Relevant bibliographies by topics / Degree Discipline: Digital Signal Processing

Academic literature on the topic 'Degree Discipline: Digital Signal Processing'

Author: Grafiati
Published: 10 December 2022
Last updated: 28 January 2023

Create a spot-on reference in APA, MLA, Chicago, Harvard, and other styles

Select a source type:

Contents

Consult the lists of relevant articles, books, theses, conference reports, and other scholarly sources on the topic 'Degree Discipline: Digital Signal Processing.'

Next to every source in the list of references, there is an 'Add to bibliography' button. Press on it, and we will generate automatically the bibliographic reference to the chosen work in the citation style you need: APA, MLA, Harvard, Chicago, Vancouver, etc.

You can also download the full text of the academic publication as pdf and read online its abstract whenever available in the metadata.

Journal articles on the topic "Degree Discipline: Digital Signal Processing"

1

Rawski, Mariusz, Bogdan Falkowski, and Tadeusz Łuba. "Digital signal processing designing for FPGA architectures." Facta universitatis - series: Electronics and Energetics 20, no. 3 (2007): 437–59. http://dx.doi.org/10.2298/fuee0703437r.

Full text
Abstract:
This paper presents the discussion on efficiency of different implementation methodologies of DSP algorithms targeted for modern FPGA architectures. Modern programmable structures are equipped with specialized DSP embedded blocks that allow implementing digital signal processing algorithms with use of the methodology known from digital signal processors. On the first place however, programmable architectures give the designer the possibility to increase efficiency of designed system by exploitation of parallelism of implemented algorithms. Moreover, it is possible to apply special techniques such as distributed arithmetic (DA) that will boost the performance of designed processing systems. Additionally, application of the functional decomposition based methods, known to be best suited for FPGA structures allows utilizing possibilities of programmable technology in very high degree. The paper presents results of comparison of different design approaches in this area.
APA, Harvard, Vancouver, ISO, and other styles
2

Sangwine, S. J. "A Digital Signal Processing Laboratory Based on the TMS320C25." International Journal of Electrical Engineering & Education 32, no. 1 (January 1995): 21–30. http://dx.doi.org/10.1177/002072099503200103.

Full text
Abstract:
A digital signal processing laboratory based on the TMS320C25 Students on a B. Eng. degree course at the University of Reading take a 20 hour lecture course on DSP and 15 hours of laboratory work using an audio-band DSP system designed around the Texas TMS320C25 DSP chip. The course and DSP system are described and experiences and conclusions are drawn.
APA, Harvard, Vancouver, ISO, and other styles
3

Barnard, Andrew, and Daniel A. Russell. "The graduate program in acoustics at Penn State." Journal of the Acoustical Society of America 152, no. 4 (October 2022): A124. http://dx.doi.org/10.1121/10.0015762.

Full text
Abstract:
The Graduate Program in Acoustics at Penn State offers graduate degrees (M.Eng., M.S., Ph.D.) in Acoustics, with courses and research opportunities in a wide variety of subfields. Our 820 alumni are employed around the world in a wide variety of military and government labs, academic institutions, consulting firms, and consumer audio and related industries. Our 40+ faculty from several disciplines conduct research and teach courses in structural acoustics, nonlinear acoustics, architectural acoustics, signal processing, aeroacoustics, biomedical ultrasound, transducers, computational acoustics, noise and vibration control, acoustic metamaterials, psychoacoustics, and underwater acoustics. Course offerings include fundamentals of acoustics and vibration, electroacoustic transducers, signal processing, acoustics in fluid media, sound and structure interaction, digital signal processing, experimental techniques, acoustic measurements and data analysis, ocean acoustics, architectural acoustics, noise control engineering, nonlinear acoustics, outdoor sound propagation, computational acoustics, biomedical ultrasound, flow induced noise, spatial sound and three-dimensional audio, and the acoustics of musical instruments. This poster highlights faculty research areas, laboratory facilities, student demographics, successful graduates, and recent enrollment and employment trends for the Graduate Program in Acoustics at Penn State.
APA, Harvard, Vancouver, ISO, and other styles
4

Saber, Mohamed, Mohamed E. Ghoneim, and Sunil Kumar. "Survey on Design of Digital FIR Filters using Optimization Models." Journal of Artificial Intelligence and Metaheuristics 2, no. 1 (2022): 16–26. http://dx.doi.org/10.54216/jaim.020102.

