[Robert Peirce]

I run Pure Music on my Mac. Presently, I use Airfoil to send the signal
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs).

I have been trying to find a substitute for the AppleTV, but so far all
I have got is the Squeezebox Touch. I say "but" because it has its own
software that resides on the Mac and I have not been able to find out if
it can receive input from Pure Music or not. Does anybody know?

The web site suggests the Squeezebox can read any file on the computer,
which is great, except Pure Music already does that and allows many
useful manipulations. For example, it will accept up to 384/32 and
downsample it to 96/24. It will also upsample 44.1/16 to 96/24. Or any
other standard sample rate and 16, 24 or 32 bit words.

As far as I have been able to determine, the Squeezebox only passes
through what it receives, and that is great IF it can receive the output
from Pure Music. So, does anybody actually know if it can do that?

A related issue, but not critical, is that the software I am actually
running is Pure Vinyl. It is primarily designed for digitizing vinyl
recordings but it included Pure Music which I have grown to like a lot.
At present, I feed it directly from my pre-amp to the mic input on the
MAC, which works OK, but a two-way solution would be even better than
just using the player. Pure Vinyl can handle up to 384/32 if there is a
way to feed that to the Mac.

[Edmund[_2_]]

On Mon, 29 Aug 2011 03:10:45 +0000, Robert Peirce wrote:

I run Pure Music on my Mac. Presently, I use Airfoil to send the signa=
l
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs).
=20
I have been trying to find a substitute for the AppleTV, but so far all
I have got is the Squeezebox Touch. I say "but" because it has its own
software that resides on the Mac and I have not been able to find out i=
f
it can receive input from Pure Music or not. Does anybody know?
=20
The web site suggests the Squeezebox can read any file on the computer,
which is great, except Pure Music already does that and allows many
useful manipulations. For example, it will accept up to 384/32 and
downsample it to 96/24.

I wonder what happens to the anti alias filter in that case.

Edmund

[Audio Empire]

On Sun, 28 Aug 2011 20:10:45 -0700, Robert Peirce wrote
(in article ):

I run Pure Music on my Mac. Presently, I use Airfoil to send the signal
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs).

I have been trying to find a substitute for the AppleTV, but so far all
I have got is the Squeezebox Touch. I say "but" because it has its own
software that resides on the Mac and I have not been able to find out if
it can receive input from Pure Music or not. Does anybody know?

The way the Squeezebox works is that you pick a folder on your computer (Mac,
Windows or Linux) and designate it as your music folder in the Squeezebox
Touch music server software. Any supported format (and there are lots of them
but DSD files are NOT among them)) will then show up on the Squeezebox Touch
menu. So, if you pick whatever folder that Pure Music stores its music files
in as your designated Squeezebox server folder, and the files in it are WAVE,
FLAC, ALC, etc. and no more than 96KHz in sampling rate, it should work. What
I do is to use my iTunes folder as the Squeezebox Touch server folder, and I
put Apple Aliases of my "hi-rez" 24/96 files in the iTunes folder and those
files are then available right along with my iTunes catalog on my Squeezebox
Touch for playback. That way there is no need to actually move the files from
where they naturally reside or to duplicate them in order for the Squeezebox
Touch server software to find and use them.


The web site suggests the Squeezebox can read any file on the computer,
which is great, except Pure Music already does that and allows many
useful manipulations. For example, it will accept up to 384/32 and
downsample it to 96/24. It will also upsample 44.1/16 to 96/24. Or any
other standard sample rate and 16, 24 or 32 bit words.

Squeezebox server deals with files only. It does not interact with either the
iTunes application or any other program (including Pure Music). If Pure Music
allows you to make permanently altered copies of the files it manipulates or
up or down-samples to 96 KHz, then Squeezebox Touch will work with those
altered files. Be advised that the Logitech device only works with 96 KHz or
lower sample rates. What I do is use one of the digital outputs on the
Squeezebox Touch and feed that into a stand-alone up-sampling engine. Then I
feed the up-sampled 24/96 SPDIF signal to my outboard 24/192 DAC.

As far as I have been able to determine, the Squeezebox only passes
through what it receives, and that is great IF it can receive the output
from Pure Music. So, does anybody actually know if it can do that?

See above.

A related issue, but not critical, is that the software I am actually
running is Pure Vinyl. It is primarily designed for digitizing vinyl
recordings but it included Pure Music which I have grown to like a lot.
At present, I feed it directly from my pre-amp to the mic input on the
MAC, which works OK, but a two-way solution would be even better than
just using the player. Pure Vinyl can handle up to 384/32 if there is a
way to feed that to the Mac.

Tower Macs have have both an SPDIF input and output on them and should handle
384/32. However, be advised, that the only thing that such a high sampling
frequency buys you is huge digital files. Today's 32-bit is usually 24-bit
digital with an 8-bit floating-point mantissa. A 32 bit data stream records
65,000 times the dynamic range of a16 bit CD audio. This gives a dynamic
range that is so much higher than either the range of human perception or the
state-of-the-art noise floor in modern electronics that it's meaningless and
quite superfluous. It's like insisting that the film in your camera be able
to capture everything from the extreme infrared all the way out to X-Rays
when humans can only see red through violet light.

Also, while 32-bit may be enticing in the "more-has-got-to-be-better"
philosophy, most DACs can't handle true 32-bit and ignore the top 8-bits in a
32-bit floating point coding scheme. A rule of thumb that I have gleaned from
lots of experience and study is that 24-bit/48KHz quantization is
indistinguishable, audibly, from bit-depths higher than 24-bit or from
sampling rates higher than 48 KHz. Nobody's hearing is good enough to
distinguish any theoretical or practical advantage to frequency responses
that go much above 22KHz, or dynamic ranges that exceed 120 dB.

If 24-bit has any REAL WORLD advantage it is that it allows for lower peak
levels on recording which lessens the danger of over-modulation without the
resultant recording ending up down in the mud where distortion and
quantization noise increase as the signal toggles fewer and fewer bits.

[Audio Empire]

On Mon, 29 Aug 2011 04:56:46 -0700, Edmund wrote
(in article ):

On Mon, 29 Aug 2011 03:10:45 +0000, Robert Peirce wrote:

I run Pure Music on my Mac. Presently, I use Airfoil to send the signa=
l
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs).
=20
I have been trying to find a substitute for the AppleTV, but so far all
I have got is the Squeezebox Touch. I say "but" because it has its own
software that resides on the Mac and I have not been able to find out i=
f
it can receive input from Pure Music or not. Does anybody know?
=20
The web site suggests the Squeezebox can read any file on the computer,
which is great, except Pure Music already does that and allows many
useful manipulations. For example, it will accept up to 384/32 and
downsample it to 96/24.

I wonder what happens to the anti alias filter in that case.

Edmund


You would certainly have to move the filter up in frequency in order to use
that extra bandwidth, otherwise the filter would simply treat the signal like
any other digital audio stream and start to roll-off the frequency response
above 22 KHz. What the use of high sampling rates does is to move any
quantization noise further out of the passband as the sampling rate
increases. Whether this is of any practical consideration is debatable.
Double-blind tests seem to show that this is no real consequence, but some
will hotly debate the point.

Seems to me that I read somewhere that modern 24/192 DAC chips move the
antialiasing frequency as the sample rate increases. If it didn't do that and
left the antialiasing filter "cutoff" at 22.05 KHz (which is normal for
16-bit/44.1 KHz CD) then any advantage (real or imagined) to higher bit-rate
audio would be wasted as everything would be severely rolled off above 22.05
KHz regardless of bit rate. .

[Robert Peirce]

In article ,
Audio Empire wrote:

On Sun, 28 Aug 2011 20:10:45 -0700, Robert Peirce wrote
(in article ):

I run Pure Music on my Mac. Presently, I use Airfoil to send the signal
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs).

I have been trying to find a substitute for the AppleTV, but so far all
I have got is the Squeezebox Touch. I say "but" because it has its own
software that resides on the Mac and I have not been able to find out if
it can receive input from Pure Music or not. Does anybody know?

The way the Squeezebox works is that you pick a folder on your computer (Mac,
Windows or Linux) and designate it as your music folder in the Squeezebox
Touch music server software. Any supported format (and there are lots of them
but DSD files are NOT among them)) will then show up on the Squeezebox Touch
menu. So, if you pick whatever folder that Pure Music stores its music files
in as your designated Squeezebox server folder, and the files in it are WAVE,
FLAC, ALC, etc. and no more than 96KHz in sampling rate, it should work. What
I do is to use my iTunes folder as the Squeezebox Touch server folder, and I
put Apple Aliases of my "hi-rez" 24/96 files in the iTunes folder and those
files are then available right along with my iTunes catalog on my Squeezebox
Touch for playback. That way there is no need to actually move the files from
where they naturally reside or to duplicate them in order for the Squeezebox
Touch server software to find and use them.

Damn! That won't work. Pure Music doesn't save files. It is server
software that uses the iTunes library to find files but its own software
to play them. In that regard, I guess it is similar to Squeezebox's
software.

Pure Music recommends using USB or firewire DACS to drive your stereo
except distance problems, and not wanting to string wires or optical
lines all over the place, forces me to use ethernet. Airfoil and
AppleTV work great for this as long as I don't want to play anything but
standard CDs. I am trying to figure out how to play high res, up to
96/24 and eventually higher.

I think, for Pure Music to be able to use a device, that it must see it.
I think it has to show up in the Audio Midi setup app. This seems to be
true for USB and firewire devices, but I don't know about ethernet
devices. The technology involved is getting way beyond my hardware
knowledge.

As a last resort, I may have to run an optical line from my computer to
my DAC. It would require a run of about 25', but I think it would work
better than an analog pair from a USB/firewire DAC to my stereo.

[Robert Peirce]

In article ,
Audio Empire wrote:

A rule of thumb that I have gleaned from
lots of experience and study is that 24-bit/48KHz quantization is
indistinguishable, audibly, from bit-depths higher than 24-bit or from
sampling rates higher than 48 KHz. Nobody's hearing is good enough to
distinguish any theoretical or practical advantage to frequency responses
that go much above 22KHz, or dynamic ranges that exceed 120 dB.

I can't actually hear any difference. At my age I am lucky to hear
anything over about 10Khz.

What I have noticed is less fatigue. This is totally subjective and
probably can't be measured. It is the difference between wanting to
listen to music all day and feeling forced to turn it off after a couple
of hours. Some people may not even notice it.

[Edmund[_2_]]

On Mon, 29 Aug 2011 23:15:38 +0000, Audio Empire wrote:

On Mon, 29 Aug 2011 04:56:46 -0700, Edmund wrote (in article
):
=20
On Mon, 29 Aug 2011 03:10:45 +0000, Robert Peirce wrote:
=20
I run Pure Music on my Mac. Presently, I use Airfoil to send the
signa=3D
l
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs). =3D20
I have been trying to find a substitute for the AppleTV, but so far
all I have got is the Squeezebox Touch. I say "but" because it has
its own software that resides on the Mac and I have not been able to
find out i=3D
f
it can receive input from Pure Music or not. Does anybody know? =3D2=
0
The web site suggests the Squeezebox can read any file on the
computer, which is great, except Pure Music already does that and
allows many useful manipulations. For example, it will accept up to
384/32 and downsample it to 96/24.
=20
I wonder what happens to the anti alias filter in that case.
=20
Edmund
=20
=20
You would certainly have to move the filter up in frequency in order to
use that extra bandwidth,=20

I don't think one get extra bandwidth when you DOWNSAMPLE 384/32 to 96/2=
4,
and the filter should be corrected to 40 kHz or so.

otherwise the filter would simply treat the
signal like any other digital audio stream and start to roll-off the
frequency response above 22 KHz. What the use of high sampling rates
does is to move any quantization noise further out of the passband as
the sampling rate increases. Whether this is of any practical
consideration is debatable. Double-blind tests seem to show that this i=
s
no real consequence, but some will hotly debate the point.

And we can debate for a long time since there are no recordings recorded=20
with much higher frequencies then 22kHz.
I for one am still waiting for such real high res recordings.
( don't know about SACD since I cannot make such files visible.... yet )
=20
Seems to me that I read somewhere that modern 24/192 DAC chips move the
antialiasing frequency as the sample rate increases. If it didn't do
that and left the antialiasing filter "cutoff" at 22.05 KHz (which is
normal for 16-bit/44.1 KHz CD) then any advantage (real or imagined) to
higher bit-rate audio would be wasted as everything would be severely
rolled off above 22.05 KHz regardless of bit rate. .

True

Edmund

[Arny Krueger[_4_]]

"Audio Empire" wrote in message
...

You would certainly have to move the filter up in frequency in order to
use
that extra bandwidth, otherwise the filter would simply treat the signal
like
any other digital audio stream and start to roll-off the frequency
response
above 22 KHz. What the use of high sampling rates does is to move any
quantization noise further out of the passband as the sampling rate
increases.

Whether that happens very strongly depends whether or not noise shaping is
used. If unshaped quantization is used, the actual change in the amount of
quantization noise at the most audible frequencies is minor or even moot.

