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Matthew Berry
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Adaptive Filtering: LMS Algorithm
(m10481)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Dima Moussa
,
Daniel Sachs
Keywords:
adaptive filtering
,
DSP
,
gradient descent
,
LMS
,
system identification
Summary:
This module introduces adaptive filters through the example of system identification using the LMS algorithm. The adaptive filter adjusts its coefficients to minimize the mean-square error between its output and that of an unknown system.
Subject:
Science and Technology
Language:
English
Popularity:
99.36%
Revised:
2009-06-01
Revisions:
15
Using the Serial Port with a MATLAB GUI
(m12062)
Authors:
Jason Laska
,
Matthew Berry
,
Daniel Sachs
Keywords:
DSP
,
GUI
,
Matlab
,
Serial Port
Summary:
Explains how to send and receive data with the DSP through the serial port. Example code is shown in both assembly and C (for the DSP) as well as MATLAB for interfacing on the PC. Some example code for creating a MATLAB GUI is also shown.
Subject:
Science and Technology
Language:
English
Popularity:
99.30%
Revised:
2004-06-25
Revisions:
New
Spectrum Analyzer: MATLAB Exercise
(m12379)
Authors:
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Jake Janovetz
,
Michael Kramer
,
Dima Moussa
,
Daniel Sachs
,
Brian Wade
,
Matt Kleffner
,
Douglas L. Jones
Keywords:
autocorrelation
,
bit-reversed
,
boxcar
,
DFT
,
Discrete Fourier Transform
,
Discrete Time Fourier Transform
,
DSP
,
DTFT
,
Fast Fourier transform
,
FFT
,
Fourier transform
,
hamming
,
mainlobe
,
Power Spectra
,
Power Spectral Density Estimate
,
PSD
,
sidelobe
,
twiddle-factor
,
windowing
,
zero-pad
Summary:
You will investigate the effects of windowing and zero-padding on the Discrete Fourier Transform of a signal, as well as the effects of data-set quantities and weighting windows used in Power Spectral Density estimation.
Subject:
Science and Technology
Language:
English
Popularity:
98.67%
Revised:
2004-09-23
Revisions:
2
IIR Filtering: Filter-Design Exercise in MATLAB
(m10623)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Jake Janovetz
,
Michael Kramer
,
Dima Moussa
,
Daniel Sachs
,
Brian Wade
Keywords:
conv
,
difference equation
,
direct fortm II
,
DSP
,
ellip
,
elliptic low-pass filter
,
freqz
,
gain factor
,
IIR
,
impulse response
,
infinite impulse response
,
linear time-invariant
,
LTI
,
notch filter
,
overflow
,
poles
,
zeros
Summary:
You will derive the transfer function of a second-order, Direct Form II, infinite impulse response (IIR) filter. Then you will create a fourth-order IIR filter, plot its frequency response, and decompose the fourth-order filter into two second-order sections, choosing an appropriate gain for each stage to prevent overflow.
Subject:
Science and Technology
Language:
English
Popularity:
98.55%
Revised:
2004-02-25
Revisions:
12
Speech Processing: Theory of LPC Analysis and Synthesis
(m10482)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Jake Janovetz
,
Michael Kramer
,
Dima Moussa
,
Daniel Sachs
,
Brian Wade
Keywords:
autocorrelation
,
autocovariance
,
correlation
,
cross-correlation
,
DSP
,
levinson-durbin
,
linear predicitive coding
,
speech
,
speech coding
,
speech compression
,
speech synthesis
Summary:
Speech analysis and synthesis with Linear Predictive Coding (LPC) exploit the predictable nature of speech signals. Cross-correlation, autocorrelation, and autocovariance provide the mathematical tools to determine this predictability. If we know the autocorrelation of the speech sequence, we can use the Levinson-Durbin algorithm to find an efficient solution ... the speech.
[Expand Summary]
Speech analysis and synthesis with Linear Predictive Coding (LPC) exploit the predictable nature of speech signals. Cross-correlation, autocorrelation, and autocovariance provide the mathematical tools to determine this predictability. If we know the autocorrelation of the speech sequence, we can use the Levinson-Durbin algorithm to find an efficient solution to the least mean-square modeling problem and use the solution to compress or resynthesize the speech.
