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DSP Laboratory: Introduction, Hardware and Software Setup
(m13036)
Author:
Erik Luther
Keywords:
6711
,
Digital
,
dsk
,
DSP
,
labview
,
LabVIEW DSP
,
lv
,
Processing
,
Signal
,
speedy
,
speedy 33
,
texas instruments
,
ti
,
tidsk
Summary:
The purpose of this lab is to familiarize students with the DSP development workstation in the signal processing lab by examining sampling, analysis, and reconstruction of continuous-time signals. Specifically, we will first look at sampling/reconstruction of continuous-time signals. We will then examine time- and frequency-domain displays. Finally ... on aliasing.
[Expand Summary]
The purpose of this lab is to familiarize students with the DSP development workstation in the signal processing lab by examining sampling, analysis, and reconstruction of continuous-time signals. Specifically, we will first look at sampling/reconstruction of continuous-time signals. We will then examine time- and frequency-domain displays. Finally, we will examine the importance of sampling frequency and its effects on aliasing.
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
94.83%
Revised:
2006-01-24
Revisions:
2
Interactive Digital Filter Design - Online Tool for IIR Filter and FIR Filter Design
(m13115)
Authors:
Erik Luther
,
Jim Cahow
Keywords:
Digital Signal Processing
,
DSP
,
Filter Design
,
FIR Filter
,
Free
,
IIR Filter
,
LabVIEW
,
Signal Processing
,
SP
,
Utility
Summary:
The Digital Filter Design Utility is software that allows you to interactively specify and design floating point digital filters. This utility was created with LabVIEW and the Digital Filter Design Toolkit and compiled into a stand-alone application for your use. The full Digital Filter Design Toolkit contains a complete ... programming language.
[Expand Summary]
The Digital Filter Design Utility is software that allows you to interactively specify and design floating point digital filters. This utility was created with LabVIEW and the Digital Filter Design Toolkit and compiled into a stand-alone application for your use. The full Digital Filter Design Toolkit contains a complete set of filter design tools for designing a wide range of floating point and fixed point digital filters within an open environment, using an easy-to-learn interactive graphical programming language.
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
98.56%
Revised:
2007-08-20
Revisions:
10
A Primer on Besov Spaces
(col10679)
Author:
Albert Cohen
Keywords:
Besov
,
Classical Measures
,
Primer
,
smoothness
,
Spaces
Subject:
Mathematics and Statistics,
Science and Technology
Language:
English
Popularity:
49.38%
Revised:
2009-09-09
Revisions:
2
Multiresolution Analysis, Filterbank Implementation, and Function Approximation using Wavelets
(col10568)
Author:
Veronique Delouille
Language:
English
Popularity:
31.39%
Revised:
2009-09-14
Revisions:
2
A Wavelet Tour of Signal Processing, The Sparse Way
(col10711)
Author:
Stephane Mallat
Subject:
Mathematics and Statistics,
Science and Technology
Language:
English
Popularity:
47.41%
Revised:
2009-09-14
Revisions:
3
Musical Signal Processing with LabVIEW -- Programming Techniques for Audio Signal Processing
(col10440)
Author:
Ed Doering
Keywords:
arrays
,
audio files
,
audio input/output
,
editing tips
,
LabVIEW tutorial
,
programming structures
Summary:
After completing this multi-media course you will be well-equipped to start creating your own audio and signal processing applications within the LabVIEW development environment. The modules in this course make extensive use of "screencasts" -- videos captured directly from the computer screen that show the LabVIEW tool in operation ... with LabVIEW".
[Expand Summary]
After completing this multi-media course you will be well-equipped to start creating your own audio and signal processing applications within the LabVIEW development environment. The modules in this course make extensive use of "screencasts" -- videos captured directly from the computer screen that show the LabVIEW tool in operation, with audio narration to explain each step. The course includes a "Getting Started" tutorial, editing tips, essential programming structures, subVIs, arrays, audio sources, audio output to the soundcard, reading and writing audio files, and real-time audio output with interactive parameter control. This course is part of the series "Musical Signal Processing with LabVIEW".
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
93.95%
Revised:
2007-07-18
Revisions:
New
Signals and Systems
(col10064)
Author:
Richard Baraniuk
Institution:
Rice University
Summary:
This course deals with signals, systems, and transforms, from their theoretical mathematical foundations to practical implementation in circuits and computer algorithms. At the conclusion of ELEC 301, you should have a deep understanding of the mathematics and practical issues of signals in continuous and discrete time, linear time invariant systems, convolution, and Fourier transforms.
Subject:
Science and Technology
Language:
English
Popularity:
99.64%
Revised:
2007-07-23
Revisions:
11
Adaptive Filters
(col10280)
Author:
Douglas L. Jones
Institution:
University of Illinois Urbana-Champaign
Keywords:
Adaptive
,
Filters
,
Wiener
,
LMS
Summary:
Covers different types of adaptive filters and their implementations. In depth analysis is done of the Wiener Filter algorithm and the LMS algorithm.
Subject:
Science and Technology
Language:
English
Popularity:
97.04%
Revised:
2005-05-12
Revisions:
New
Frequency and Music
(col10338)
Authors:
Douglas L. Jones
,
Catherine Schmidt-Jones
Institution:
University of Illinois at Urbana-Champaign
Keywords:
frequency
,
harmonics
,
music
Summary:
An overview of frequency, harmonic (Fourier) series, and their relationship to music.
Subject:
Science and Technology
Language:
English
Popularity:
89.73%
Revised:
2006-02-21
Revisions:
New
An Introduction to Wavelet Analysis
(col10566)
Author:
Veronique Delouille
Language:
English
Popularity:
42.13%
Revised:
2009-09-14
Revisions:
3
Notes on the Design of Optimal FIR Filters
(col10553)
Author:
John Treichler
Keywords:
digital signal processing
,
FIR
,
FIR digital filters
,
Parks-McClellan
Summary:
A recurring technical task development of digital signal processing products and systems is the design of FIR digital filters. Fortunately some excellent software packages exist for the automatic synthesis of impulse responses for such filters, many of them based on the now-famous Parks-McClellan algorithm. Unfortunately, there is still ... is used.
[Expand Summary]
A recurring technical task development of digital signal processing products and systems is the design of FIR digital filters. Fortunately some excellent software packages exist for the automatic synthesis of impulse responses for such filters, many of them based on the now-famous Parks-McClellan algorithm. Unfortunately, there is still some mystery about how to use the software and, equally important, how to estimate impulse response lengths short of actually designing the filter itself. This technical note primarily addresses the second problem and indirectly discusses a bit the first. We examine here how to convert a typical filter specification in terms of cutoff frequency, passband ripple, etc., into a reasonably accurate estimate of the length of the impulse response. Not only does this estimate suffice for most design tradeoff exercises, it usually allows the Parks-McClellan routines to be employed only once or twice rather than the multiple times needed when the “cut-and-try" method is used.
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
50.36%
Revised:
2009-09-14
Revisions:
3
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