Comment by hackermom
14 years ago
There is no point with going over 16 bits, but there is definitely a point with going over 44.1khz, as it allows you to actually reproduce waveforms more accurately than 44.1khz. Try reproducing f.e. a sinewave accurately over 4-5khz with a sample rate of just 44.1khz - it cannot be done, and at this point we haven't even taken into account the issue of varying slew-rate characteristics of the thousands or so different DAC output stages in use in personal audio equipment.
44.1khz gives too much aliasing distortion, but 192khz is quite the overkill. Ideally, digital audio could sit on 16 bits of depth sampled at 96khz.
No. This really is not the case. The article _specifically_ addresses this misconception.
The signal reproduced from your 44.1kHz sampled digital input is not a stair-step like some broken waveform editor might display: On output it goes through a matched reconstruction filter (which may, in fact, be digital and involve an oversampled DAC or it could be analog though those are harder to build without compromise). After the reconstruction filter the output is _EXACT_, assuming the input only contained energy below the nyquist (well, and was sufficiently far away from the reconstruction lowpass).
So even a 5khz sine wave is reproduced perfectly with 44.1kHz sampling.
@nullc: of course you're right, and the commenter you're replying to does not understand the Nyquist-Shannon sampling theorem. Which is a shame, because the article specifically addressed this point.
These discussions of audio standards always get sidetracked by people who don't understand or believe this result. (Have to admit, the result is surprising).
I think there may be problems with the argument in TFA, which is based exclusively on standard linear systems theory.
Of course, the ear and some of its perceptual components may be significantly nonlinear, and thus not covered by the frequency response graphs of TFA.
These graphs assume linear systems, in which you put two frequencies in, and the same frequencies pop out in scaled form. Nonlinear systems can produce new frequencies in response, and this possibility is not discussed in TFA. Probably these effects are quite minor, but may be audible to some listeners on some equipment for some choices of source material.
Indeed, but if there were non-linearies in the ear (there are many, of course) which allowed detection of ultrasonics (less likely, because the first stage of the ear is impressively linear) you'd expect them to show up in the actual listening tests.
The TFA does at least make this the-proof-is-in-the-pudding point somewhere in its depths. :)
Couldn't agree more with you! 192kHz is overkill as a "final" format.
16 bits is very limiting for music with lots of dynamics (ie: classical). Very quiet sounds sound quite bad at 16 bits, but since most pop music has about 6-12db of dynamic range, it doesn't make much of a difference.
I always thought the sweet spot would be 96-24. But the truth is, the market wants smaller and portable digital files, not higher quality music. Anything MP3 encoded will sound significantly worse than a CD anyways.
16 bits is not "very limiting" for anything, unless you think your ears themselves are very limiting.
Many things are mastered poorly— recording engineers crushing the dynamics in order to get the loudest possible signals— mostly a problem for pop music, but nothing is immune.
It's been observed that the various 'higher-definition' recordings have less brutal mastering— no doubt owing to the different audience they are marketed to. But this isn't a property of 24-bit vs 16-bit distribution.