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
98.29%
Revised:
2009-06-01
Revisions:
20
Digital Transmitter: Introduction to Quadrature Phase-Shift Keying
(m10042)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Jake Janovetz
,
Michael Kramer
,
Dima Moussa
,
Daniel Sachs
,
Brian Wade
Keywords:
DSP
,
gray coding
,
in-phase signal
,
pseudo-noise generator
,
QPSK
,
quadrature phase-shift keying
,
quadrature signal
,
signal constellation
Summary:
Quadrature phase shift keying (QPSK) is a method for transmitting digital data across an analog channel. Data bits are grouped into pairs and represented by a unique waveform, called a symbol. Data may be simulated with a pseudo-noise sequence generator.
Subject:
Science and Technology
Language:
English
Popularity:
97.89%
Revised:
2004-02-25
Revisions:
20
Two's Complement and Fractional Arithmetic for 16-bit Processors
(m10808)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Dima Moussa
,
Daniel Sachs
,
Jason Laska
Keywords:
DSP
,
fractional arithmetic
,
overflow
,
two's complement
Summary:
Two's-complement notation is a mathematically convenient way of representing signed numbers in microprocessors. The most significant bit of a two's complement number represents its sign, and the remaining bits represent its magnitude. Fractional arithmetic allows one to multiply numbers on an integer processor without incurring overflow. Fractional ... one bit.
[Expand Summary]
Two's-complement notation is a mathematically convenient way of representing signed numbers in microprocessors. The most significant bit of a two's complement number represents its sign, and the remaining bits represent its magnitude. Fractional arithmetic allows one to multiply numbers on an integer processor without incurring overflow. Fractional arithmetic requires sign-extension of multipliers and multiplicands, and it requires the product of two numbers to be left-shifted one bit.
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
97.54%
Revised:
2005-01-30
Revisions:
10
Digital Receivers: Symbol-Timing Recovery for QPSK
(m10485)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Jake Janovetz
,
Michael Kramer
,
Dima Moussa
,
Daniel Sachs
,
Brian Wade
Keywords:
delay-locked loop
,
DSP
,
matched filter
,
noise
,
quadrature phase-shift keying
,
receiver
Summary:
The goal of symbol-timing recovery is to sample message signals at the receiver for best performance. After the in-phase and quadrature signals pass through a matched filter, a delay-locked loop attempts to find the peaks in the output waveforms.
Subject:
Science and Technology
Language:
English
Popularity:
97.48%
Revised:
2005-07-29
Revisions:
15
Digital Receiver: Carrier Recovery
(m10478)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Dima Moussa
,
Daniel Sachs
Keywords:
BPSK
,
carrier recovery
,
coherent demodulation
,
digital communications
,
DSP
,
interpolation
,
numerically-controlled oscillator
,
phase-locked loop
,
QPSK
,
voltage-controlled oscillator
Summary:
The phase-locked loop (PLL) is a critical component in coherent communications receivers that is responsible for locking on to the carrier of a received modulated signal. A PLL adjusts the phase of a numerically-controlled oscillator to match that of the received signal. You will simulate a carrier recovery ... the DSP.
[Expand Summary]
The phase-locked loop (PLL) is a critical component in coherent communications receivers that is responsible for locking on to the carrier of a received modulated signal. A PLL adjusts the phase of a numerically-controlled oscillator to match that of the received signal. You will simulate a carrier recovery sub-system in MATLAB and then implement the sub-system on the DSP.
[Collapse Summary]
Subject:
Science and Technology
Language:
English
Popularity:
97.14%
Revised:
2009-06-03
Revisions:
17
Speech Processing: LPC Exercise in MATLAB
(m10824)
Authors:
Douglas L. Jones
,
Swaroop Appadwedula
,
Matthew Berry
,
Mark Haun
,
Jake Janovetz
,
Michael Kramer
,
Dima Moussa
,
Daniel Sachs
,
Brian Wade
Keywords:
autocorrelation
,
DSP
,
levinson-durbin algorithm
,
linear predictive coding
,
speech
,
speech analysis
,
speech coding
,
speech compression
,
speech synthesis
,
xcorr
Summary:
You will write MATLAB code to compute the autocorrelation sequence of a simple signal. Then you will implement the Levinson-Durbin algorithm in MATLAB and analyze a recording of your own voice.
Subject:
Science and Technology
Language:
English
Popularity:
96.98%
Revised:
2004-02-25
Revisions:
6
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