Why are more bits/ higher sampling rate better?

[preiter]

I'm wondering if anyone has a good explanation as to why we need more than 16 bits and 44.1 KHz sampling.

16-bits = 65,536 discreet levels = 96dB worth of dynamic range.

Most albums don't use nearly that much dynamic range. Why do we need more dynamic range if we aren't using all that we have now?

According to the Nyquist theorem, 44.1 KHz sampling rate should be sufficient to capture > 20KHz of frequency range. I know my 40 year-old ears can't reach up to 20 KHz, I doubt most people's ears can.

So why do we need to sample at a higher frequency?

I'm not arguing that higher bit rates aren't better, I'm sure there wouldn't be 24 bit / 96 KHz formats out there if somebody couldn't tell the difference. I'm just curious why. Where does the theory fall short?

[TheShaman]

Perhaps this audioholics article will help a bit (pun intended ).

[el`Ol]

The steep anti-aliasing filters in PCM coding create artefacts that reach far below the Nyquist frequency. This is the reason why higher sample rate gives an audible improvement.
The SACD has no filter ringing problem, but high sample rate is necessary to get a reasonable signal to noise ratio. The high bandwidth is a side effect.

[oshifis]

16 bits is not 96 dB dynamic range. This were true if the music at -96 dB level could be represented on 1 bit (the LSB), but then this is no music just jumping between two levels. Similarly, a signal of -60 dB is represented on 6 bits only. 6 bits has 64 discrete steps, imagine what happens to the music with such poor resolution. Also most recordings never reach near 0 dB level, and I assume the average could be around -30 to -40 dB.
The steep analog filter after the DAC has very large phase shift much below the cutoff frequency, that may also be audible.

[BlackCatSound]

Nyquist only really applies to frequency reproduction.

As you get closer to the nyquist freq you lose amplitude and phase information. Really from about nyquist/4 you're starting to lose this information.

As for up and oversampling, as the source is 16bit 44.1k you've lost the information already and it can't be recovered.

[rfbrw]

Originally posted by oshifis
Also most recordings never reach near 0 dB level,

If you mean 0dBFS, they can and frequently do.

[oshifis]

I mean by 0 dB the analog level that is resulted by all 0s or all 1s in the digital word.

[rfbrw]

The twos complement number format is the norm in audio so all zeros would correspond to bipolar zero and all ones to -1 decimal. The peak, after which one goes into overload, is 0dBFS and recordings often hit that peak

[error401]

Originally posted by BlackCatSound
Nyquist only really applies to frequency reproduction.

As you get closer to the nyquist freq you lose amplitude and phase information. Really from about nyquist/4 you're starting to lose this information.

As for up and oversampling, as the source is 16bit 44.1k you've lost the information already and it can't be recovered.

No. The definition of sampling theorem is basically that for any frequency under fs/2 you have all of the information necessary to exactly reproduce the signal (emphasis mine):

Exact reconstruction of a continuous-time baseband signal from its samples is possible if the signal is bandlimited and the sampling frequency is greater than twice the signal bandwidth.

Practical considerations (and quantization) aside, the frequency and phase response is perfect until you get to fs/2.

[BlackCatSound]

This is something we've discussed at great length in the office with respect to digital scopes.

Not audio I know:
http://www.clarkvision.com/imagedetail/sampling1.html

http://www.themusicpage.org/articles...ingTheory.html