Full text
Abstract:
As the discipline of Digital Signal Processing develops, digital filters play an increasingly vital role in modern technology (DSP). The FIR filter, which stands for finite impulse response, is the most common type of filter. As a result of its versatility, FIR filters find widespread application in many fields, including image filtering, frequency modulation, precision arithmetic, and many more. For this reason, digital FIR filters are designed using various optimization techniques. Using various optimization strategies yields the best results when optimizing for different filter coefficients (concerning control parameters, dependence, premature convergence, etc.). They're advantageous due to several factors, including their straightforward implementation, low error function, high-quality searching ability, and rapid convergence. In this paper, we have covered the topic of designing efficient digital filters for signal, image, and video processing using various optimization techniques.
APA, Harvard, Vancouver, ISO, and other styles
5

Yu, Yan Xin, Chun Yang Wang, Yu Chen, and Ke Yang. "Design of Digital Pulse Compression System Based on FPGA." Advanced Materials Research 1049-1050 (October 2014): 1718–21. http://dx.doi.org/10.4028/www.scientific.net/amr.1049-1050.1718.

Full text
Abstract:
Pulse compression technology is one of the key technologies in the field of modern radar signal processing, can effectively solve the contradiction between action distance and resolution. In this paper, a radar digital pulse compression system is designed and implemented based on FPGA with linear frequency modulated signal. The digital pulse compression module is designed using FFT IP core which can be reused in different periods of DPC, respectively performing FFT and IFFT calculation, so that the hardware consumption is saved significantly. Therefore, compared with other systems, the system designed in this paper has the characters of fast processing speed, high degree of modularity, real-time processing and short development cycle.
APA, Harvard, Vancouver, ISO, and other styles
6

Shchukin, A. A., and A. E. Pavlov. "Parameterization of user functions in digital signal processing for obtaining angular superresolution." Russian Technological Journal 10, no. 4 (July 30, 2022): 38–43. http://dx.doi.org/10.32362/2500-316x-2022-10-4-38-43.

Full text
Abstract:
Objectives. One of the most important tasks in the development of goniometric systems is improving resolution in terms of angular coordinates. This can be achieved in two ways: firstly, by increasing the aperture, which is very expensive and often technically challenging to implement; secondly, with the help of digital signal processing methods. If the recorded signal sources are located close to each other and not resolved by the Rayleigh criterion, it can be impossible to determine their number, location and reflection characteristics. The aim of the present work is to develop a digital signal processing algorithm for obtaining angular superresolution.Methods. Mathematical methods for solving inverse problems are used to overcome the Rayleigh criterion, i.e., obtain angular superresolution. These problems are unstable, since there is an infinite number of approximate solutions and false targets may occur. The search for the optimal solution is carried out by minimizing the standard deviation.Results. A description of a mathematical model for a goniometric system is presented. A signal processing algorithm is developed based on existing methods according to the principle of parameterization of user functions. Results of numerical experiments for achieving superresolution by algebraic methods are given along with an estimation of solution stability. The accuracy and correspondence of the amplitude of the obtained objects to the initial parameters are measured. The degree of excess of the Rayleigh criterion by the obtained solution is estimated.Conclusions. Algebraic methods can be used to obtain stable solutions with angular superresolution. The results obtained correctly reflect the location of objects with a minor error. Errors in the distribution of the signal amplitude are small, appearing false targets have negligible amplitude.
APA, Harvard, Vancouver, ISO, and other styles
7

A. Asker, Mshari, Khalaf S. Gaeid, Nada N. Tawfeeq, Humam K. Zain, Ali I. Kauther, and Thamir Q Abdullah. "Design and Analysis of Robot PID Controller Using Digital Signal Processing Techniques." International Journal of Engineering & Technology 7, no. 4.37 (December 13, 2018): 103. http://dx.doi.org/10.14419/ijet.v7i4.37.23625.

