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Allen Watson · Guest

In article
<591b3f55-e2e1-408e-8b84-d8e8016cb452@z72g2000hsb.googlegroups.com>,
Industrial One <industrial_one@hotmail.com> wrote:

[quote]Most audio files on the net are recorded at a 44 KHz sampling rate,
but it>s mainly referred as "frequency." Now, humans can only hear up
to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
hearing range?) Obviously, one can notice the difference if the song
was downsampled to 22, so why not coin the standard frequency at 22
KHz instead of 44, why is the number doubled? Also, just where the
hell did the number 44,100 emerge from? Why not 40,000?

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

And if this ain>t the case, why would the sampling rate be called
"frequency?"
[/quote]

I think a graphical explanation or demonstration of audio sampling and
reconstruction processes might help. Take a look at

http://www.aw3rd.us/hearingdigital.htm

Cheers!
- Allen

Pete Fraser · Guest

"geoff" <geoff@nospam-paf.co.nz> wrote in message
news:XOKdnf5nMaKGCe_VnZ2dnUVZ_judnZ2d@giganews.com...

[quote]Ringing has not been such a factor since 20KHZ brick-wall filters went out
of favour several decades ago.
[/quote]
I don>t understand this.
Regular CDs have a 44.1 kHz sampling rate.
If you want a 20 kHz audio bandwidth you need a (fairly) sharp filter.
The fact that you use a mild analog filter, oversample generously,
then filter digitally before / while decimating and writing the data
to disk does not affect the ringing.

The sharp filter is done digitally, is cheaper, and is not dispersive.
The sloppy filter is in analog, and can be minimally dispersive.
However, you still have a sharp filter which will cause ringing.

What am I missing?

Pete

geoff · Guest

Pete Fraser wrote:
[quote]"geoff" <geoff@nospam-paf.co.nz> wrote in message
news:XOKdnf5nMaKGCe_VnZ2dnUVZ_judnZ2d@giganews.com...

Ringing has not been such a factor since 20KHZ brick-wall filters
went out of favour several decades ago.

I don>t understand this.
Regular CDs have a 44.1 kHz sampling rate.
If you want a 20 kHz audio bandwidth you need a (fairly) sharp filter.
The fact that you use a mild analog filter, oversample generously,
then filter digitally before / while decimating and writing the data
to disk does not affect the ringing.

The sharp filter is done digitally, is cheaper, and is not dispersive.
The sloppy filter is in analog, and can be minimally dispersive.
However, you still have a sharp filter which will cause ringing.

What am I missing?
[/quote]

A 20KHz brick-wall filter can have gross ringing at audio frequencies. I>m
not sure that a digital filter has ringing at all, but if it has it is moved
up waaaay higher than audio, and can then be addressed buy a kinder and
gentler analogue filter.

geoff

Pete Fraser · Guest

"geoff" <geoff@nospam-paf.co.nz> wrote in message
news:pZmdnS-QWoMdLO_VnZ2dnUVZ_ofinZ2d@giganews.com...

[quote]A 20KHz brick-wall filter can have gross ringing at audio frequencies.
[/quote]
It will have, if it>s presented with a square wave.

[quote]I>m not sure that a digital filter has ringing at all
[/quote]
It certainly does.

[quote]but if it has it is moved up waaaay higher than audio, and can then be
addressed buy a kinder and gentler analogue filter.
[/quote]
Not really.
A brick wall digital filter will have similar ringing amplitude to that
of a brick wall analog filter. The main difference is that the digital
filter>s
ringing will be symmetrical, whereas the analog filter>s ringing will be
asymmetrical (unless the filter designer has been very careful with
group delay correction).

Earl Kiosterud · Guest

"Industrial One" <industrial_one@hotmail.com> wrote in message
news:ea5a18a7-cf34-429e-bd45-2272e356b53e@34g2000hsh.googlegroups.com...
[quote]
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?
[/quote]
If you feed a 15 KHz triangle to a CD system, you>ll get only a sine out of it. The
harmonics of such a wave are at 45 KHz, 60 KHz, etc, and would not pass the 20 KHz
pre-filter in the ADC. All you have left is the fundamental at 15 Khz. That would be a
sine. Remember what I said about a single frequency in the spectrum, etc. It>s useful to
think in the frequency domain, applying what>s known about frequency components, filters,
aliases, modulation products, bad music, etc., not just in the amplitude domain, where you
think in terms of waveforms.

