[LAU] Rolling off high frequencies when mastering?

Monty Montgomery xiphmont at gmail.com
Wed Apr 21 19:12:20 UTC 2010


> Ogg and mp3 roll off before the nyquist-frequency because of their algorithm.
> So I wouldn't take that as a reference.

Speaking for Vorbis, it depends on the quality mode or bitrate you
select.  Vorbis is also fully aware ot the ATH and absolute dynamic
depth of the human ear.  If its psychoacoustics say a tone is
inaudble, it won't code it.

> The highest frequency possible to reproduce -with correct amplitude- is half
> the sampling-rate _only_ if the phase is aligned to the sampling-clock so that
> minima/maxima of the sinus are correctly sampled. If its out of phase, the
> amplitude is not reproduced correctly.

Energy is preserved, even at the exact nyquist frequency, however
modulating the input signal against the sampling clock results in the
signal being partly shifted to DC.  The signal energy is there... it
appears somewhere else.  This is the case only with a signal exactly
at Nyquist, unless you also have a system exhibiting IMD (most modern
ADCs and DACs are very very clean.  I can't measure the IMD even on my
ten year old samplers, though they're much noisier than today's).

The aliasing caused by signals exceeding the nyquist and folding back
over into the sampled spectrum as well as aliasing caused by IMD at or
near the sampling frequency are both mitigated/practically eliminated
by lowpassing somewhere south of Nyquist.

> It is easy to understand that this correlation between phase and correct
> amplitude also affects frequencies below half the sampling-rate. Might be as
> low as quarter of the sampling-rate, which in case of the CD is 11kHz. Below
> that you will have more then four samples to reproduce the sinus wave.
> That is in fact another reason to do the recording, mixing and mastering in
> more then 44kHz...

This is utter hogwash.  All the signal energy and resolution is
preserved to exactly the limits of the sampling rate at all lower
frequencies, even if your 'biggest dots' aren't landing at 0dB or
wherever.  True of audio signals, true of  images, true of video.  The
same discrete sampling lessons apply to all three equally.

Monty


More information about the Linux-audio-user mailing list