bhaagensen;282186 Wrote: > > it's Easter and I seem to have time to wonder about all kinds of > strange things, among which is subj. > Happy Easter, Bjørn
bhaagensen;282186 Wrote: > > I think that according to the Nyquist theorem sampling frequencies > above 22.05 at 44.1 results in aliasing, therefore everything above > 22.05 is filtered in the analoge domain before sampling to avoid > aliasing ? But why/how can one then see from the reconstructed waveform > that clipping is present (due to e.g. loud mastering)? Modulo being a > bit flat, it shouldn't clip as far as I understand? > Actually "filtered" is not the correct term, folded would be better. If the sample rate is 44.1 an input signal or transient with a frequency of 33 would "fold" to a copy of the signal around 33-22.05=10.95 and would be perceived as distortion. To prevent distortion, before sampling the signal is low-pass filtered. Because unsampling (in your CD player or SqueezeBox) again creates high frequency components also here is low-pass filtered. Now clipping is something entirely different. Is Nyquist related to the frequency domain, clipping is related to the amplitude domain. If I have a 16 bit sample there are 65536 discrete "levels" lets say from 0db to 96db. If I input a signal of 100db the range would be to small but the sample would be 65536 was we ran out of numbers; the signal gets quite literary "clipped". It is than not possible to restore the signal back at unsampling as the unsampler does not "know" that the signal was 100db bhaagensen;282186 Wrote: > > Next question. Is compression not preferable to overly aggressive > filtering as in above? Sure one gets a distorted waveform compared to > the original source, on the other hand one can potentially avoid > throwing too much stuff in the bin. If so what is an acceptable amount > of compression? Or when is it unacceptable? > I assume you talk about audio compression or limiting (like Dolby A, B and C or Dolby HXpro) All these systems work in the analgue domain and where a means to deal with the rather poor S/N ratio of Tape of 50dB or so. It would "squeeze" a signal with a range of 70dB to the available 50dB. Compression in the digital domain means simple removing bits from the signal that would not be audible anyway like a soft high pitch sound over a loud lower pitch sound. Of course it is debateble what is audible or not. Hence the, rather heated, debate on compressed or uncompressed. One still has to consider that you can only control the last bit of the long trajectory the sound has made. What happened in the studio or at mastering is out of you hands. So use your ears, if it sounds fine for you it probably is ok :-) -- th00ht SqueezeBox v3, SqueezeCenter (7.0.1 - 17793) Quad 303 + Two Quad Electrostats ------------------------------------------------------------------------ th00ht's Profile: http://forums.slimdevices.com/member.php?userid=15656 View this thread: http://forums.slimdevices.com/showthread.php?t=45107
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