I don>t know what you mean by "Many units of triangles that form one unit of a sine wave."
I wouldn>t get into all that triangle stuff. Learn the frequency domain fundamentals, and
you>ll be rocking and rolling soon.

I notice that Audacity displays sample points connected with straight lines. I don>t think
that>s meaningful. Other editors show what a post-filter, presumably at 20 KHz, would show,
along with the sample points.

Your posts are a bit confusing where you combine other posts. I need an extra bourbon.
--
Earl

geoff · Guest

Earl Kiosterud wrote:
[quote]I notice that Audacity displays sample points connected with straight
lines. I don>t think that>s meaningful. Other editors show what a
post-filter, presumably at 20 KHz, would show, along with the sample
points.
[/quote]

That would be very presumptive of editing software. SOundForge shows
straight lines between samples - I would be most peeved if SF made an
arbitrary decision about a filter.

geoff

Mr.T · Guest

"geoff" <geoff@nospam-paf.co.nz> wrote in message
news:-a6dnd63EsdiCe_VnZ2dnUVZ_o_inZ2d@giganews.com...
[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.

Not sheer lunacy in a technical sense, one way ort the other. It>s just a
production decision, like it or not.
[/quote]
Personally I>m quite happy we don>t have recordings with over 100dB DNR of
the actual music material. No one would ever be able to listen to them
without an isolation room and perfect hearing. (and your hearing wouldn>t
stay that way long with >110-120dB peaks)
And what would it prove in any case, simply that it can now be done?

Most recordings these days do go a bit far the other way however. :-(

MrT.

Edmund · Guest

On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

[quote]"Edmund" <nomail@hotmail.com> wrote in message
news:48708734$0$17237$bf4948fe@news.tele2.nl

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

As a rule, speakers are nowhere near phase linear, regardless of the
implementation of the crossover.

However, similar tests have been done with transducers that have better
phase response, and same results.
[/quote]
Which transducers and how much better?
[quote]
Furthermore, you are ignoring the fact that linear phase microphones are
only a little bit easier to find, and as a rule they are not used to
record music.
[/quote]
Strange do you know why?
[quote]
same problem with electrostatic speakers with step up transformers .

Same problem with 99,9% (more or less) of all loudspeakers ever made.

So what?
[/quote]
So what? if the problem lies there, then we need to improve
the loudspeakers don>t we?
[quote]
So no matter how much better SACD or DVDA can be, played over such
loudspeakers all the advantages are down the drain.

Even if you were right, you>re basically admitting that SACD and DVDA
have no real world application.
[/quote]
The word admitting suggests that I know it all and even suggests I
"know" that SACD cannot improve anything. I don>t know these things.
[quote]
Don>t know if this story is true but I very much like to attend such a
listening test an judge for myself.

I doubt that, the tests are blind tests.
[/quote]
So?
[quote]
Did anyone here did attend such a test and on what kind of speakers was
it played?

I can guarantee you that they weren>t phase linear.
[/quote]
Can it be that that is why no differences where heard?

Edmund

Arny Krueger · Guest

"Edmund" <nomail@hotmail.com> wrote in message
news:48733c04$0$451$bf4948fe@news.tele2.nl
[quote]On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" <nomail@hotmail.com> wrote in message
news:48708734$0$17237$bf4948fe@news.tele2.nl

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

As a rule, speakers are nowhere near phase linear,
regardless of the implementation of the crossover.

However, similar tests have been done with transducers
that have better phase response, and same results.

Which transducers and how much better?
[/quote]
Headphones - full range and no crossovers.

[quote]Furthermore, you are ignoring the fact that linear phase
microphones are only a little bit easier to find, and as
a rule they are not used to record music.

Strange do you know why?
[/quote]
Phase linear response does not give sufficient audible advantages to offset
the difficulty of building loudspeakers that have it.

[quote]same problem with electrostatic speakers with step up
transformers .

Same problem with 99,9% (more or less) of all
loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to
improve the loudspeakers don>t we?
[/quote]
You>ve missed two points:

(1) is that in the current context were non phase-linear speakers are the
rule, DVD-A and SACD formats have no audible benefits.

(2) Even when listening using headphones, which have often have better phase
response than speakers, DVD-A and SACD formats still have no audible
benefits.

[quote]So no matter how much better SACD or DVDA can be,
played over such loudspeakers all the advantages are
down the drain.

Even if you were right, you>re basically admitting that
SACD and DVDA have no real world application.

The word admitting suggests that I know it all and even
suggests I "know" that SACD cannot improve anything. I
don>t know these things.

Don>t know if this story is true but I very much like
to attend such a listening test an judge for myself.