Full text
Abstract:
Recently robotic is a playing vital role in the life In our modern society, the usage of robotic arms are increasing and much of the work in the industry is now performed by robots. As robots begin to behave like humans in an intelligent manner, control system becomes a major concern. In this paper, design and analyses of the pick and place robot due to control, the forearm, wrist, desired turntable and desired bicep is introduced to construct a closed system with four degrees of freedom (4DOFs). The main performance specifications are the accuracy and stability of the input system for obtaining a good system performance. Implementation of the control system using PID parameters for stability, minimum steady state error, minimum overshoot and faster system response has been carried out. The design of two degree of freedom PID(2DoFPID) to control robotic arm along with first order low pass filter(LPF) to compensate the unwanted signal is improved. To be able to implement such a precise and effective system, feedback system has to be made to improve the overall performance specifications. The digital signal processing controller (Arduino Uno) is used as it is active, cheap , it has open source code and easy to use in the software and hardware applications.Experimental set up developed in addition to the Matlab/Simulink implementation of the complete system. The results and the communication signals test ensure smooth operation of the control system and the effectiveness of the proposed algorithm.
APA, Harvard, Vancouver, ISO, and other styles
8

Wang, Jun Hong, Zong Rui Li, and Xi Bin Wang. "Application Research of Visual Processing Technology in the Industrial Production Line." Applied Mechanics and Materials 563 (May 2014): 338–41. http://dx.doi.org/10.4028/www.scientific.net/amm.563.338.

Full text
Abstract:
The CCD image sensor is set in the different production position, whose output signal is converted into digital signals to a dedicated image processing system by A/D. Using the image enhancement, smoothing, sharpening, segmentation, feature extraction, image recognition and understanding of digital image processing techniques,the system can identify the image, compare with feature information preservation, decide whether to enter the next process according to the similarity degree of alignment. Visual inspection having high precision, fast speed, working in the industrial field is stable and reliable, and improves the level of automation of production, make the products more competitive.
APA, Harvard, Vancouver, ISO, and other styles
9

Hans, Volker, and C. Filips. "Detection of Vortex Frequency in Gas Flow with Ultrasound." Key Engineering Materials 295-296 (October 2005): 515–20. http://dx.doi.org/10.4028/www.scientific.net/kem.295-296.515.

Full text
Abstract:
The coincidence of vortices generated by a bluff body in a gaseous flow (Karman vortex street) with an ultrasonic beam crossing these vortices raises a lot of questions concerning physics and signal processing. The ultrasonic signal will be complex modulated. The spectrum of the resulted signal shows the carrier frequency of ultrasound and two narrow sidebands with the information about the modulation. For further signal processing the carrier frequency must be filtered. The carrier frequency can be shifted to zero by digital processing and undersampling the signal by an integer multiple. Then the sideband with its low frequency range can be analysed. The real and imaginary parts of the signal can be determined by sampling the signal shifted by 90 degrees (Hilbert transform). Even the 90 degree shifted angle can be measured by undersampling. The sensitivity of the vortex meter depends on the bluff body size. A simple relation between the bluff body dimension and the sensitivity, the vortex frequency, respectively, is shown.
APA, Harvard, Vancouver, ISO, and other styles
10

Saidov, B. B., V. F. Telezhkin, N. N. Gudaev, V. N. Bagaev, and M. A. Devyatov. "Development of equipment for experimental study of digital algorithms in nonstationary signal processing problems." Ural Radio Engineering Journal 6, no. 2 (2022): 186–204. http://dx.doi.org/10.15826/urej.2022.6.2.004.

Full text
Abstract:
The article deals with the issues of improving the quality of information using ultrasound, as well as existing analog and digital methods of its processing. The characteristics of signal filtering are analyzed depending on the signal-to-noise ratio. A number of modern applications in medical diagnostics are subject of high requirements for accuracy, noise immunity, reliability, continuity of operation and other quality indicators with a high dynamics of changes in non-stationary signals characterizing the functional state of the human body. On this basis, an algorithm and a device capable of extracting the informative components of noisy non-stationary signals are developed. In this paper, an algorithm of processing non-stationary signals of an ultrasonic transceiver is proposed. At the same time, three modules are developed and manufactured: 1) ultrasonic receiver module; 2) ultrasonic noise transmitter module (AM modulation); 3) an ultrasonic signal transmitter module (AM modulation) and an information processing algorithm based on a wavelet-forming function (wavelet threshold) consisting of the Coiflets 5 basis using a heuristically determined threshold value of the wavelet expansion coefficients. In this paper, the wavelet decomposition is carried out up to the 4th level. Based on the analysis of the obtained data, it was concluded that the second decomposition level is the most optimal for filtering non-stationary signals. As the decomposition level increases, the output signal-to-noise ratio decreases, and at the level N = 4, the output signal-to-noise ratio almost does not exceed the input one, therefore, filtering becomes inefficient. As a result of the synthesis of effective parameters of filtering electrocardiosignals, it is found that the maximum degree of signal processing from noise occurs using the Coiflets 5 wavelet using a heuristically determined threshold value. Experimental research is carried out in the training and production laboratory of electronics (FabLab), FSAEI HE “South Ural State University (National Research University)”.
APA, Harvard, Vancouver, ISO, and other styles
More sources

Dissertations / Theses on the topic "Degree Discipline: Digital Signal Processing"

1

Huang, Beilei. "A resampling theory for non-bandlimited signals and its applications : a thesis presented for the partial fulfillment of the requirements for the degree of Doctor of Philosophy in Engineering at Massey University, Wellington, New Zealand." 2008. http://hdl.handle.net/10179/773.

Full text
Abstract:
Currently, digital signal processing systems typically assume that the signals are bandlimited. This is due to our knowledge based on the uniform sampling theorem for bandlimited signals which was established over 50 years ago by the works of Whittaker, Kotel'nikov and Shannon. However, in practice the digital signals are mostly of finite length. This kind of signals are not strictly bandlimited. Furthermore, advances in electronics have led to the use of very wide bandwidth signals and systems, such as Ultra-Wide Band (UWB) communication systems with signal bandwidths of several giga-hertz. This kind of signals can effectively be viewed as having infinite bandwidth. Thus there is a need to extend existing theory and techniques for signals of finite bandwidths to that for non-bandlimited signals. Two recent approaches to a more general sampling theory for non-bandlimited signals have been published. One is for signals with finite rate of innovation. The other introduced the concept of consistent sampling. It views sampling and reconstruction as projections of signals onto subspaces spanned by the sampling (acquisition) and reconstruction (synthesis) functions. Consistent sampling is achieved if the same discrete signal is obtained when the reconstructed continuous signal is sampled. However, it has been shown that when this generalized theory is applied to the de-interlacing of video signals, incorrect results are obtained. This is because de-interlacing is essentially a resampling problem rather than a sampling problem because both the input and output are discrete. While the theory for the resampling for bandlimited signals is well established, the problem of resampling without bandlimited constraints is largely unexplored. The aim of this thesis is to develop a resampling theory for non-bandlimited discrete signals and explore some of its potential applications. The first major contribution is the the theory and techniques for designing an optimal resampling system for signals in the general Hilbert Space when noise is not present. The system is optimal in the sense that the input of the system can always be obtained from the output. The theory is based on the concept of consistent resampling which means that the same continuous signal will be obtained when either the original or the resampled discrete signal is presented to the reconstruction filter. While comparing the input and output of a sampling/reconstruction system is relatively simple since both are continuous signals, comparing the discrete input and output of a resampling system is not. The second major contribution of this thesis is the proposal of a metric that allows us to evaluate the performance of a resampling system. The performance is analyzed in the Fourier domain as well. This performance metric also provides a way by which different resampling algorithms can be compared effectively. It therefore facilitates the process of choosing proper resampling schemes for a particular purpose. Unfortunately consistent resampling cannot always be achieved if noise is present in the signal or the system. Based on the performance metric proposed, the third major contribution of this thesis is the development of procedures for designing resampling systems in the presence of noise which is optimal in the mean squared error (MSE) sense. Both discrete and continuous noise are considered. The problem is formulated as a semi-definite program which can be solved effciently by existing techniques. The usefulness and correctness of the consistent resampling theory is demonstrated by its application to the video de-interlacing problem, image processing, the demodulation of ultra-wideband communication signals and mobile channel detection. The results show that the proposed resampling system has many advantages over existing approaches, including lower computational and time complexities, more accurate prediction of system performances, as well as robustness against noise.
APA, Harvard, Vancouver, ISO, and other styles

Book chapters on the topic "Degree Discipline: Digital Signal Processing"

1

Milic, Ljiljana. "IIR Filters to Sampling Rate Conversion." In Multirate Filtering for Digital Signal Processing, 136–70. IGI Global, 2009. http://dx.doi.org/10.4018/978-1-60566-178-0.ch005.

