Gibbs Phenomenon_pinerton or others?

Archived from groups: rec.audio.high-end (More info?)

The Gibbs phenomenon is simply what happens when higher frequency
components are removed from a Fourier reconstruction of a square wave.
The resulting waveform is bandwidth limited and is analogous to
brick-wall filtering. Providing that the removed components are beyond
the frequency response of humans, this would not appear to be a
signicant source of sound degradation nor would it be audible.


randy wrote:
> I will start out by saying this is way over my head-but I have been
> told this is a signifcant problem that digital has that analog
> doesn't-and is audible-
>
> http://mwt.e-technik.uni-ulm.de/wo [...] node4.php3
>
> Comment.

Archived from groups: rec.audio.high-end (More info?)

randy wrote:
> I will start out by saying this is way over my head-but I have been
> told this is a signifcant problem that digital has that analog
> doesn't-and is audible-
>
> http://mwt.e-technik.uni-ulm.de/wo [...] node4.php3
>
> Comment.

In audio there is *no* squarewave to be digitized, so this problem doesn't
occurr. The maximal (theoretical) slew rate would be happening with a 20kHz
signal at full output power and would already blow your tweeters, so do not
try to create this signal. In a music feed the voltage will be down at least
12dB, which corresponds to max. power/16.
Whoever told you this is technically as uninformed as you are. But this
seems to be one of the prerequisites of high-end "gurus".

--
ciao Ban
Bordighera, Italy

Archived from groups: rec.audio.high-end (More info?)

On 27 Sep 2005 02:11:59 GMT, "randy" <rbessinger@deloitte.com> wrote:

>I will start out by saying this is way over my head-but I have been
>told this is a signifcant problem that digital has that analog
>doesn't-and is audible-
>
>http://mwt.e-technik.uni-ulm.de/world/lehre/basic_mathematics/fourier/node4.php3
>
>Comment.

You have been told this by people with an agenda, who are trying to
baffle you with science. The Gibbs phenomenon is simply what happens
to a discontinuous function, such as a square wave, when you remove
the higher frequency components, and this applies just as much to
analogue as to digital. In any properly implemented digital sampling
system, the *input* signal is band-limited to less than half the
sampling rate, so that no aliasing takes place, and the output signal
is a virtually perfect representation of that band-limited input
signal. Analogue systems also have bandwidth limits, and a 5kHz
squarewave fed through the ubiquitous 24/96 digital system will look
much 'squarer' than the same signal fed through a standard 15ips
studio analogue recorder.

Note that a 20kHz square wave will come out of *both* machines as a
sine wave - but it will be a *clean* sine wave out of the digital
system.
--

Stewart Pinkerton | Music is Art - Audio is Engineering