I doubt that the tests were blind tests.

So?
[/quote]
Then they were invalidated by listener bias.

[quote]Did anyone here did attend such a test and on what kind
of speakers was it played?

I can guarantee you that they weren>t phase linear.

Can it be that is why no differences where heard?
[/quote]
No.

Arny Krueger · Guest

"Pasi Ojala" <albert@mustatilhi.cs.tut.fi> wrote in message
news:slrng73qof.9d3.albert@mustatilhi.cs.tut.fi
[quote]On 2008-07-07, glennchan@gmail.com <glennchan@gmail.com
wrote:
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.

Ringing is not an artifact, it is how a band-limited
signal behaves. It may look funny on the computer screen,
but you don>t hear it because there is nothing wrong with
it in the first place.
[/quote]
A very relevant insight. The so-called ringing is not due to the addition of
sound, it is due to the absence of sound.

In classic ringing, high frequency components of the signal are emphasized
by a rising frequency response characteristic.

If this happens in the audible frequency range, then the timbre of the sound
is changed.

In the case of so-called ringing due to a sharp frequency cut-off, there is
no emphasis of high frequencies. Therefore, there is no logical reason to
expect a timbre change.

The nature of the problem of high frequency ringing due to a sharp frequency
cut-off is actually due to people>s brains misinterpreting an oscilloscope
trace.

Edmund · Guest

On Tue, 08 Jul 2008 06:12:30 -0400, Arny Krueger wrote:

[quote]"Edmund" <nomail@hotmail.com> wrote in message
news:48733c04$0$451$bf4948fe@news.tele2.nl
On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" <nomail@hotmail.com> wrote in message
news:48708734$0$17237$bf4948fe@news.tele2.nl

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

As a rule, speakers are nowhere near phase linear, regardless of the
implementation of the crossover.

However, similar tests have been done with transducers that have
better phase response, and same results.

Which transducers and how much better?

Headphones - full range and no crossovers.
[/quote]
Hmm, that is exactly what I had in mind :-)
[quote]
Furthermore, you are ignoring the fact that linear phase microphones
are only a little bit easier to find, and as a rule they are not used
to record music.

Strange do you know why?

Phase linear response does not give sufficient audible advantages to
offset the difficulty of building loudspeakers that have it.
[/quote]
For me that is no reason not to try to record music the best
way possible.
[quote]
same problem with electrostatic speakers with step up transformers .

Same problem with 99,9% (more or less) of all loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to improve the
loudspeakers don>t we?

You>ve missed two points:

(1) is that in the current context were non phase-linear speakers are
the rule, DVD-A and SACD formats have no audible benefits.
[/quote]
Again if the current loudspeakers aren>t perfect, that is no
reason to stop searching for a "perfect" way to record music!
[quote]
(2) Even when listening using headphones, which have often have better
phase response than speakers, DVD-A and SACD formats still have no
audible benefits.
[/quote]
that>s weird.
[quote]
So no matter how much better SACD or DVDA can be, played over such
loudspeakers all the advantages are down the drain.

Even if you were right, you>re basically admitting that SACD and DVDA
have no real world application.

The word admitting suggests that I know it all and even suggests I
"know" that SACD cannot improve anything. I don>t know these things.

Don>t know if this story is true but I very much like to attend such
a listening test an judge for myself.

I doubt that the tests were blind tests.

So?

Then they were invalidated by listener bias.

Did anyone here did attend such a test and on what kind of speakers
was it played?

I can guarantee you that they weren>t phase linear.

Can it be that is why no differences where heard?

No.
[/quote]
Suppose you are right what should be improved first in order
to realize a better sound reproduction? I easily hear the
difference between an unplugged chancel and a recorded one.:-)

Edmund

Arny Krueger · Guest

"Edmund" <nomail@hotmail.com> wrote in message
news:487395f2$0$26728$bf4948fe@news.tele2.nl
[quote]On Tue, 08 Jul 2008 06:12:30 -0400, Arny Krueger wrote:

"Edmund" <nomail@hotmail.com> wrote in message
news:48733c04$0$451$bf4948fe@news.tele2.nl
On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" <nomail@hotmail.com> wrote in message
news:48708734$0$17237$bf4948fe@news.tele2.nl

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

As a rule, speakers are nowhere near phase linear,
regardless of the implementation of the crossover.

However, similar tests have been done with transducers
that have better phase response, and same results.

Which transducers and how much better?

Headphones - full range and no crossovers.