Whether this is of any practical consideration is debatable.
Double-blind tests seem to show that this is no real consequence,

True, and at this point the number of such tests performed by both experts
and talented amateurs is very significant.

but some will hotly debate the point.

It takes reliance on sighted evaluations in a situation where they don't
work to reach or support this conclusion. This is one of those cases where
its not hard to hear what is there to hear. It is also fairly easy to set up
training runs where the effect is highly audible.

Seems to me that I read somewhere that modern 24/192 DAC chips move the
antialiasing frequency as the sample rate increases.

Virtually all of them. It is a natural consequence of digital filtering. It
costs extra to keep that from happening. So, it is almost never done.

[Arny Krueger[_4_]]

"Edmund" wrote in message
...
On Mon, 29 Aug 2011 23:15:38 +0000, Audio Empire wrote:

And we can debate for a long time since there are no recordings recorded
with much higher frequencies then 22kHz.
I for one am still waiting for such real high res recordings.
( don't know about SACD since I cannot make such files visible.... yet )

Then you are waiting for the Second Coming and the New Heaven and the New
Earth! ;-)

Bt that I mean that the (current) laws of physics are running strongly
against you.

Creating high frequencies at high amplitudes takes more energy.

For thousands of years people have been designing and building musical
instruments to be as audible and pleasing as possible. Evolution did the
same with our voices. This means that they very intentionally *avoid wasting
energy* with frequencies that are barely heard or not heard at all.

Good luck trying to get the musicans of the world to switch over to musical
instruments that make SACDs sound different than CDs! ;-)

[Arny Krueger[_4_]]

"Audio Empire" wrote in message
...

A rule of thumb that I have gleaned from
lots of experience and study is that 24-bit/48KHz quantization is
indistinguishable, audibly, from bit-depths higher than 24-bit or from
sampling rates higher than 48 KHz. Nobody's hearing is good enough to
distinguish any theoretical or practical advantage to frequency responses
that go much above 22KHz, or dynamic ranges that exceed 120 dB.

Few if anybody's recording space or listening room is good enough, either.

JJ had a listening room at the AT&T labs that was designed to support a -100
dB noise floor for practical and typical listening levels per EBU
recommendation BS 1116.. He can tell you about the slings and arrows and
costs of actually doing such a thing. If memory serves, a freeway a fraction
of a mile away was one of the hurdles that they had to overcome, all at
great cost to the management.

If 24-bit has any REAL WORLD advantage it is that it allows for lower peak
levels on recording which lessens the danger of over-modulation

Audible distortion due to approaching the point of over-modulation does not
exist in the digital domain, and only exists for the upper 1 to 3 dB in the
analog domain except for things like magnetic tape. Adding some 50 dB of
dynamic range with 24 bits versus 16 bits looks great on paper, but it does
not help with problems that small.

without the resultant recording ending up down in the mud where
distortion and
quantization noise increase as the signal toggles fewer and fewer bits.

Quantization noise is a straw man with modern digital equipment because its
hard to do Sigma-Delta converters without introducing dither or something
like it.

[Edmund[_2_]]

On Tue, 30 Aug 2011 12:17:36 +0000, Arny Krueger wrote:

"Edmund" wrote in message
... On Mon, 29 Aug 2011 23:15:38
+0000, Audio Empire wrote:
=20
And we can debate for a long time since there are no recordings
recorded with much higher frequencies then 22kHz. I for one am still
waiting for such real high res recordings. ( don't know about SACD
since I cannot make such files visible.... yet )
=20
Then you are waiting for the Second Coming and the New Heaven and the
New Earth! ;-)
=20
Bt that I mean that the (current) laws of physics are running strongly
against you.
=20
Creating high frequencies at high amplitudes takes more energy.

Not so much and even so, I don't care.
=20
For thousands of years people have been designing and building musical
instruments to be as audible and pleasing as possible. Evolution did th=
e
same with our voices. This means that they very intentionally *avoid
wasting energy* with frequencies that are barely heard or not heard at
all.
=20
Good luck trying to get the musicans of the world to switch over to
musical instruments that make SACDs sound different than CDs! ;-)

I read " there is life above 20kHz " ( or something ) and there are=20
quite a few instruments that produce sounds above 20k. Then every=20
change from silence requires an infinite bandwidth to make it perfect.
Just shaking your keyring produces frequencies above 20k but -agreed- tha=
t isn't
a sound that is recognizable as music is but it does show that such high
frequencies are easily produced.

So as long noone is recording these high frequencies in real music it rem=
ains
pointless to discus whether or not it is audible.

Edmund



=20

[Audio Empire]

On Tue, 30 Aug 2011 03:47:44 -0700, Robert Peirce wrote
(in article ):

[quoted text deleted -- deb]

As a last resort, I may have to run an optical line from my computer to
my DAC. It would require a run of about 25', but I think it would work
better than an analog pair from a USB/firewire DAC to my stereo.


Excellent build quality TOSLINK optical cables:

http://www.mycablemart.com/store/car...duct_list&c=11

Part # HA-TOS-25 or KM-TOS-35

The usual disclaimer applies. I have no commercial connection with "My
Cable Mart" , other than just being a satisfied repeat customer.

[Audio Empire]

On Tue, 30 Aug 2011 03:47:52 -0700, Robert Peirce wrote
(in article ):

In article ,
Audio Empire wrote:

A rule of thumb that I have gleaned from
lots of experience and study is that 24-bit/48KHz quantization is
indistinguishable, audibly, from bit-depths higher than 24-bit or from
sampling rates higher than 48 KHz. Nobody's hearing is good enough to
distinguish any theoretical or practical advantage to frequency responses
that go much above 22KHz, or dynamic ranges that exceed 120 dB.

I can't actually hear any difference. At my age I am lucky to hear
anything over about 10Khz.

And, of course, there is that....

What I have noticed is less fatigue. This is totally subjective and
probably can't be measured. It is the difference between wanting to
listen to music all day and feeling forced to turn it off after a couple
of hours. Some people may not even notice it.

But do you know for sure that's due to extended supersonic frequency
response?

[Audio Empire]

On Tue, 30 Aug 2011 05:18:27 -0700, Arny Krueger wrote
(in article ):

"Audio Empire" wrote in message
...

A rule of thumb that I have gleaned from
lots of experience and study is that 24-bit/48KHz quantization is
indistinguishable, audibly, from bit-depths higher than 24-bit or from
sampling rates higher than 48 KHz. Nobody's hearing is good enough to
distinguish any theoretical or practical advantage to frequency responses
that go much above 22KHz, or dynamic ranges that exceed 120 dB.

Few if anybody's recording space or listening room is good enough, either.

