MAE 2381: Chapters 6 & 7

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Chapter 6: Windowing and Filtering - Chapter 7: Sampling

Last updated 2:07 PM on 10/7/26
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16 Terms

1
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sampling rate

how often a sample is taken (a frequency, Hz)

fs = 1/δt where:

fs = sampling rate, Hz δt = time between samples, s


if sampling rate is too low, then the sampled signal can appear lower than the actual signal (aliasing)

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nyquist frequency

the maximum frequency that can be detected for a given sampling rate

fmax = fN = fs/2 = 1/2δt where:

fN = Nyquist frequency

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minimum sample rate

to accurately reconstruct the frequency content of a measured signal

fs >= 2fmax where:

fs = minimum sample rate

fmax = maximum frequency of signal

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maximum sample time increment

to accurately reconstruct the frequency content of a measured signal:

δts < 1/2fmax where:

δts = maximum sample time increment

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alias frequency

a sampling rate too low causes frequencies to be misinterpreted → this false/lower frequency is called ____

<p>a sampling rate too low causes frequencies to be misinterpreted → this false/lower frequency is called ____</p>
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folding diagram

a diagram in which the original input frequency axis is folded back over itself at the folding point of the Nyquist frequency and again for each of its harmonics

<p>a diagram in which the original input frequency axis is folded back over itself at the folding point of the Nyquist frequency and again for each of its harmonics</p>
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gain

ratio between the amplified amplitude (Aa) and the original amplitude (A0)
G = Aa/A0

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magnitude ratio (M(w))

the degree of modification (ratio) when a signal’s amplitude is modified by some process (like filtering or a system response)

M(w) = Mm/M0 where:

Mm = modified magnitude

M0 = original magnitude

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windowing

cleans up data in the time domain to reduce the effect of clipping

basically like zooming in on a section of data

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clipping

incomplete sinusoid periods in sample windows

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rectangular window

1 = keep the data 0 = throw it away

fixed interval: multiply desired data by 1 and the signals before/after it by 0

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filtering

removes unwanted frequencies from a signal (in the frequency domain)

you have to convert everything from the time domain to the frequency domain to filter

  • an analog electronic circuit can filter the signal before sampling


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Hanning window

convert the data window into a periodic function that begins and ends at y = 0

  • reduces the clipping effect of the rectangular window by multiplying discrete data by a “Hanning” window-like function


<p>convert the data window into a periodic function that begins and ends at y = 0</p><ul><li><p>reduces the <strong>clipping </strong>effect of the <strong>rectangular window</strong> by multiplying discrete data by a “Hanning” window-like function</p></li></ul><p></p>
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low-pass filter

keeps all low frequencies, removes high frequencies

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high-pass filter

keep all high frequencies, removes low frequencies

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band-pass filter

allows a specific range of frequencies to pass through while blocking frequencies both above and below that range