RE: Ask an Audio geek
April 12, 2016 at 11:59 pm
(This post was last modified: April 13, 2016 at 12:36 am by Alex K.)
This is an example I found of what frequency limitation does to a sharp pulse:
![[Image: af-f16.gif]](https://images.weserv.nl/?url=sound.westhost.com%2Farticles%2Faf-f16.gif)
Not a perfect illustration but the best I found. Notice that the resulting filter response (red) has these typical broad movements which will be picked up if you sample at twice its critical frequency, whike the hard transients are gone. This is not an artefact of bad electronics, that's what it means to limit the frequency of the signal. A turntable cartridge stylus would do the same thing to short pulses due to it mechanical inertia. Our ears do the same thing. If you don't hear or transmit above 20k, you cannot physically distinguish between that pulse and the response curve - they become the same (up to phase shifts near the cutoff frequency which depend on the type and steepness of filter used, but these effects can be reduced arbitrarily by choosing a higher sample rate if they bother you. But they are also present in any analog gear and your ears. Still, one might be motivated to sample at 96k in order to be removed two octaves from the hearing range. That way, one can use a given filter and more perfectly avoid aliasing effects and such, and other weird cutoff effects like phase shifts will be even further removed from the hearing range)
Not a perfect illustration but the best I found. Notice that the resulting filter response (red) has these typical broad movements which will be picked up if you sample at twice its critical frequency, whike the hard transients are gone. This is not an artefact of bad electronics, that's what it means to limit the frequency of the signal. A turntable cartridge stylus would do the same thing to short pulses due to it mechanical inertia. Our ears do the same thing. If you don't hear or transmit above 20k, you cannot physically distinguish between that pulse and the response curve - they become the same (up to phase shifts near the cutoff frequency which depend on the type and steepness of filter used, but these effects can be reduced arbitrarily by choosing a higher sample rate if they bother you. But they are also present in any analog gear and your ears. Still, one might be motivated to sample at 96k in order to be removed two octaves from the hearing range. That way, one can use a given filter and more perfectly avoid aliasing effects and such, and other weird cutoff effects like phase shifts will be even further removed from the hearing range)
(April 12, 2016 at 11:35 pm)Alex K Wrote: No no, it has nothing to do with the psychoacoustic effect (that's relevant for dropping in-between frequencies in mp3 compression), it has to do with the limited frequency range of hearing.
The point of the sampling theorem is that with a signal of limited highest frequency, these discrete points uniquely determine what goes on between them. If the signal you send in is cut off sharply at say 20 khz using a steep filter, it will not have any features which are so quick that they can be missed if you only measure 44100 times per second. Such a signal can only change its direction of up-down movement so fast, because changing faster would require containing higher frequencies. If you have ever seen what a low pass filter does to short pulses, it might become clearer.
The fool hath said in his heart, There is a God. They are corrupt, they have done abominable works, there is none that doeth good.
Psalm 14, KJV revised edition