JJ had a listening room at the AT&T labs that was designed to support a -100
dB noise floor for practical and typical listening levels per EBU
recommendation BS 1116.. He can tell you about the slings and arrows and
costs of actually doing such a thing. If memory serves, a freeway a fraction
of a mile away was one of the hurdles that they had to overcome, all at
great cost to the management.

If 24-bit has any REAL WORLD advantage it is that it allows for lower peak
levels on recording which lessens the danger of over-modulation

Audible distortion due to approaching the point of over-modulation does not
exist in the digital domain, and only exists for the upper 1 to 3 dB in the
analog domain except for things like magnetic tape. Adding some 50 dB of
dynamic range with 24 bits versus 16 bits looks great on paper, but it does
not help with problems that small.

I think you misunderstand me. In recording, you have two opposing goals: (1)
to record peaks at as high a level possible without over-modulating
(allowable in analog recording, with occasional, momentary, excursions to
+3dB being of no consequence but anathema in digital recordings where trying
to use bits that don't exist results in nasty noise.) and (2) while
simultaneously trying to keep the low-level info in the recording out of the
mud and to do so without gain riding or using analog audio compression to
restrict the dynamic range.

without the resultant recording ending up down in the mud where
distortion and
quantization noise increase as the signal toggles fewer and fewer bits.

Quantization noise is a straw man with modern digital equipment because its
hard to do Sigma-Delta converters without introducing dither or something
like it.

Quantization noise might not be a problem with dithering, but rising
distortion certainly is a problem. Low level signals are much better served
by 24-bit than by 16. It might not matter with pop music, but it certainly
does with classical. If you don't believe me, try recording a clavichord as
I recently did.

[Audio Empire]

On Tue, 30 Aug 2011 03:47:58 -0700, Edmund wrote
(in article ):

On Mon, 29 Aug 2011 23:15:38 +0000, Audio Empire wrote:

On Mon, 29 Aug 2011 04:56:46 -0700, Edmund wrote (in article
):
=20
On Mon, 29 Aug 2011 03:10:45 +0000, Robert Peirce wrote:
=20
I run Pure Music on my Mac. Presently, I use Airfoil to send the
signa=3D
l
over ethernet to my AppleTV. The AppleTV has an optical output to my
DAC. Pure Music and my DAC both support 96/24, but the Apple TV only
does 44.1/16 (or, maybe, 48/16 - hard to find specs). =3D20
I have been trying to find a substitute for the AppleTV, but so far
all I have got is the Squeezebox Touch. I say "but" because it has
its own software that resides on the Mac and I have not been able to
find out i=3D
f
it can receive input from Pure Music or not. Does anybody know? =3D2=
0
The web site suggests the Squeezebox can read any file on the
computer, which is great, except Pure Music already does that and
allows many useful manipulations. For example, it will accept up to
384/32 and downsample it to 96/24.
=20
I wonder what happens to the anti alias filter in that case.
=20
Edmund
=20
=20
You would certainly have to move the filter up in frequency in order to
use that extra bandwidth,=20

I don't think one get extra bandwidth when you DOWNSAMPLE 384/32 to 96/2=
4,
and the filter should be corrected to 40 kHz or so.

No one said that you did. In fact no one mentioned downsampling in
conjunction with this question at all.

otherwise the filter would simply treat the
signal like any other digital audio stream and start to roll-off the
frequency response above 22 KHz. What the use of high sampling rates
does is to move any quantization noise further out of the passband as
the sampling rate increases. Whether this is of any practical
consideration is debatable. Double-blind tests seem to show that this i=
s
no real consequence, but some will hotly debate the point.

And we can debate for a long time since there are no recordings recorded
with much higher frequencies then 22kHz.

I don't think that's true at all. Any recording mastered at 48 KHz, 88.2 KHz,
96 Khz, 176.4 KHz, or 192 KHz (not to mention 384 Khz or DSD) certainly have
info on them above 22 KHz. The frequency response plot that came with my
Avantone CK-40 stereo microphone shows significant (albeit attenuated) output
to slightly above 30 KHz and my mixer is flat to 50 KHz. I know that it's
there on my DSD masters and on the 24/96 copies that I run-off for my
clients.

I for one am still waiting for such real high res recordings.
( don't know about SACD since I cannot make such files visible.... yet )

Of course, you won't find that kind of frequency response on analog mastered
recordings or on any Red Book CD, for that matter. That should be obvious.

[Robert Peirce]

I am coming to the conclusion that a simple optical line from my MacBook
Pro to my DAC might be the best way to go. The problem is it would
probably be about 25'. Does anybody have any experience with optical
lines of that length?

I tend to think of optical as being purer than analog over long
distances, but I have no idea if that is correct or what problems could
occur. An analog line might actually be better.

In other words, I could move my DAC near my MBP and run a pair of analog
lines from the DAC to the stereo. This would likely be much bulkier and
more expensive, but that is how I am driving my power amps from my
pre-amp and that works fine. Of course, the signal level is much higher.

[Robert Peirce]

In article ,
Audio Empire wrote:

What I have noticed is less fatigue. This is totally subjective and
probably can't be measured. It is the difference between wanting to
listen to music all day and feeling forced to turn it off after a couple
of hours. Some people may not even notice it.

But do you know for sure that's due to extended supersonic frequency
response?

Nope. I don't know what causes it. I just know that I notice it, and
it is very subjective.

[Edmund[_2_]]

On Wed, 31 Aug 2011 00:21:13 +0000, Audio Empire wrote:

On Tue, 30 Aug 2011 03:47:58 -0700, Edmund wrote (in article
):

On Mon, 29 Aug 2011 23:15:38 +0000, Audio Empire wrote:

On Mon, 29 Aug 2011 04:56:46 -0700, Edmund wrote (in article
):
=20
On Mon, 29 Aug 2011 03:10:45 +0000, Robert Peirce wrote: =20
I run Pure Music on my Mac. Presently, I use Airfoil to send the
signa=3D
l
over ethernet to my AppleTV. The AppleTV has an optical output to
my DAC. Pure Music and my DAC both support 96/24, but the Apple TV
only does 44.1/16 (or, maybe, 48/16 - hard to find specs). =3D20 I
have been trying to find a substitute for the AppleTV, but so far
all I have got is the Squeezebox Touch. I say "but" because it has
its own software that resides on the Mac and I have not been able to
find out i=3D
f
it can receive input from Pure Music or not. Does anybody know?
=3D2=
0
The web site suggests the Squeezebox can read any file on the
computer, which is great, except Pure Music already does that and
allows many useful manipulations. For example, it will accept up to
384/32 and downsample it to 96/24.
=20
I wonder what happens to the anti alias filter in that case. =20
Edmund
=20
=20
You would certainly have to move the filter up in frequency in order
to use that extra bandwidth,=20

I don't think one get extra bandwidth when you DOWNSAMPLE 384/32 to
96/2= 4,
and the filter should be corrected to 40 kHz or so.

