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- Jul 21, 2020
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On time domain dac filter tests here and pretty much everywhere else the response is always shown with an squarewave that has more bandwidth than possibly could be in a real recorded digital signal. During adc or while downsampling to 44.1khz higher frequencies have to be filtered out to prevent aliasing. If this is done properly, the recorded signal does not include any frequencies higher than 22kHz anymore. So a squarewave impulse response seems senseless.
We need a low pass filter during dac to attenuate imaging and want the filter to be mostly accurate to the recorded signal in the time domain too. Mathematically unavoidable sharp filters do more harm in time response and less in frequency response and slow filters the other way around.
But why not use the steepest 'square' wave a signal can have at a given sampling rate and bandwidth and compare that to the filter response? Wouldn't that be much more a real world approach to see if the filter really does harm to the time response of actual correctly recorded music than use a square wave that couldn't ever possibly be in a correctly recorded signal at that sampling rate?
We need a low pass filter during dac to attenuate imaging and want the filter to be mostly accurate to the recorded signal in the time domain too. Mathematically unavoidable sharp filters do more harm in time response and less in frequency response and slow filters the other way around.
But why not use the steepest 'square' wave a signal can have at a given sampling rate and bandwidth and compare that to the filter response? Wouldn't that be much more a real world approach to see if the filter really does harm to the time response of actual correctly recorded music than use a square wave that couldn't ever possibly be in a correctly recorded signal at that sampling rate?