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![]() Quote:
Frequencies above 20kHz are always filtered out somewhere between the musical instrument (or any source of sound) and the human brain (with the exception of directly applying the signal to a fluid the individual is submerged in or to a metal rod attached to the skull). Once filtered out, all harmonics generated by a particular frequency is removed from the signal. For analog the last catchall for this filtering if not done in hardware somewhere is the Basilar Membrane in the human ear. This piece of anatomy is basically a 20Hz to 20kHz bandpass filter. In the digital world anti-aliasing filters are use to remove any content at or above the Nyquist frequency (22.05kHz for audio CD) before sampling. Data is then sampled for analog to digital conversion at a rate twice the Nyquist frequency. Contrary to popular belief increasing the sample rate has no benefit as the sampled analog signal can be accurately recreated when sample at just twice the Nyquist frequency. Higher sample rates makes for good marketing though. There is one situation in the digital world where content above 20kHz can effect lower frequencies in the final recreated signal. Reconstruction filters assume that anti-alias filters have been properly utilized. If anti-aliasing is not properly performed then any frequency content between the Nyquist Frequency and the sample rate will be mirrored into the final signal causing distortion. Nika’s book does assume some technical savvy but not to the level of needing an EE degree. What is doesn’t do is use a lot of audio professional jargon. |
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