No one said that you did. In fact no one mentioned downsampling in
conjunction with this question at all.

otherwise the filter would simply treat the signal like any other
digital audio stream and start to roll-off the frequency response
above 22 KHz. What the use of high sampling rates does is to move any
quantization noise further out of the passband as the sampling rate
increases. Whether this is of any practical consideration is
debatable. Double-blind tests seem to show that this i=
s
no real consequence, but some will hotly debate the point.

And we can debate for a long time since there are no recordings
recorded with much higher frequencies then 22kHz.

I don't think that's true at all. Any recording mastered at 48 KHz, 88.2
KHz, 96 Khz, 176.4 KHz, or 192 KHz (not to mention 384 Khz or DSD)
certainly have info on them above 22 KHz. The frequency response plot
that came with my Avantone CK-40 stereo microphone shows significant
(albeit attenuated) output to slightly above 30 KHz and my mixer is flat
to 50 KHz. I know that it's there on my DSD masters and on the 24/96
copies that I run-off for my clients.

I haven't found a single piece of music with much higher frequencies then
22k and cannot find any information about any recording studio which advertise
to do so or even have equipment to do that. ( mics )
Of course it is possible to record high frequencies with 176 kHz sample rate
but I don't know what anti alias filters they use or what microphones.


I for one am still waiting for such real high res recordings. ( don't
know about SACD since I cannot make such files visible.... yet )

Of course, you won't find that kind of frequency response on analog
mastered recordings or on any Red Book CD, for that matter. That should
be obvious.

So we can forget HDtracks, where I find mostly ( only ?) old analog recordings.

Edmund

[Arny Krueger[_4_]]

"Edmund" wrote in message
...
On Tue, 30 Aug 2011 12:17:36 +0000, Arny Krueger wrote:

"Edmund" wrote in message
... On Mon, 29 Aug 2011 23:15:38
+0000, Audio Empire wrote:

And we can debate for a long time since there are no recordings
recorded with much higher frequencies then 22kHz. I for one am still
waiting for such real high res recordings. ( don't know about SACD
since I cannot make such files visible.... yet )

Then you are waiting for the Second Coming and the New Heaven and the
New Earth! ;-)

Bt that I mean that the (current) laws of physics are running strongly
against you.

Creating high frequencies at high amplitudes takes more energy.

Not so much and even so, I don't care.

Not so much? All other things being equal, energy is proportional to
frequency. You say you don't care but you actuall do or at least should.

For thousands of years people have been designing and building musical
instruments to be as audible and pleasing as possible. Evolution did the
same with our voices. This means that they very intentionally *avoid
wasting energy* with frequencies that are barely heard or not heard at
all.

Good luck trying to get the musicans of the world to switch over to
musical instruments that make SACDs sound different than CDs! ;-)

I read " there is life above 20kHz " ( or something ) and there are
quite a few instruments that produce sounds above 20k.

Simply producing any sound at all should be of no interest to you. What you
should matter to all of us is whether we can hear those sounds. To hear a
sound it has to be above the threshold of audibility and not masked by
other, stronger sounds. The flaw in the article you cite is that it ignores
actual audibility. Think of that article as being a journalist for the
Detrot Free Press making a big fuss out of the fact that there is a huge
pile of pure gold only a few hundred miles from Detroit. It's called Ft.
Knox and its presence does not enrich Detroiters any more than anybody else
in this country. In its way, its existence is mostly irrelevant to us all.

The existance of microscopic sounds at high frequencies is a truism, and
generally meaningless to musical enjoyment.

Then every change from silence requires an infinite bandwidth to make it
perfect.

Several flaws there. First off, nothing in the real world is perfect.
Secondly, what we seek is reproduction that is audibly indistinguishable
from the origionial soun, not some sound that all conforms to some imaginary
criteria that some audiophile or journalist makes up.

Just shaking your keyring produces frequencies above 20k but -agreed- that
isn't
a sound that is recognizable as music is but it does show that such high
frequencies are easily produced.

Not news to reasonably knowlegable people. No concern to more knowlegable
people.

So as long noone is recording these high frequencies in real music it
remains
pointless to discus whether or not it is audible.


The problem is that these frequencies have been recorded, so *someone* has
recorded them. Once recorded, a vast number of independent experimenters
have found that their presence or absence makes no noticable difference.

[Arny Krueger[_4_]]

"Edmund" wrote in message
...

I haven't found a single piece of music with much higher frequencies then
22k

I posted them on the web for many years. I took the recordings down for lack
of interest.

and cannot find any information about any recording studio which advertise
to do so or even have equipment to do that. ( mics )

There are a few mics that are reasonably flat up to 50 Khz. It is hard to
make a mic that is useful for recording with flat response above that.

The world's recordists have not rushed to spend the extra money for mics
with response above 15-20 Khz, presumably because in their experience they
serve no purpose.

I borrowed a pair of mics with response up to 50 Khz and had them at my
disposal for more than a year. I made a few recordings with them, compared
those recordings to themselves with everything above various lower
frequencies removed. I found that it is generally safe to brick wall filter
off everything above about 16 Khz. The difference that makes is not reliably
audible by a number of people, even when using speakers with far more high
frequency extension.

Of course it is possible to record high frequencies with 176 kHz sample
rate
but I don't know what anti alias filters they use or what microphones.

The anti alias filters generally come with the converters and they have as
much bandpass as possible - about 96 KHz in this case.

I know of no microphones that are practical for use in general recording
practice that have response above 50 KHz.

I for one am still waiting for such real high res recordings. ( don't
know about SACD since I cannot make such files visible.... yet )

Of course, you won't find that kind of frequency response on analog
mastered recordings or on any Red Book CD, for that matter. That should
be obvious.

So we can forget HDtracks, where I find mostly ( only ?) old analog
recordings.

The interesting thing is that there are all sorts of glowing reviews for
many of those recordings, some from prestigious reviewers. As long as they
thought the high frequency response was extended, they heard better sound.
Is your skeptic's bone itchnig yet? ;-)

BTW I've reviewed this matter with John Atkinson and he remains silent with
neither apology nor explanation.

[Arny Krueger[_4_]]

"Robert Peirce" wrote in message
...
I am coming to the conclusion that a simple optical line from my MacBook
Pro to my DAC might be the best way to go. The problem is it would
probably be about 25'. Does anybody have any experience with optical
lines of that length?

