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Industrial One · Guest

On Jul 5, 9:18 pm, "Earl Kiosterud" <some...@nowhere.com> wrote:

[quote]Sampling theory tells us that it takes at least two samples per cycle, hence the 44.1 KHz
sample rate. The highest frequency that can be captured is 22.05 KHz (Nyquist frequency);
frequencies higher than that will create alias frequencies below 22.05 For example, an
audio frequency at 30 KHz would produce an alias frequency component at 14.1 KHz (44.1 -
30). It also produces one at 44.1 + 30, but who cares? The 20KHz audio upper limit allows
for comfortable guard band to the Nyquist frequency.
[/quote]
Do I understand correct: hz is one sine loop per second, I generate a
sine sweep from 0 - 20 KHz with a specified duration, when I view with
an audio application and zoom 'till individual samples are visible, I
notice that as frequency increases, the sine waves become shorter, and
gradually begin to appear more triangular as the smaller sample
interval makes a perfect, smooth sine shape impossible. Finally, when
it reaches 20 KHz (20,000 sampling rate) the waves have reached their
limit on appearing anything that resembles a sine, and is now a
perfect triangle: one sample at the bottom, one at the top, and one at
the bottom again, like /\/\/\/\/\/\/\/\/\/\. This would technically be
the maximum, but instead, as I continue scrolling, I see the waveform
look something like a private-case of sine waves. This time, a sine
block composed of triangles. What you>re saying is that beyond 22.05
is a hack that simulates higher frequencies, but don>t technically
exist on a digital medium, like the waveform of the sine sweep I
created?

[quote]DVD audio is just for marketing. No one, with the possible exception of a few young people
who can hear above 20 KHz, and many dogs, can hear the difference between regular 44.1K
[/quote]
Pffft... I>m 18 and I can hear 17 KHz maximum, assuming 20 KHz for
anyone is an exaggeration, and whoever claims to tell the difference
between 44/96 is some autistic motherfucker that probably pisses
himself in class 'cuz he keeps hearing the "whistling" from the rat
repellant. Also, I>ve subtracted a 44 track by a downsampled 32
version to hear what EXACTLY is stripped and all I heard was extremely
faint clicks, jingles and dings (from the beat of the treble
percussion instruments) and had to amplify the waveform up 30 dB to
hear it clearly (real headache inducer, yo.) So... what I>m really
missing from a 96 KHz track are some faint jingles TWICE as high-
pitched and inaudible from the upper freqs of a 44 KHz track?
WOW........ is THIS really what those assramming audiophiles bitch
about?

[quote]16-bit audio and 96 or 192K sampling and 24 bits -- it>s been proven, though some will tell
you they can. It>s something they call "resolution" for which they have an altar, dogma and
lots of ritual. They get this dreamy look in their eyes. Challenge it, and their veins pop
out and they go on rampages. It>s likely that much of the stuff you get on DVD-audio discs
[/quote]
Proof drugs are awesome.

[quote]is better stuff, and has been more meticulously recorded, hence the good sound of many of
them. It ain>t the extravagant bit depth and sampling rate. There are some damned
good-sounding CDs too. Even if you had a regular CD version and a DVD-audio version, and
the DVD-audio version sounded better, would you actually believe that the improvement was
because of the bit depth and sample rate? Couldn>t be anything else, could it? How are
they going to sell DVD-audio discs if they let the CDs sound the same?
[/quote]
QFT.

[quote]Hope this helps.
--
Earl
[/quote]
Was informative, thanks.

On Jul 5, 10:31 pm, "Richard Crowley" <rcrow...@xp7rt.net> wrote:
[quote]The "Red Book" convention for making audio CDs was developed
back in the early 1980s and established 44.1KHz as the sampling
rate. In order to maintain forwards and backwards compatibility,
all CDs must use that sample rate.

Any periodic occurance can be referred to as "freqency". Whether
it is something that happens every femtosecond (like light) or every
1000 years (like the century).

The Nyquist-Shannon sampling therom tells us that you must
sample at *twice* the desired highest frequency to adequately
reproduce the original waveform. It is said that 22KHz was
selected as the top end (x2 = 44KHz) because of the state of
the art in filters back in those days.
[/quote]
I see.

[quote]Obviously, one can notice the difference if the song was
downsampled to 22,

You notice it because reducing the sampling rate to 22KHz
actually reduces the top end to 11KHz which many people
can detect.
[/quote]
I don>t follow. Why 11?

On Jul 6, 2:06 am, "Chronic Philharmonic" <karl.uppi...@verizon.net>
wrote:
[quote]It was because they used video recorders for mastering prototype and first
generation CDs, and it was the nearest available frequency that was greater
than 40KHz needed to meet the sampling Nyquist requirement of at least two
samples for the highest frequency to be recorded (20KHz). See also:http://www.cs.columbia.edu/~hgs/audio/44.1.html
[/quote]
Damn, I didn>t know that.

[quote]Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
point?

Marketing. There is no defensible mathematical requirement for it.
[/quote]
's what I thought. I wouldn>t be surprised if them retard scene
rippers start releasing audio in 192 KHz with a bitrate of probably
1024+ and advertise "DVD QUALITY AUDIO!"

I>ll read the other posts later.

Willem · Guest

Industrial One wrote:
) Do I understand correct: hz is one sine loop per second,

Hz is one <whatever> per second.
Your heart beats at roughly 80 Hz.
The moon revolves around the earth at roughly 0.38 uHz (Microhertz)

) I generate a
) sine sweep from 0 - 20 KHz with a specified duration, when I view with
) an audio application and zoom 'till individual samples are visible, I
) notice that as frequency increases, the sine waves become shorter, and
) gradually begin to appear more triangular as the smaller sample
) interval makes a perfect, smooth sine shape impossible. Finally, when
) it reaches 20 KHz (20,000 sampling rate) the waves have reached their
) limit on appearing anything that resembles a sine, and is now a
) perfect triangle: one sample at the bottom, one at the top, and one at
) the bottom again, like /\/\/\/\/\/\/\/\/\/\. This would technically be
) the maximum, but instead, as I continue scrolling, I see the waveform
) look something like a private-case of sine waves. This time, a sine
) block composed of triangles. What you>re saying is that beyond 22.05
) is a hack that simulates higher frequencies, but don>t technically
) exist on a digital medium, like the waveform of the sine sweep I
) created?

Play your sample at half speed and listen to what happens when
you reach 11.025 Khz.

)> > Obviously, one can notice the difference if the song was
)> > downsampled to 22,
)>
)> You notice it because reducing the sampling rate to 22KHz
)> actually reduces the top end to 11KHz which many people
)> can detect.
)
) I don>t follow. Why 11?

Because 11 is half of 22. See above.

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
made in the above text. For all I know I might be
drugged or something..
No I>m not paranoid. You all think I>m paranoid, don>t you !
#EOT

Peter Schepers · Guest

In article <slrng74cnd.41a.willem@snail.stack.nl>,
Willem <willem@stack.nl> wrote:
[quote]Industrial One wrote:
) Do I understand correct: hz is one sine loop per second,

Hz is one <whatever> per second.
Your heart beats at roughly 80 Hz.
[/quote]
I think you mean the heart beats 1.333 Hz.

PS.

Guest · Posted: Mon Jul 07, 2008 3:52 pm Post subject: Re: Frequency/Sample rate

On Jul 7, 11:02 am, Industrial One <industrial_...@hotmail.com> wrote:
[quote]Do I understand correct: hz is one sine loop per second,
[/quote]
No, "Hz" simply is the abbreviation for "Hertz" which
is the SI unit for reciprocal second. It has nothing to do
per se with sine waves, square waves, are waves of
any kind,. Rather it is simply the rate at which whatever
semi-periodic phenomenon occurs. It could be the
rotational period of a neutron star, like 792 Hz.

[quote]I generate a
sine sweep from 0 - 20 KHz with a specified duration, when I view with
an audio application and zoom 'till individual samples are visible, I
notice that as frequency increases, the sine waves become shorter, and
gradually begin to appear more triangular as the smaller sample
interval makes a perfect, smooth sine shape impossible. Finally, when
it reaches 20 KHz (20,000 sampling rate) the waves have reached their
limit on appearing anything that resembles a sine, and is now a
perfect triangle: one sample at the bottom, one at the top, and one at
the bottom again, like /\/\/\/\/\/\/\/\/\/\. This would technically be
the maximum, but instead, as I continue scrolling, I see the waveform
look something like a private-case of sine waves. This time, a sine
block composed of triangles. What you>re saying is that beyond 22.05
is a hack that simulates higher frequencies, but don>t technically
exist on a digital medium, like the waveform of the sine sweep I
created?
[/quote]
Uhm, I think, if I am able to parse what you are saying, yes,
though your grammar is a bit tortured.

But a couple of points:

What you "see" in an audio application often does NOT,
depending upon the application, reflect what the resulting
weaveform actually looks like. The EASIEST graphic display
of a series of sample valies is to simply connect the dots,
but that is NOT what D/A converters do. If you had a
series of alternating +- and -1 values, near 1/2 the sample
rate, and if simply "drew" them by connecting the dots,
what you would see could be what looked like a triangle
wave carrier modulated by a sine wave. FOr example,
at 22 kHz, you>d see the modulating wave as having a
frequency of 50 Hz.

But that waveform, AS DRAWN, has components that
extend to infinity. JUstv as there is a component at
22.05 kHz - 50 Hz = 22 khz, there is also one at 22.05+50 Hz
or 23 kHz. If you had only three such components, then
the waveform would be a 22 kHz SINE wave modulated by
50 Hz sine. But the triangle carrier has compoenents
way out as well, so you>ll see images at 44.1 +- 50 Hz, and
88.2 kHz +- 50 Hz and so on.

ALL these are IMAGES of the original 22 kHz sine wave and,
to be reconstructed properly, requires the additon of the
necessary anti-IMAGING filter (also known as the reconstruction
filter). Now, get rid of all the components at an above 22.05
kHz, and all you>re left with is your original 22 kHz sine, just
as it was recorded.

Now, where OVERsampling techniques come in is HOW
you implement this filter. YOu can, as some assumed incorrectly,
try to implement it in an analog filter, and that get>s real tough.

You can also try to implement it as a digital filter at 44.1 kHz,
but it>s effectively impossible to implement a 22.05 kHz low
pass filter at a sample rate of 44.1 kHz.

So what we do is we OVERsample the 44.1 kHz stream that
we already have: inbetween every real sample, lets insert, oh
63 samples whose value is, let>s say, 0. Or anything slse, for
that matter.

What we>ve done is that we>ve taken the original images
spaced 22 kHz apart and now spaced them instead
22 kHz * 64 or 1411 Khz apart.

Now you can build a VERY GOOD digital filter with very
good out-of-band rejection and very good in-band response,
and then all you have to do in your final analog stage is
make sure 20 kHz is getting through fine, but 1411 Khz is
properly attenuated.

That>s a LOT easier than building a filter which lets
20 Khz through but attentuates 22.05 kHz completely.

Willem · Guest

Peter Schepers wrote:
) In article <slrng74cnd.41a.willem@snail.stack.nl>,
) Willem <willem@stack.nl> wrote:
)>Industrial One wrote:
)>) Do I understand correct: hz is one sine loop per second,
)>
)>Hz is one <whatever> per second.
)>Your heart beats at roughly 80 Hz.
)
) I think you mean the heart beats 1.333 Hz.

D>oh, you>re right. Roughly, that is. :-)
Did you check the other example (frequency of the moon) ? I made
a mistake there too, although it is not by an order of magnitude...

Must be a bad day.

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
made in the above text. For all I know I might be
drugged or something..
No I>m not paranoid. You all think I>m paranoid, don>t you !
#EOT

Peter Schepers · Guest

In article <slrng74fna.5lj.willem@snail.stack.nl>,
Willem <willem@stack.nl> wrote:
[quote]Peter Schepers wrote:
) In article <slrng74cnd.41a.willem@snail.stack.nl>,
) Willem <willem@stack.nl> wrote:
)>Industrial One wrote:
)>) Do I understand correct: hz is one sine loop per second,
)
)>Hz is one <whatever> per second.
)>Your heart beats at roughly 80 Hz.
)
) I think you mean the heart beats 1.333 Hz.

D>oh, you>re right. Roughly, that is. :-)
Did you check the other example (frequency of the moon) ? I made
a mistake there too, although it is not by an order of magnitude...
[/quote]
Well, I originally calc>d .41uHz instead of your number.

PS.

Peter Schepers · Guest

In article <DaidnQv0G_P22e_VnZ2dnUVZ8vqdnZ2d@posted.plusnet>,
Don Pearce <nospam@nospam.com> wrote:
[quote]Peter Schepers wrote:
In article <slrng74cnd.41a.willem@snail.stack.nl>,
Willem <willem@stack.nl> wrote:
Industrial One wrote:
) Do I understand correct: hz is one sine loop per second,

Hz is one <whatever> per second.
Your heart beats at roughly 80 Hz.

I think you mean the heart beats 1.333 Hz.

Better do some exercise then. Mine is more like 0.85Hz.
[/quote]
I was using his 80 beats/min, not mine.

PS.

Steven Sullivan · Guest

In rec.audio.tech Earl Kiosterud <someone@nowhere.com> wrote:

[quote]DVD audio is just for marketing. No one, with the possible exception of a few young people
who can hear above 20 KHz, and many dogs, can hear the difference between regular 44.1K
16-bit audio and 96 or 192K sampling and 24 bits -- it>s been proven, though some will tell
you they can. It>s something they call "resolution" for which they have an altar, dogma and
lots of ritual. They get this dreamy look in their eyes. Challenge it, and their veins pop
out and they go on rampages. It>s likely that much of the stuff you get on DVD-audio discs
is better stuff, and has been more meticulously recorded, hence the good sound of many of
them.
[/quote]
Actually, with the advent of DVD-A ripping software, I>ve found that the stereo mixes on many
of them (the rock/pop ones at least) are just as dynamically compressed as their modern CD
counterparts. And this at 24 bits! Sheer lunacy.

The 'hi rez' format where you are more likely to get 'full-range' dynamics is SACD, due to
restrictions built in to the Scarlet Book spec. But even there it>s now possible to squeeze
the range, from what I hear.

In any case, there>s no reason why these full-range recordings couldn>t be presented on
good old 16-bit CD. The perversion of one of the original benefits of Redbook (extended
dynamic range) into the 'new' benefit (heretofore unheard-of amounts of compression)
is just sad.

--
-S
Poe>s Law: Without a winking smiley or other blatant display of humorous
intent, it is impossible to create a parody of a religious Fundamentalist that
SOMEONE won>t mistake for the real thing.

Steven Sullivan · Guest

In rec.audio.tech Edmund <nomail@hotmail.com> wrote:

[quote]I heard about that tests and it was criticized because the music was
played over a pair of passive loudspeakers with passive filters that
where nowhere near phase linear same problem with electrostatic
speakers with step up transformers . So no matter how much better
SACD or DVDA can be, played over such loudspeakers all the
advantages are down the drain.
Don>t know if this story is true but I very much like to attend such
a listening test an judge for myself.
Did anyone here did attend such a test and on what kind of speakers
was it played?
[/quote]
No doubt the test is being 'swift-boated' by anxious 'audiophiles', but here are the facts
(clipped from
http://www.bostonaudiosociety.org/explanation.htm )

The Principal System

The playback equipment in this system consisted of an Adcom GTP-450 preamp and a Carver M1.5t
power amplifier. Speaker cables were 8 feet of generic 12-gauge stranded wire; the line-level
connecting cables were garden-variety. Three different players were used: a Pioneer DV-563A
universal player, a Sony XA777ES SACD model, and a Yamaha DVD-S1500. The loudspeakers were a
pair of Snell C5s. The CD-standard A/D/A loop was an HHB CDR-850 professional CD recorder

..
..

System 2

We also conducted a series of tests at a local CD/DVD mastering facility. I do not currently
have a detailed equipment list for this venue, but the speakers were very large and capable
high-end monitors, approximately 7 feet tall, and the power amps were sufficient to drive the
speakers to very high levels without audible distortion. Some of the source material for these
trials was a classical production which was then in process at this establishment. Like all
the others, these trials, which were done under a promise of anonymity made to those involved,
produced no significant correlations on music at normal levels.

..
..