Full text
Abstract:
Infinite impulse response (IIR) filters are used in applications where the computational efficiency is the highest priority. It is well known that an IIR filter transfer function is of a considerably lower order than the transfer function of an FIR equivalent. The drawbacks of an IIR filter are the nonlinear phase characteristic and sensitivity to quantization errors. In multirate applications, the computational requirements for FIR filters can be reduced by the sampling rate conversion factor as demonstrated in Chapter IV. However, such a degree of computation savings cannot be achieved in multirate implementations of IIR filters. This is due to the fact that every sample value computed in the recursive loop is needed for evaluating an output sample. Based on the polyphase decomposition, several techniques have been developed which improve the efficiency of IIR decimators and interpolators as will be shown later on in this chapter. In this chapter, we consider first the direct implementation structures for IIR decimators and interpolators. In the sequel, we demonstrate the computational requirements for direct form IIR decimators and interpolators. The polyphase decomposition of an IIR transfer function is explained with its application to decimation and interpolation. Then, we demonstrate an efficient IIR polyphase structure based on all-pass subfilters, which is applicable to a restricted class of decimators and interpolators. In this chapter, we discuss the application of the elliptic minimal Q factor (EMQF) filter transfer function in constructing high-performance decimators and interpolators. The chapter concludes with a selection of MATLAB exercises for the individual study.
APA, Harvard, Vancouver, ISO, and other styles

Conference papers on the topic "Degree Discipline: Digital Signal Processing"

1

Sanderson, J. "Experiences of teaching digital signal processing at degree level." In IEE Colloquium on The Teaching of Digital Signal Processing (DSP) in Universities. IEE, 1995. http://dx.doi.org/10.1049/ic:19950216.

Full text
APA, Harvard, Vancouver, ISO, and other styles
2

Burrus, C., James Fox, Gary Sitton, and Sven Treitel. "A Parallel Version of the Lindsey-Fox algorithm for factoring High Degree Polynomials in Signal Processing." In 2006 IEEE 12th Digital Signal Processing Workshop & 4th IEEE Signal Processing Education Workshop. IEEE, 2006. http://dx.doi.org/10.1109/dspws.2006.265415.

Full text
APA, Harvard, Vancouver, ISO, and other styles
3

Singh, Ravinder, K. Roy, B. R. Kapuriya, and C. Bhattacharya. "A ROM-less direct digital frequency synthesizer based on fifth-degree Bezier curve approximation." In 2013 International Conference on Intelligent Systems and Signal Processing (ISSP). IEEE, 2013. http://dx.doi.org/10.1109/issp.2013.6526866.

Full text
APA, Harvard, Vancouver, ISO, and other styles
4

Zheng, Yifeng, and Jianxiu Jin. "A novel image scrambling degree blind evaluation scheme based on Bhattacharyya coefficient." In 2014 9th International Symposium on Communication Systems, Networks & Digital Signal Processing (CSNDSP). IEEE, 2014. http://dx.doi.org/10.1109/csndsp.2014.6923817.

Full text
APA, Harvard, Vancouver, ISO, and other styles
5

Higaki, Y., K. Higuchi, K. Nakano, T. Kajikawa, and Kosin Chamnongthai. "Design method of a 1-bit digital filter using a new 2-Degree-of-Freedom control." In 2008 International Symposium on Intelligent Signal Processing and Communications Systems (ISPACS 2008). IEEE, 2009. http://dx.doi.org/10.1109/ispacs.2009.4806737.

Full text
APA, Harvard, Vancouver, ISO, and other styles
6

Dai, Chenyan. "An Intelligent Evaluation Algorithm for the Matching Degree of Music Lyrics Based on LabVIEW Digital Image." In ICGSP 2022: 2022 The 6th International Conference on Graphics and Signal Processing. New York, NY, USA: ACM, 2022. http://dx.doi.org/10.1145/3561518.3561524.

Full text
APA, Harvard, Vancouver, ISO, and other styles
You might also be interested in the extended bibliographies on the topic 'Degree Discipline: Digital Signal Processing' for particular source types:
Journal articles
We offer discounts on all premium plans for authors whose works are included in thematic literature selections. Contact us to get a unique promo code!

To the bibliography