Long ago I bought a fairly typical pice of Toslink cable that was 30' long.
It seemed to work the same as shorter cables.

[Arny Krueger[_4_]]

"Audio Empire" wrote in message
...

I think you misunderstand me. In recording, you have two opposing goals:
(1)
to record peaks at as high a level possible without over-modulating
(allowable in analog recording, with occasional, momentary, excursions to
+3dB being of no consequence but anathema in digital recordings where
trying
to use bits that don't exist results in nasty noise.) and (2) while
simultaneously trying to keep the low-level info in the recording out of
the
mud and to do so without gain riding or using analog audio compression to
restrict the dynamic range.

Right, but no way is that difficult to do with the usual run of professional
gear running at 16/44.

Quantization noise is a straw man with modern digital equipment because
its
hard to do Sigma-Delta converters without introducing dither or something
like it.

Quantization noise might not be a problem with dithering, but rising
distortion certainly is a problem.

No it isn't. You may have been misled by plots showing THD+N. The rise is
due to the same noise floor appearing to contribute more as the signal level
went down.

The relevant spec is "dynamic range" which is measured with a -60 dB sine
wave. Generally the result of the measurement is dominated by noise, and if
you get at the actual spurious products due to nonlinear distortion, they
are equal or lower what you see with a -10 dB sine wave.

Please compa

http://home.comcast.net/~arnyk/pcavt...p-24192-60.gif

to:

http://home.comcast.net/~arnyk/pcavt...p-24192-12.gif

One is made with a -60 dB 1 KHz sine wave and the other is made with a -12
dB sine wave.

In both cases we see very similar spurious responses for the clearly
identifiable second and third harmonics (the only clearly identifiable
harmonics present) at about -128 dB down. The spurious response around 40
KHz is due to a switchmode power supply that was near by in a display.

These harmonics are so low as to be well below audibility by any known
generally-agreed upon criteria, and there is no rise in their amplitude even
though the signal level has been drastically reduced.

[Audio Empire]

On Wed, 31 Aug 2011 06:04:57 -0700, Edmund wrote
(in article ):

On Wed, 31 Aug 2011 00:21:13 +0000, Audio Empire wrote:


[quoted text deleted -- deb]

I haven't found a single piece of music with much higher frequencies
then 22k and cannot find any information about any recording studio
which advertise to do so or even have equipment to do that. ( mics )
Of course it is possible to record high frequencies with 176 kHz
sample rate but I don't know what anti alias filters they use or
what microphones.

That's because NOBODY CAN HEAR IT. Therefore it's irrelevant.

I for one am still waiting for such real high res recordings. ( don't
know about SACD since I cannot make such files visible.... yet )

Of course, you won't find that kind of frequency response on analog
mastered recordings or on any Red Book CD, for that matter. That
should be obvious.

So we can forget HDtracks, where I find mostly ( only ?) old analog
recordings.

That's correct. The 88.2 or 96 KHz sampling rate of any of these
formats is irrelevant. Even if higher frequencies than 20 KHz were to
make it to tape, self erasure and the migrating of magnetic domains
over time would have long since "dispersed" them. Most pro recorders
in the old analog days were maintained to be flat to 15 KHz. Above
that it is almost impossible to maintain head alignment and decent
head contact with the tape. Digitally, speaking, the difference in
actual perceived sound between 44.1 KHz and the higher sampling rates
is inaudible. Pure and simple. The 24-bit part might make a
difference, but I'm not even really convinced of that except for the
extra headroom it gives (by allowing a lower, overall recording level)
the recording engineer - especially in live recording situations.

[Audio Empire]

On Wed, 31 Aug 2011 07:02:39 -0700, Arny Krueger wrote
(in article ):

"Robert Peirce" wrote in message
...
I am coming to the conclusion that a simple optical line from my MacBook
Pro to my DAC might be the best way to go. The problem is it would
probably be about 25'. Does anybody have any experience with optical
lines of that length?

Long ago I bought a fairly typical pice of Toslink cable that was 30' long.
It seemed to work the same as shorter cables.



It does, and it it should. No surprises there.

[Edmund[_2_]]

On Wed, 31 Aug 2011 14:01:58 +0000, Arny Krueger wrote:

"Edmund" wrote in message
... On Tue, 30 Aug 2011 12:17:36
+0000, Arny Krueger wrote:

"Edmund" wrote in message
... On Mon, 29 Aug 2011 23:15:38
+0000, Audio Empire wrote:

And we can debate for a long time since there are no recordings
recorded with much higher frequencies then 22kHz. I for one am still
waiting for such real high res recordings. ( don't know about SACD
since I cannot make such files visible.... yet )

Then you are waiting for the Second Coming and the New Heaven and the
New Earth! ;-)

Bt that I mean that the (current) laws of physics are running strongly
against you.

Creating high frequencies at high amplitudes takes more energy.

Not so much and even so, I don't care.

Not so much? All other things being equal, energy is proportional to
frequency. You say you don't care but you actuall do or at least
should.

Well I admit I am a bit rusty here, but are you saying the ultrasonic sound
of a bat requires more energy to produce then the 7 Hz sound of an elephant?
Looking at instruments too I see the same phenomenon, low frequencies require
more air to be moved and much bigger instruments en more power to drive these
instruments. In loudspeakers too, the bass is bigger and need far more energy
then a tweeter.
Anyway it is not a problem to deliver the energy to drive a tweeter for the
very high frequencies.



For thousands of years people have been designing and building musical
instruments to be as audible and pleasing as possible. Evolution did
the same with our voices. This means that they very intentionally
*avoid wasting energy* with frequencies that are barely heard or not
heard at all.

Good luck trying to get the musicans of the world to switch over to
musical instruments that make SACDs sound different than CDs! ;-)

I read " there is life above 20kHz " ( or something ) and there are
quite a few instruments that produce sounds above 20k.

Simply producing any sound at all should be of no interest to you. What
you should matter to all of us is whether we can hear those sounds. To
hear a sound it has to be above the threshold of audibility and not
masked by other, stronger sounds. The flaw in the article you cite is
that it ignores actual audibility. Think of that article as being a
journalist for the Detrot Free Press making a big fuss out of the fact
that there is a huge pile of pure gold only a few hundred miles from
Detroit. It's called Ft. Knox and its presence does not enrich
Detroiters any more than anybody else in this country. In its way, its
existence is mostly irrelevant to us all.

The existance of microscopic sounds at high frequencies is a truism, and
generally meaningless to musical enjoyment.

Then every change from silence requires an infinite bandwidth to make
it perfect.

