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> Even though higher frequencies are indeed possible with a 192K > sampling rate, it also means > that each waveform has more samples representing it, thus higher >fidelity. > I never did understand > how the Nyquist theorum could claim fidelity in the upper end by only > having two samples per > waveform. Seems to me like you wouldn't be able to distinguish a >sawtooth > from a sine from a > square wave if you only sampled at twice the highest frequency. Stephen - It is my understanding that the Nyquist theorum actually specifies the theoretical maximum frequency that can be represented with a digital signal. It says nothing of the quality of that representation. There's a key issue that makes the nyquist theorum all important to the A/D conversion process - frequencies higher than the Nyquist freq. will manifest themselves in the resultant digital signal as much lower frequency noise. That noise is basically garbage and is nearly unpredictable. Furthermore, frequencies that are close to but still less than the Nyquist frequency will be represented, however the amplitude of the representation will vary over time based on the _difference_ in frequencies. How much the output varies in amplitude is a more complex equasion - important thing is that frequencies close to the Nyquist freq. (even though they are still less than) will be VERY distorted but will still be represented "perfectly in-tune". Example: 48KHz digital A/D converter could capture a 23KHz pure sine wave, and output a 23KHz sine wave whose amplitude varies at 1Khz - you would hear that 1Khz signal for sure! So this is a long story to explain the real meat of the issue - LOW PASS FILTERING. Any digital audio capture system must have low pass filtering before the A/D converter. Ideally you would like to make sure that absolutely no signals whose frequency is greater than the Nyquist frequency can be input to the A/D converter - because those will just cause useless garbage noise out the other end. You also want to roll off frequencies that are close to the Nyquist frequency, because though could create ugly garbage noise too even though they will be represented somewhat. These filters are called anti-aliasing filters and every digital audio system has them. The thing about filters is that even the most expensive filters don't cut off sharply at a certain frequency - instead they "roll off" slowly starting at 0Hz and gradually roll off more and more as the frequency goes up. People refer to the "cut-off" frequency of a filter, but be aware that this is just the frequency at which the sound is sufficiently attenuated so that its much quiter than the original signal. This means that a simple filter with cutoff frequency of 20KHz still passes 22KHz signals they're just real quiet. It also means that this filter is rolling off your 18KHz signals too. The more money a manufacturer spend on the filter design and implimentation, the "steeper" it is, but none can be infinately steep. So you've got to set your anti-aliasing filter well below the nyquist frequency to be sure that nothing above the nyquist freq. of significant volume can get through to the A/D converter. If you don't have a lot of money to spend on anti-aliasing filters (and most music equipment falls into this catagory) you use an off-the-shelf filter solution which doesn't have very steep filtering (and therefore has a cutoff freq. well below the nyquist freq.). This cheap filter will roll off frequencies well below the cutoff freq. too. This could be what most people are hearing when they say 44.1KHz digital doesn't sound good. Ok, tired yet??? Well how about this one here - Time to learn about anti-imaging filters. This is just about the same thing as an anti-aliasing filter, but its on the other side of things - the outputs. Every device that uses digital audio has to have one. Typically it will be a similar or identical low pass filter to the anti-aliasing filter on the input. This further "rolls-off" some of the high end of your signal (even stuff below the Nyquist freq. - same as above). As a side note - the anti-imaging filter is the reason why you can buy a really high end CD player and it can actually sound much better than a crappy one even though both are using the exact same digital content. (There are other reasons like upsampling though too) Same with sound cards, mixing desks, ect. That's why a lot of people use SPDIF and do the final conversion only once on a high quality audio system. All of these things lead me to believe that even the most high quality digital audio system can only do a very good job of representing frequencies less than about 1/3 of the sampling frequency. This means that I consider 44.1KHz digital audio systems can represent up to 14.7KHz very well. That's probably fine for most of the public, but there are many people who can hear this. I believe that 48KHz is a pretty significant improvement with the ability to represent frequencies up to 16KHz quite well. This is probably right around the actual threshold of most human beings and above this starts to become somewhat esoteric and specialized. (keep in mind though that DVD supports 192KHz - so its not THAT esoteric -smile-). Jon If you got this far you don't need this link, but here is a decent explaination of some of this with diagrams: http://www.cems.uwe.ac.uk/~lrlang/multimedia/audio2.pdf upsampling on high end cd players: http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf