Sampling - week 4
- Created by: harveyf2801
- Created on: 14-01-21 00:56
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- Sampling
- Turns a signal into a sequence of numbers that approximately represent it
- Digital over analogue
- Easy duplication with no further errors or loss of quality
- More durable mediums can store the data
- Can perform signal processing
- Cheaper to run
- Analogue over digital
- Analogue varies continuously in time whereas digital systems contain discrete numbers
- Therefore errors occur due to approximation / 'loss' of data between sample points
- Analogue varies continuously in time whereas digital systems contain discrete numbers
- Nyquist theorem
- The signal must be sampled at at least twice the highest frequency
- Sample rate = Fs = 2 * fmax
- Nyquist frequency = fN = Fs / 2
- For some practical applications the signal should be sampled at 5 or 10 times the highest frequency
- Aliasing will occur if the audio isn't sampled at twice the Nyquist frequency
- The aliased frequency can be calculated with this equation
- F alias = abs( ¦ Fs - Input F ¦ )
- Aliasing always creates a frequency lower than expected
- The aliased frequency can be calculated with this equation
- The signal must be sampled at at least twice the highest frequency
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