System 3

Another series of trials took place at a facility at the University of Massachusetts - Lowell
campus, using students in their recording program as subjects. Their large monitoring room is
custom-designed and has very good acoustics, with a system to match. The system has a center
channel and surrounds, but as in the other trials we restricted ourselves to the two-channel
versions of our sources, so only the left and right were working. The equipment list for the
two channels is:

Klark Teknik DN-410 custom-modified 2-channel parametric equalizer
Stage Accompany PPA-1200 Dig Control class AB amplifier w/ crossover card
Stage Accompany ES-20 Class G amplifier w/crossover card
SLS S1266 3-way monitor (two 12" dynamic drivers, two 6" dynamic drivers, one 6" ribbon
tweeter)
Bag End ELF-1 8-Hz 2-channel low-frequency integrator
Bag End D18E-I dual-18" ELF subwoofer system

This is another professional monitoring system, installed in a large custom-built listening
room with auditorium-type seating. It was capable of very high levels with no audible
distortion as well as imaging of a quality not usually found in large spaces of this kind. We
were interested to find that our informal high-frequency-hearing tests, which we administered
to most of our subjects, indicated that these students had taken unusually good care of their
hearing. Most of them had an upper limit in our test of 16 to 18 kHz
..
..
..
System 4

Another set of trials was performed during the evening at another suburban location . a
custom-built listening room with good acoustics (with the help of an assortment of
professional absorbers and diffusers), very low background noise, and equipment that we
trusted would pass muster with most audiophiles:

Denon 2900 Universal Player with full PartsConnexion mods
Conrad-Johnson 17 LS line stage preamp
Sim Audio Moon 7 monoblock power amplifiers
Quad ESL 989 electrostatic speakers
Muse Model 18 subwoofer, 24 dB/octave crossover @ 50 Hz
Nordost SPM interconnects and speaker cable

//

--
-S
Poe>s Law: Without a winking smiley or other blatant display of humorous
intent, it is impossible to create a parody of a religious Fundamentalist that
SOMEONE won>t mistake for the real thing.

Industrial One · Guest

On Jul 7, 8:14 am, Willem <wil...@stack.nl> wrote:
[quote]Play your sample at half speed and listen to what happens when
you reach 11.025 Khz.
[/quote]
I didn>t notice anything, what was I supposed to hear?

[quote]D>oh, you>re right. Roughly, that is. :-)
Did you check the other example (frequency of the moon) ? I made
a mistake there too, although it is not by an order of magnitude...

Must be a bad day.
[/quote]
Duh, you>ve escaped one year of incarceration of that bottle I
"putput" you in last year, remember? Must still be suffering from
some... psychological damage.

[quote]news:b53f0b1f-ff56-437d-a322-aa4d11f16270@l64g2000hse.googlegroups.com...

On Jul 5, 9:18 pm, "Earl Kiosterud" <some...@nowhere.com> wrote:

Sampling theory tells us that it takes at least two samples per cycle, hence the 44.1 KHz
sample rate. The highest frequency that can be captured is 22.05 KHz (Nyquist
frequency);
frequencies higher than that will create alias frequencies below 22.05 For example, an
audio frequency at 30 KHz would produce an alias frequency component at 14.1 KHz (44.1 -
30). It also produces one at 44.1 + 30, but who cares? The 20KHz audio upper limit
allows
for comfortable guard band to the Nyquist frequency.

Do I understand correct: hz is one sine loop per second, I generate a
sine sweep from 0 - 20 KHz with a specified duration, when I view with
an audio application and zoom 'till individual samples are visible, I
notice that as frequency increases, the sine waves become shorter, and
gradually begin to appear more triangular as the smaller sample
interval makes a perfect, smooth sine shape impossible. Finally, when
it reaches 20 KHz (20,000 sampling rate) the waves have reached their
limit on appearing anything that resembles a sine, and is now a
perfect triangle: one sample at the bottom, one at the top, and one at
the bottom again, like /\/\/\/\/\/\/\/\/\/\. This would technically be
the maximum, but instead, as I continue scrolling, I see the waveform
look something like a private-case of sine waves. This time, a sine
block composed of triangles. What you>re saying is that beyond 22.05
is a hack that simulates higher frequencies, but don>t technically
exist on a digital medium, like the waveform of the sine sweep I
created?