Several flaws there. First off, nothing in the real world is perfect.
Secondly, what we seek is reproduction that is audibly
indistinguishable from the origionial soun, not some sound that all
conforms to some imaginary criteria that some audiophile or journalist
makes up.

Just shaking your keyring produces frequencies above 20k but -agreed-
that isn't
a sound that is recognizable as music is but it does show that such
high frequencies are easily produced.

Not news to reasonably knowlegable people. No concern to more
knowlegable people.

So as long noone is recording these high frequencies in real music it
remains
pointless to discus whether or not it is audible.


The problem is that these frequencies have been recorded, so *someone*
has recorded them. Once recorded, a vast number of independent
experimenters have found that their presence or absence makes no
noticable difference.

I read about it and also that one younger man ( boy) scored a ten out
of ten and thus he was able to tell the difference.
Then people tell a lot of stories about high end and I like to hear
it for myself.

Edmund

[Arny Krueger[_4_]]

"Edmund" wrote in message
...
On Wed, 31 Aug 2011 14:01:58 +0000, Arny Krueger wrote:

"Edmund" wrote in message
... On Tue, 30 Aug 2011 12:17:36
+0000, Arny Krueger wrote:


Creating high frequencies at high amplitudes takes more energy.

Not so much and even so, I don't care.

Not so much? All other things being equal, energy is proportional to
frequency. You say you don't care but you actuall do or at least
should.

Well I admit I am a bit rusty here, but are you saying the ultrasonic
sound
of a bat requires more energy to produce then the 7 Hz sound of an
elephant?

This response seems to be very unclear about the concept of "all other
things being equal".

A little basic physics - sound has properties of both intensity and
frequency.

Is the intensity of an elephant bellowing the same as that of a tiny bat
ranging inside a cave? The frequencies are obviously different, but what
else is different.

Of course not!

Now let's see if you can put these pieces together?

[Dick Pierce[_2_]]

Edmund wrote:
On Wed, 31 Aug 2011 14:01:58 +0000, Arny Krueger wrote:
Not so much? All other things being equal, energy is proportional to
frequency.

Uh, no. If, "by all other things being equal," you mean
that for equal sound pressure level, your claim that
energy is proportional is provably wrong.

Sound pressure level is a measure of the amount
of acoustic power per unit area. Specifically,
0 dB SPL is defined as the equivalent of 10^-12,
and sound pressure level is defined as:

SPL = 20 log10 (Px/Pref)

where Pref = 10^-12 W

This same defining equation can be found in any number
of sources, such as Beranek, Blackenstock, Kinsler and
Frey, and many, many others. Not a single one of them
shows a frequency term.

Now, energy is power integrated over time. In the
simplest case, a speaker producing 1 acoustic watt
for 10 seconds produces 10 watt-seconds of energy
REGARDLESS of whether it's radiating 10 Hz, 100 Hz
or 100,000 Hz.

Now, it MAY, under some very narrowly constrained
circumstances, REQUIRE more input power to produce a
given acoustic power at some frequency than another,
but that's a matter of conversion efficiency, which
has no such instrinsic property of "energy is
proportional to frequency," as claimed.

Well I admit I am a bit rusty here, but are you
saying the ultrasonic sound of a bat requires more
energy to produce then the 7 Hz sound of an elephant?

That does appear to what he is saying and "all other
things being equal," which they simply can't be*, the
statement is wrong.

*All other things simply cannot be equal: for example,
the same surface radiating different frequencies has
different radiation patterns, different radiation
impedance, and so on. Two sources at different
frequencies with the same radiation impedance are
likely to have different radiating ares, moving
masses and the like.

Looking at instruments too I see the same phenomenon, low
frequencies require more air to be moved and much bigger
instruments en more power to drive these instruments.

No, they do not: they move a larger vlume of air, but,
for the same sound pressure level, they move it at a
substantially lower speed, indeed, the linear velocity
goes as the reciprocal of frequency. The net result is
that the volume ve,ocity of the source is constant with
frequency for a flat frequency response.

In loudspeakers too, the bass is bigger and need far more
energy then a tweeter.

Completely false. The amount of output acoustic power
compared to the input power is simnply a measure of
the efficiency of the system, and, for a flat frequency
response, that efficiency is, by definition, constant
over the pass band of the system. If your assertion was
correct, then the acoustic power radiated by a speaker
would would have a intrinsic direct dependence on
frequency for a constant input power, and this is
simply not the case. Conversely, if Mr. Krueger's
assertion were correct, the acoustic power radiated
by a speaker would have an intrinsic reciprocal
dependence on frequency, and this, as well, is not the
case.

The conditions required for a speaker to produce a
given sound pressure level is a specific volume
velocity. This is, essentially, the rate at which
a given volume of air can be moved. That means,
at low frequencies, the excursion of a diaphragm is
large, but the velocity is low, while at high
frequencies, the velocity is high, but the excursion
is low. It is the PRODUCT of these that determines
the sound pressure level, a measurement of power.
And that power, integrated over a given time interval,
is the energy radiated by the speaker (or whatever
happens to be producing the sound).

There simply is no intrinsic property that states
either that high frequencies requires more energy
or low frequencies require more energy. At least
not over a substantially broader bandwidth and at
much higher power than what we are talking about
for musci production. One can argue that the
basic gas laws go non-linear at extremely high
frequencies (orders of magnitude above the highest
imaginable musical sound, captured or otherwise)
or at sound levels where shock waves are forming.

Anyway it is not a problem to deliver the energy to drive
a tweeter for the very high frequencies.

No more than it is a problem to deliver the energy to
drive a woofer at very low frequencies. What's missing
is a definition of "very high" and "very low". Within
their pass band, a tweeter and a woofer of equal electro-
acoustic efficiency requires the same input power to
produce the same acoustic power REGARDLESS of the
frequency. Both drivers at "very high" and "very low"
frequencies, require substantially more power, if such
frequencies are outside the pass band of the drivers.


--
+--------------------------------+
+ Dick Pierce |
+ Professional Audio Development |
+--------------------------------+

[Robert Peirce]

In article ,
Audio Empire wrote:

On Wed, 31 Aug 2011 07:02:39 -0700, Arny Krueger wrote
(in article ):

"Robert Peirce" wrote in message
...
I am coming to the conclusion that a simple optical line from my MacBook
Pro to my DAC might be the best way to go. The problem is it would
probably be about 25'. Does anybody have any experience with optical
lines of that length?

Long ago I bought a fairly typical pice of Toslink cable that was 30' long.
It seemed to work the same as shorter cables.



It does, and it it should. No surprises there.

I am pleased that it does, but my question was based on wondering why it
should.

