In article
,
adamdea wrote:
On Sep 22, 5:38 pm, Jim Lesurf wrote:


FWIW I use SOX for various purposes, and find it works well. However
I've not tried it as a base for different filter functions.

I notice that the manual contained the tantalizing words

"All resamplers use filters that can sometimes create 'echo' (a.k.a.
'ringing') artefacts with transient signals such as those that occur
with 'finger snaps' or other highly percussive sounds. Such artefacts are
much more noticeable to the human ear if they occur before the transient
('pre-echo') than if they occur after it ('post-echo'). Note that
frequency of any such artefacts is related to the smaller of the
original and new sampling rates but that if this is at least 44.1kHz,
then the artefacts will lie outside the range of human hearing."

From my inept questioning and faltering understanding of your answers it
appears that this is probably true (ie the artefacts probably can't be
heard)

The problem with the quote from sox is that it examples a common behaviour
in audio. It presents a mix of fact, opinion, and assumed 'fact' without
clear distinctions as as sweeping generalisations. This makes it very hard
for the unwary reader to tease out reliable portions from guesswork or
error. Or assertions that may be correct in some cases, but not others.

Slainte,

Jim

--
Please use the address on the audiomisc page if you wish to email me.
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

"adamdea" wrote in message
...
On Sep 22, 5:38 pm, Jim Lesurf wrote:
In article
28850714.1859.1316703150761.JavaMail.geo-discussion-forums@yqaa12,
adamdea wrote:

In fact Sox is currently used for sample rate conversion of everything
up to 24/96 prior to application of DRC. I think this measn i have
filters (minimum phase i think), at lots of points in the audible range.

FWIW I use SOX for various purposes, and find it works well. However I've
not tried it as a base for different filter functions.

I notice that the manual contained the tantalizing words

"All resamplers use ?lters that can sometimes create 'echo' (a.k.a.
'ringing') artefacts with transient signals such as those that occur
with '?nger snaps' or other highly percussive sounds.

Finger snaps and other percussive sounds have spectral contents that are so
irrelevant to the unit impulses that I think that we are talking about that
I dispute the relevance of that paragraph.

Just because people write good software and interesting manuals doesn't mean
they know squat about real world signal analysis or psychoacoustics.

Such artefacts are much more noticeable to the human ear if they occur
before the transient ('pre-echo') than if they occur after it ('post-echo').

That is true for musical sounds that have peak energy in the 2 to 7 kHz
range. Above 20 KHz, not so much (to say the least!).


Note that frequency of any
such artefacts is related to the
smaller of the original and new sampling rates but that if this is at
least 44.1kHz, then the artefacts
will lie outside the range of human hearing."

Now that is a sentence to live by! (especially that last phrase).

From my inept questioning and faltering understanding of your answers
it appears that this is probably true (ie the artefacts probably can't
be heard) although not because the artefacts are simply composed of
frequencies we can't hear, but because they represent the time domain
effect of changes in frequencies we can't hear (?).

Right, both issues being the frequency that the artifacts happen at and also
the nature of the artifacts, are relevant to the fact that we generally
can't hear them.

Or maybe something like a beat frequency where the beating is too fast to
hear(?)

Beating presumes relatively large amounts of nonlinearity by the standards
of good modern electronics.

Jim, Arny

Thanks very much for both your helpful advice and contributions.

I think I am slowly getting somewhere, although every time I think I
have grasped the essence of the matter, I find that most of what I
think I have grasped slips through my fingers. It strikes me that the
issues involved here are rather deep because we are not entirely sure
what we are measuring, and are therefore not clear how accurately we
are measuring it.

I take John's point on board entirely that this is not a problem with
digital systems as such- it seems to be a problem of what happen when
you band limit a signal. That problem arises with any information
channel but in the ca of digital audio it is specfied by theory and
implemented intentionally (even if reluctantly) . I can appreciate
that this band limitation at the ADC input (or more realistically
prior to downsampling) may possibly be irrelevant because of the real
bottleneck is in the microphone, the speakers or the ear.

That seems to get us into the territory of what bandwidth is good
enough for the ear. Here I can see as number of different questions
which may of may not be distinct

a) what is the highest frequency humans can hear

b) what is the lowest low pass transition frequency at which if a
sensible filter is put in human beings will not notice any difference
between the filtered output and the pre-filtered signal

c) at what sort of filter frequency will I hear something called pre-
ringing before some sort of transient sound- ie where that sound will
after filtering sound noticeably different from the pre-filtered
sound, or possibly subject to distinct audible pre-echos eg before the
finger click I would hear a distinct albeit somewhat quiet finger
click as a distinct event (this effect being indistinguishable from
someone clicking his fingers quietly and then louder?). This begs i
suppose the question of how far apart sounjds need to be to be heard
as distinct.

d) at what Low pass filter transition frequency might 2 sounds which
would pre-filtering sound different become just one sound. .

So now, that takes me to the question of what real world sound should
best be used to test these limits.

I noticed some material discussing the frequencies generated by
musical instruments. This seemed to suggest that some instruments
generate energy up to 50kHz but mainly at quite low levels the
exception seems to be the cymbals which seem to have a lot of energy
at 100kHz plus. It therefore seems that a cymbal stroke would be the
thing to look at as the most extreme case of a real instrument with
high frequency content .

It occurs to me that perhaps this is not the instrument with fastest
attack however. any idea what that would be and where one could see
what leading edge sound would be in the time domain.

Presumably what we need is to identify a real world transient so that
we can

a) analyse what the effect of applying a filter (especially the
ubiquitous half band linear phase 20-24kHz transition -100db at
stopband, -6db at nyquist) on that transient.

b) listen to hear if we can tell any difference between versions
fimtered at (eg) 16, 18, 20, 40 or 80 kHz,
[I am assuming that this could be captured by a microphone in the
first place. I think we may also have to assume that a sinc like AA
filter would be used.