With regard to your swept sine, as you go towards the higher frequency sines, you>ll see
fewer and fewer samples per cycle, until there are barely more than two per cycle at 20 KHz.
Your audio program should not connect them with straight lines. If anything, it should show
them as a post filter (a brick-wall at 20KHz, probably) would see them. That is the
waveform of those samples with the above-Nyquist (above 22.05 KHz) frequency components
removed. It should draw a sine. Anything above 10 KHz should be sinusoidal, since any
other function (waveshape) would need harmonics, which would fall above the 20 KHz point,
and could not appear. For example, there ain>t no such thing as a 15KHz triangle, sawtooth,
square etc., wave in audio. If there were, we>d only hear the fundamental, and that, by
definition, is a single frequency component, thus a sine.
[/quote]
http://i28.tinypic.com/11so6lu.png (note the milliseconds = current hz
since the wave is exactly 22.05 seconds)

No such thing as 15Khz triangle? I beg to differ. But that wasn>t the
point, I was asking if anything above 22.05 was possible to reproduce
digitally, you said no, but frequencies above that can still be
simulated with hacks (combining different frequencies, 14+30=44 KHz,
or... some shit like that) so I asked if that was why (see picture)
the waveform at 20 KHz looked like many units of triangle waves that
form one unit of a sine wave. Wait, 20 KHz... 15+10, the size of that
one sine wave composed of tiny triangles looked about 10 KHz... the
world makes sense again! Is this what you call "modulating 10 KHz onto
15" in order to create that alias wave?

Earl Kiosterud · Guest

"Industrial One" <industrial_one@hotmail.com> wrote in message
news:b53f0b1f-ff56-437d-a322-aa4d11f16270@l64g2000hse.googlegroups.com...
[quote]On Jul 5, 9:18 pm, "Earl Kiosterud" <some...@nowhere.com> wrote:

Sampling theory tells us that it takes at least two samples per cycle, hence the 44.1 KHz
sample rate. The highest frequency that can be captured is 22.05 KHz (Nyquist
frequency);
frequencies higher than that will create alias frequencies below 22.05 For example, an
audio frequency at 30 KHz would produce an alias frequency component at 14.1 KHz (44.1 -
30). It also produces one at 44.1 + 30, but who cares? The 20KHz audio upper limit
allows
for comfortable guard band to the Nyquist frequency.

Do I understand correct: hz is one sine loop per second, I generate a
sine sweep from 0 - 20 KHz with a specified duration, when I view with
an audio application and zoom 'till individual samples are visible, I
notice that as frequency increases, the sine waves become shorter, and
gradually begin to appear more triangular as the smaller sample
interval makes a perfect, smooth sine shape impossible. Finally, when
it reaches 20 KHz (20,000 sampling rate) the waves have reached their
limit on appearing anything that resembles a sine, and is now a
perfect triangle: one sample at the bottom, one at the top, and one at
the bottom again, like /\/\/\/\/\/\/\/\/\/\. This would technically be
the maximum, but instead, as I continue scrolling, I see the waveform
look something like a private-case of sine waves. This time, a sine
block composed of triangles. What you>re saying is that beyond 22.05
is a hack that simulates higher frequencies, but don>t technically
exist on a digital medium, like the waveform of the sine sweep I
created?

[/quote]

With regard to your swept sine, as you go towards the higher frequency sines, you>ll see
fewer and fewer samples per cycle, until there are barely more than two per cycle at 20 KHz.
Your audio program should not connect them with straight lines. If anything, it should show
them as a post filter (a brick-wall at 20KHz, probably) would see them. That is the
waveform of those samples with the above-Nyquist (above 22.05 KHz) frequency components
removed. It should draw a sine. Anything above 10 KHz should be sinusoidal, since any
other function (waveshape) would need harmonics, which would fall above the 20 KHz point,
and could not appear. For example, there ain>t no such thing as a 15KHz triangle, sawtooth,
square etc., wave in audio. If there were, we>d only hear the fundamental, and that, by
definition, is a single frequency component, thus a sine.

A sine has only one frequency component in the spectrum, and a single frequency component in
the spectrum is a sine.

There>s your sine.
--
Earl

Don Pearce · Guest

Peter Schepers wrote:
[quote]In article <slrng74cnd.41a.willem@snail.stack.nl>,
Willem <willem@stack.nl> wrote:
Industrial One wrote:
) Do I understand correct: hz is one sine loop per second,

Hz is one <whatever> per second.
Your heart beats at roughly 80 Hz.

