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Originally Posted by brendank310
The bottom is not accurate as what you would listen to. The signal is run through a low pass filter in the digital to analog conversion that occurs in digital audio. The entire content of the analog signal is contained in the digital data set, and can be re-constructed by using the appropriate filter. Because nothing is free in this universe(damn conservation of energy) there can be a small amount of loss, but thats not avoided in analog either. The thing that gets people about the whole digital is worse than analog is that the sampling theory says you need to sample at 2 times the frequency of the sampled signal which needs to be "bandlimited" which actually can't happen, but we can get pretty close nowadays. So if human hearing is bandlimited from 0Hz to 22Khz or so, we sample at 44Khz. Some oversampling occurs, to prevent aliasing errors.
If your interested in this stuff, look up Shannon-Nyquist Sampling Theory.
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Here's another interesting theory though, which explains why CD never seems to cut it despite the Nyquist theory looking good on paper:
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44.1kHz sampling can detect high frequencies up to half that sampling rate - 22.05kHz, BUT it CANNOT accurately sample the voltage levels of high frequency transients with only 2, or a few more, samples, regardless of the number of Bits.
Philips, co-inventor, of the technology knew that then, AND published in their Technical Information for their Test Equipment that for ACCURATE voltage measurements the sampling rate needs to be 80 times the highest frequency to be measured - yes EIGHTY, thus 44.1kHz sampling will only measure precisely up to about 550 Hz! Given how the human ear/brain compensates, and that most of the music signal is below about 2kHz, we can get away with about 16 times the sampling rate - I think ... thus 256kHz sampling for 16kHz -{the highest frequency I can still hear, and sufficient for the Harmonics of musical instruments}-, but we have only sampling up to 192kHz available, -{and please some-one correct me if higher is now available}-, thus to 12kHz in the treble for harmonics is reasonably accurate.
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Correction from me – we have DXD recorders now that can work at 32-bit/384kHz rates, it’s called Digital eXtreme Definition. It’s used so that the one source will be accurate for both DSD encodings (SACD) and PCM encodings (Blu-ray – 24/192). 2L Recordings from Norway is putting out SACD/CD/Blu-ray combo discs with all tracks sourced from DXD recordings – they are so far reviewed as phenomenal. However this theory lends some credence to those that say they can hear a difference between 24/96 and 24/192.
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OK for older listeners with treble hearing loss, and for most Commercial Radio intended pop music for people who don't really listen -{the "watch" phenomena some-one posted about earlier in this thread}- and that is fine if they are happy.
IF we further relax the threshold for accuracy to minimum 8 times, for 20kHz, to ensure >x8 times for audible frequencies, then the 178.4 and 192kHz chips suffice, but most of you have only 96kHz sampling in your Hi-Spec format discs - correct ?
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Not quite correct, the “hi-spec” discs allow for 24/192 up to 5.1 now (blu-ray) and 24/96 for 7.1, while most receivers sold over the last 4~5 years have 24/192 capable DAC’s in them.
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To my hearing, comparing musical instruments recorded with good high frequency capable microphones to a good quality high speed analog tape recorder, well aligned, to the same signals recorded to digital at the original Professional sampling of 48kHz, -{and that is still used in a lot of studios}-
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Interjection – it might still be used in a lot of indie studios, but most professional studios are using 24/96 or higher
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for all harmonics above about 5kHz the analog replay is noticably more true-to-life than the digital - compared to the direct sound of the instrument I have just recorded. ALSO, the the transient response is subliminally messed up as result of the pre-echo, filter resonance from the steep slope filters that are necessary with all digital to remove aliasing noise from intermodulating with the treble harmonics of the music. These filters are in all CD players except for a very few new models which have a new type of filter, thus a second reason for the mediocre treble quality from all CDs to date. Now that better filters are available, and the almost high enough sampling rate, is the time that Digital for music recording should have been introduced - not 20+ years ago!
Ideally, audio should have waited till Blu-ray was ready also, and it was known, I think almost 20 years ago that Blu-ray was possible, though I have forgotten exactly when I first read of it.
The MP3/data reduced/reduced dynamic range satisfied listeners' ancestors were quite happy with portable cassette players and low priced, mediocre quality turntable/tonearm/cartridge combinations years ago, and to them the past 20 years of CD has been wonderful, and more than some of them seem to need, given what they are buying now !
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Anyway it’s worth keeping in mind that Nyquist is still a theory and in practice nothing can reconstruct what wasn’t there to begin with, it’s just like the whole DVD upscaling scam. 480i is still 480i.
DSD comes closest to an analog pulse – CD is just pretty messy.
http://www.merging.com/2002/images/dsdresponseneon.gif
The entire analog content is NOT maintened in the digital set at 16/44.1 - simple listening tests will show that much to be true.