Hmm, that is exactly what I had in mind :-)

Furthermore, you are ignoring the fact that linear
phase microphones are only a little bit easier to
find, and as a rule they are not used to record music.

Strange do you know why?

Phase linear response does not give sufficient audible
advantages to offset the difficulty of building
loudspeakers that have it.

For me that is no reason not to try to record music the
best way possible.
[/quote]
The recording/playback process is rife with serious imperfections, but for
the past 25 years, the digital medium has not been a significant problem.

[quote]same problem with electrostatic speakers with step up
transformers .

Same problem with 99,9% (more or less) of all
loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to
improve the loudspeakers don>t we?

You>ve missed two points:

(1) is that in the current context were non phase-linear
speakers are the rule, DVD-A and SACD formats have no
audible benefits.

Again if the current loudspeakers aren>t perfect, that is
no reason to stop searching for a "perfect" way to record
music!
[/quote]
Nothing is perfect. The weakest links are more worthy of our time and effort
than the strongest ones.

[quote](2) Even when listening using headphones, which have
often have better phase response than speakers, DVD-A
and SACD formats still have no audible benefits.

that>s weird.
[/quote]
Not at all. The digital mediums have not been a serious problem for over 25
years.

[quote]So no matter how much better SACD or DVDA can be,
played over such loudspeakers all the advantages are
down the drain.

Even if you were right, you>re basically admitting
that SACD and DVDA have no real world application.

The word admitting suggests that I know it all and even
suggests I "know" that SACD cannot improve anything. I
don>t know these things.

Don>t know if this story is true but I very much like
to attend such a listening test an judge for myself.

I doubt that the tests were blind tests.

So?

Then they were invalidated by listener bias.

Did anyone here did attend such a test and on what
kind of speakers was it played?

I can guarantee you that they weren>t phase linear.

Can it be that is why no differences where heard?

No.

Suppose you are right what should be improved first in
order to realize a better sound reproduction?
[/quote]
Venues and transducers. Rooms, speakers, and microphones.

[quote]I easily hear the difference between an unplugged chancel and a
recorded one.:-)
[/quote]
Having years of experiences, involving 100s of recordings I can tell you
that the worst violence is done to recorded sound by loudspeakers and
listening rooms.

In military life there is a saying: "Amateurs worry about strategy,
professionals worry about logistics". In audio, amateurs worry about
digital recording media, and professionals worry about venues and
transducers.

geoff · Guest

Arny Krueger wrote:
[quote]
Phase linear response does not give sufficient audible advantages to
offset the difficulty of building loudspeakers that have it.
[/quote]
I want a phase-linear room.

geoff

ChrisCoaster · Guest

On Jul 5, 11:47 pm, Industrial One <industrial_...@hotmail.com> wrote:
[quote]Most audio files on the net are recorded at a 44 KHz sampling rate,
but it>s mainly referred as "frequency." Now, humans can only hear up
to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
hearing range?) Obviously, one can notice the difference if the song
was downsampled to 22, so why not coin the standard frequency at 22
KHz instead of 44, why is the number doubled? Also, just where the
hell did the number 44,100 emerge from? Why not 40,000?

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

And if this ain>t the case, why would the sampling rate be called
"frequency?"
___________________[/quote]
Very simple. CDs have two channels of sound: Left, Right, A, B, 1, 2,
whatever you want to call them. L22kHz + R22kHz = 44kHz.

If you lower the sampling rate to 22kHz, each channel gets a freq
response up to only 11kHz. Fine for recorded speech, or archiving
78RPM & wax cylinder recordings, but quite insufficient for anything
recorded after 1950.

-CC

ChrisCoaster · Guest

On Jul 5, 11:47 pm, Industrial One <industrial_...@hotmail.com> wrote:
[quote]Most audio files on the net are recorded at a 44 KHz sampling rate,
but it>s mainly referred as "frequency." Now, humans can only hear up
to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
hearing range?) Obviously, one can notice the difference if the song
was downsampled to 22, so why not coin the standard frequency at 22
KHz instead of 44, why is the number doubled? Also, just where the
hell did the number 44,100 emerge from? Why not 40,000?

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

And if this ain>t the case, why would the sampling rate be called
"frequency?"
______________________________[/quote]
Google is really screwed up tonight so I>ll have to repost - sorry
Usenetters!!

The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
reason: Two channels - Left, Right, A,B, whatever you want to call
them. Each channel gets up to 22kHz.

This is the second time I>ve tried to post this so I hope this time it
goes through.

-ChrisCoaster


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