I think you mean the heart beats 1.333 Hz.

PS.

[/quote]
Better do some exercise then. Mine is more like 0.85Hz.

It has to be said, though, that in virtually every calculation that
involves frequency, the number you actually want is radians per second -
2pi * normal frequency. It would make life much easier if the world
adopted that measure.

d

Arny Krueger · Guest

<glennchan@gmail.com> wrote in message
news:72cb2187-c9c3-4fed-8aae-0f2ad262ad3a@t54g2000hsg.googlegroups.com
[quote]To take a contrarians position here:

Here is a good article for understanding sampling theory:
http://www.wescottdesign.com/articles/Sampling/sampling.html

When you sample a signal, you have to tradeoff between
frequency response, aliasing, and ringing artifacts. For
audio I believe it>s ok to have ringing since we don>t
notice it.

On reproducing that signal, there is that set of
tradeoffs a second time. So you can lose frequency
response there again.
[/quote]
All sorts of bad things can happen, and back in the late 1970s and early
1980s they did sometimes happen.

But, its 30 years later, and digital has become very efficient and
cost-effective.

[quote]In practical systems, you aren>t working with idealized
sinc filters (brickwall) so there is some dropoff in
frequency response when you sample that signal and again
when you reproduce it as sound.
[/quote]
In modern good digital systems, most of these problems are confined to > 90%
of the Nyquist frequency.

[quote]So depending on what
analog filters cost, etc. etc. there might be some sense
in going with 96khz systems.
[/quote]
Never happened in the past 30 years. Remember that the Nyquist frequency of
96 KHz is 48 KHz. All we really need to do is have good performance up to
16 KHz, 20 KHz at the most. At no time in the past 30 years has good digital
audio been so bad that you needed > 100% safety margin for acceptable
performance.

[quote]It definitely does make
sense to sample at 96khz at acquisition... the
oversampling is beneficial (if you sample at 48khz, you
can>t get very good frequency response because the analog
filters won>t let you do that).
[/quote]
Analog filters operating at or below 48 KHz have not been part of a modern
digital audio system for at least a decade.

[quote]2- Anyways this is just speculating. The real way to
figure it out is to do a test.
[/quote]
Many of us have been there and done that.

[quote]Unfortunately I haven>t done so myself.
[/quote]
That>s rather evident from the tone of your post. :-(

[quote]But according to one audio engineer,
there is an audible difference.
[/quote]
So what? That>s one guy of how many 100,000>s of people doing technical work
related to audio. I>m sure there are at least a 1,000 audio engineers who
believe that they were victims of an alien abduction. So, when are you going
to make your next pilgrimage to Area 51 or Roswell, to find what you
consider to be an authoritative opinion about high sample rate audio?

geoff · Guest

glennchan@gmail.com wrote:
[quote]To take a contrarian position here:

Here is a good article for understanding sampling theory:
http://www.wescottdesign.com/articles/Sampling/sampling.html

When you sample a signal, you have to tradeoff between frequency
response, aliasing, and ringing artifacts. For audio I believe it>s
ok to have ringing since we don>t notice it.
[/quote]
Ringing has not been such a factor since 20KHZ brick-wall filters went out
of favour several decades ago.

geoff

geoff · Guest

Steven Sullivan wrote:
[quote]In rec.audio.tech Earl Kiosterud <someone@nowhere.com> wrote:

DVD audio is just for marketing. No one, with the possible
exception of a few young people who can hear above 20 KHz, and many
dogs, can hear the difference between regular 44.1K 16-bit audio and
96 or 192K sampling and 24 bits -- it>s been proven, though some
will tell you they can. It>s something they call "resolution" for
which they have an altar, dogma and lots of ritual. They get this
dreamy look in their eyes. Challenge it, and their veins pop out
and they go on rampages. It>s likely that much of the stuff you get
on DVD-audio discs is better stuff, and has been more meticulously
recorded, hence the good sound of many of them.

Actually, with the advent of DVD-A ripping software, I>ve found that
the stereo mixes on many of them (the rock/pop ones at least) are
just as dynamically compressed as their modern CD counterparts. And
this at 24 bits! Sheer lunacy.
[/quote]
Not sheer lunacy in a technical sense, one way ort the other. It>s just a
production decision, like it or not.

geoff


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