We present an implementation of linear prediction coefficient calculation and fast Fourier transform computation which are "unrolled" so that a small and constant portion of the calculation is done on each sample.

The work presented in this paper focuses on the
effects of high leakage current in field effect transistors and the
possible ways to play down with the leakage currents. This paper
combines density functional theory and non equilibrium Green’s
function formalism to perform atomic scale calculation of tunnel
currents through SiO2, HfO2, Ta2O5, ZrO2 and Dy2O3 dielectrics
in MOSFETs. The tunnel currents for different bias voltages
applied to Si/Insulator/Si systems have been obtained along with
tunnel conductance v/s bias voltage plots for each system and the plots have been analyzed with reference to the presently used
bulk Si/SiO2/Si systems that have SiO2 as the gate dielectric
material. The results justify the use of high dielectric constant
materials as gate dielectric in FET devices so as to enable further
downscaling of MOSFETs with reduced gate leakage currents
thereby enabling ultra large scale integration.

— Voice recognition is the identification of a speaker on the basis of the characteristics of voices. For this, features of speech patterns that differ between individuals are used to achieve the objective. In this paper speaker recognition system are discussed. Implementation of speaker's voice recognition system with MATLAB makes possible use of voice for real life applications. This paper provides a brief review of different DSP based techniques applied for speech recognition.

This is MATLAB's 10 most easy & most basic programs that I's supposed to submit in my practicals. In this document I've complied 10 MATLAB programs from basic to advanced through intermediate levels, But overall they are for beginners only. It's only a 26 pages doc. for academic purposes. well, What else a student can offer you, huh? LOLz

In this paper, an implementation of a DC/DC buck converter for electric vehicles charging station and a DSP based closed-loop digital controller design are presented and analyzed. The aim of this work is to achieve an improved control strategy for a Li-ion battery charger implemented on a Real-time test platform. The test platform consists of a popular power pole board (MPCA75136) dedicated to studying the DC/DC converters, and a DSP development kit (TMS320F28379D) that is used to drive the DC/DC buck converter. The control strategy is based on a digital control system containing the closed-loop current controller followed by a pulse width modulation block, and on a real time state of charge estimation technique for a Li-ion battery. However, the overall control design is modeled on Simulink via block diagrams, and automatically generated code that is targeted into the DSP processor. Simulation and experimental results have shown the effectiveness of the proposed test bench and its ...

This project presents optimal design of digital FIR and IIR filters using evolutionary optimization methods. Some evolutionary optimization methods named as Normal Particle Swarm Optimization (PSO) , PSO with Constriction Factor and Inertia Weight Approach (PSOCFIWA), PSOCFIWA with Wavelet Mutation (PSOCFIWA-WM), Harmonic Search (HS), and Harmonic Search with Wavelet Mutation (HS-WM) are discussed in this project work and have been used for the optimal design of digital low pass, high pass, band pass and band stop filters. In order to show the comparative effectiveness of the discussed algorithms, the simulation results have been compared with the already existing well established results. Further to demonstrate the efficacy of the proposed methods, these have been implemented via simulink models in MATLAB.

A new control logic is constructed for bidirectional buck-boost converter (BBC) through mathematical modeling and implemented in Simulink. The BBC bridges 12 V and 24 V in dual battery automotive systems to meet the power load requirements of advanced automotive electronics. From mathematical modeling, gains of proportional, integral, derivative with filter (PIDN) control law are obtained. Mode selection circuit is developed for Buck and Boost mode control. For each mode of operation, separate PIDN control law and pulse width modulation (PWM) system is developed in control topology under voltage feedback control method. Auto mode transition is realized based on DC bus voltage. Load and line regulations in both buck and boost mode of BBC are realized based on load voltage. The constructed control logic offers accurate bidirectional voltage regulation which ensures precise power transfer in both the directions. Battery charging and discharging characteristics are realized. PIDN control law is compared with proportional, integral, derivative (PID) control law.

With the advancement of statistical methodology inextricably linked to the use of computers, new methodological ideas must be translated into usable code and then numerically evaluated relative to competing procedures. In response to this, Statistical Computing in C++ and R concentrates on the writing of code rather than the development and study of numerical algorithms per se. The book discusses code development in C++ and R and the use of these symbiotic languages in unison. It emphasizes that each offers distinct features that, when used in tandem, can take code writing beyond what can be obtained from either language alone. The text begins with some basics of object-oriented languages, followed by a " boot-camp " on the use of C++ and R. The authors then discuss code development for the solution of specific computational problems that are relevant to statistics including optimization, numerical linear algebra, and random number generation. Later chapters introduce abstract data structures (ADTs) and parallel computing concepts. The appendices cover R and UNIX Shell programming. Features • Includes numerous student exercises ranging from elementary to challenging • Integrates both C++ and R for the solution of statistical computing problems • Uses C++ code in R and R functions in C++ programs • Provides downloadable programs, available from the authors' website The translation of a mathematical problem into its computational analog (or analogs) is a skill that must be learned, like any other, by actively solving relevant problems. The text reveals the basic principles of algorithmic thinking essential to the modern statistician as well as the fundamental skill of communicating with a computer through the use of the computer languages C++ and R. The book lays the foundation for original code development in a research environment.

Convolution reverb is the process used for reverberating a signal by an impulse response of an actual space. The product of this operation is a third signal, containing reverberation of the space where the impulse response is captured. This project is based on creating a convolution reverb plugin on MaxMSP and evaluating the perceptual quality of convolution reverb.
The real-time fast convolution reverb plugin was implemented using multiple frequency delay line non-uniform partitioned convolution method on MaxMSP. An experimental design methodology was introduced in order to evaluate the perceptual quality of convolution reverb. In order to realise, anechoic drum kit music samples were recorded and re-recorded in a chamber to form the control signal. Also, using same equipment and setup, an impulse response was captured in the same chamber to form the test sample. These samples were then used for subjective analysis in a pair-wise categorical preference type listening test. The samples were also analysed in their spectrogram views in order to analyse the quality objectively.
Listening tests were designed so that realism, quality and personal preference categories were present. Null hypothesis was proved, where the realism difference of the two samples resulted in 52% of the participants preferring chamber reverb (control signal). However personal preference category resulted in 61%of the subjects preferring the chamber reverb. This was justified by objective analysis, where the convolution reverb shown to be having a faster decay rate for high frequency bands, thus sounding unnatural.

Software defined radio (SDR) technology enables implementation of wireless devices that support multiple air-interfaces and modulation formats, which is very important if consider proliferation of wireless standards. To enable such functionality SDR is using reconfigurable hardware platform such as Field Programmable Gate Array (FPGA). In this paper, we present design procedure and implementation result of SDR based QPSK modulator on Altera Cyclone IV FPGA. For design and implementation of QPSK modulator we used Altera DSP Builder Tool combined with Matlab/Simulink, Modelsim and Quartus II design tools. As reconfigurable hardware platform we used Altera DE2-115 development and education board with AD/DA daughter card. Software and Hardware-in-the-loop (HIL) simulation was conducted before hardware implementation and verification of designed system. This method of design makes implementation of SDR based modulators simpler ad faster.