Music, Programming, and other topics related to the modern Music Engineering Scene.


As mentioned at the end of the last post, aliasing can be a problem when recording digitally. Aliasing occurs when you try to record frequencies that are too fast. You have a limit on which frequencies you can record based on your sample rate. The Nyquist Theorem says that you can only accurately record frequencies that are less than half of your sample rate. For a 44,100 Hz sample rate, you can record frequencies up to 22,050 Hz. This covers everything that human ears can hear.

When you try to record something greater than 22,050 Hz, you will get aliasing. This creates high pitched "chirpies" that don't actually exist, but get reproduced because of how samples are taken. Consider the following picture:
The black sine wave is the original signal. The blue dots are the samples that get taken, and the green dashed line is the false frequency that gets reproduced. You can see it is a much lower frequency than the original signal, and is not at all what we wanted to capture.

To get around aliasing, we apply anti-aliasing filters. Basically, this is a low-pass filter with a cutoff frequency just below your Nyquist frequency (half of your sampling rate). This prevents any frequencies that are too high from ever trying to get recorded, protecting the signal that we can capture from unwanted aliasing.


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