I think light through fiber is more efficient than electricity through
wire (although I might be wrong), but it seems to me both must suffer if
pushed over unreasonable distances. I believe even long fiber optic
lines use repeaters or amplifiers to boost the signal.

I am also not sure about how light behaves in long lines perhaps going
around corners. I can imagine (again perhaps wrongly) that there might
be some loss of energy or disruption of signal in going around corners.

What I got from this response was that 30' is not an unreasonable length
in the real world, which was all that really mattered.

[Audio Empire]

On Thu, 1 Sep 2011 16:27:56 -0700, Robert Peirce wrote
(in article ):

In article ,
Audio Empire wrote:

On Wed, 31 Aug 2011 07:02:39 -0700, Arny Krueger wrote
(in article ):

"Robert Peirce" wrote in message
...
I am coming to the conclusion that a simple optical line from my MacBook
Pro to my DAC might be the best way to go. The problem is it would
probably be about 25'. Does anybody have any experience with optical
lines of that length?

Long ago I bought a fairly typical pice of Toslink cable that was 30'
long.
It seemed to work the same as shorter cables.



It does, and it it should. No surprises there.

I am pleased that it does, but my question was based on wondering why it
should.

I think light through fiber is more efficient than electricity through
wire (although I might be wrong), but it seems to me both must suffer if
pushed over unreasonable distances. I believe even long fiber optic
lines use repeaters or amplifiers to boost the signal.

True. At some length, fiber requires in-line repeaters, but 30 feet is NOT
that length. Fiber. like everthing else, is lossy. Light leaks out the sides
of the cable, there are internal reflections that can cancel or otherwise
compromise signal integrity, Mainly optical's strengths are much wider
bandwidth (an optical signal can carry much more information than wire
without nearly as much loss because light is a much higher frequency than an
electrical signal). Optical is also electrically isolatory and can isolate
one electrical structure from another while still transmitting data between
them. In the case of digital audio, optical has easy duty. The amount of
data is not so great (7 or 8 channels of audio at most) and the frequency is
fairly low. For the most part, there is really little difference in the
quality of a coaxial digital connection and an optical one.

I am also not sure about how light behaves in long lines perhaps going
around corners. I can imagine (again perhaps wrongly) that there might
be some loss of energy or disruption of signal in going around corners.

There is, but over short domestic runs, it's inconsequential.

What I got from this response was that 30' is not an unreasonable length
in the real world, which was all that really mattered.

Neither would 50 ft or even 100 ft.

[Arny Krueger[_4_]]

"Robert Peirce" wrote in message
...

I think light through fiber is more efficient than electricity through
wire (although I might be wrong), but it seems to me both must suffer if
pushed over unreasonable distances. I believe even long fiber optic
lines use repeaters or amplifiers to boost the signal.

Depends on the fiber. The plastic fiber used in Toslink has relatively large
loss, much more loss than a reasonble sized copper wire.

The glass fibers used for long distance communication have far lower losses.

I am also not sure about how light behaves in long lines perhaps going
around corners.

Sharp corners are avoided, but gently bent curves can be followed by the
light beam

I can imagine (again perhaps wrongly) that there might
be some loss of energy or disruption of signal in going around corners.

There are modwerate losses associated with those gentle bends.

What I got from this response was that 30' is not an unreasonable length
in the real world, which was all that really mattered.

30 feet toslink just looks. There are special low loss cables that can break
the 30' rule.

[Dick Pierce[_2_]]

Audio Empire wrote:
At some length, fiber requires in-line repeaters, but 30 feet is NOT
that length. Fiber. like everthing else, is lossy. Light leaks out the sides
of the cable, there are internal reflections that can cancel or otherwise
compromise signal integrity,

Actually, in media like Toslink, it is the internal
reflections that make it work by reducing the leakage
out the sides.

Mainly optical's strengths are much wider bandwidth (an optical
signal can carry much more information than wire without nearly
as much loss because light is a much higher frequency than an
electrical signal).

Well, elements of this are true, but, as a whole, it's quite
incorrect. The fact that the transmission medium uses a carrier
with a very high frequency (light) is completely irrelevant to
the system's bandwidth. The fact is the bandwidth limit is NOT
set by the freuqency of the light: is is, in this case, set by
the bandwidth of the transducers at each end of the link:
the transmitter and receiver. Their bandwidth is FAR less than
what the intervening cable might or moght not support. We could
double the frequency of the light: go into the near UV as
opposed to the near IR, and the bandwidth of the system will
not change on iota unless the transducers' bandwidth changes,
and that's almost totally independent on the frequency of the
carrier.

Indeed, one of the big problems with Toslink is NOT the optical
cable, but the crappy electro-optics at either end: unsymmetrical
thrsehold hysteresis, rise and fall times and more in the detectors
can lead to all sorts of problems, effectively reducing the bandwidth
to far less than what might otherwise be obtained.

When I had done a fairly extensive amount of testing on digital
interfaces during the development of a digital audio workstation,
I explored a large number of interfaces, both electrical and optical.
Without a single exception that stands out enough to be noteworthy,
every Toslink interface had substantially less bandwidth than every
electrical interface I tested, and not by a small amount. The lone
exception to this was not one optical interface that was speedy,
but one electrical one (a S/P-DIF output form a DAT recorder) that
was abyssmal.

Optical is also electrically isolatory and can isolate
one electrical structure from another while still
transmitting data between them.

This much is very true, and can have some real benefits.

In the case of digital audio, optical has easy duty. The amount of
data is not so great (7 or 8 channels of audio at most) and the frequency is
fairly low.

And Toslink transducers are just barely able to keep up.

For the most part, there is really little difference in the
quality of a coaxial digital connection and an optical one.

In practice, this is largely true, assuming competent
implementation, which is not a givenm in the high-end
audio realm.

I am also not sure about how light behaves in long lines perhaps going
around corners. I can imagine (again perhaps wrongly) that there might
be some loss of energy or disruption of signal in going around corners.

A "corner" that you could put in anb optical fiber without
breaking it is thousands of times larger than the wavelengths
you are dealing with. At those wavelengths, it's difficult to
actually tell the difference between a straight run and on that
has a radius of a fraction of an inch (assuming, again, you
don't break it).

What I got from this response was that 30' is not an unreasonable length
in the real world, which was all that really mattered.

Neither would 50 ft or even 100 ft.

Well, again, that assumes comtentent implementation.
Toslink is NOT a lang-haul medium, indeed, Toshiba, the
company who invented and promoted it, specifies its maximum
length a whopping 6 feet.

In actual practice, the available length limits depend
heavily upon the actual transmitter and receiver modules used.
What's interesting is that while Toshiba gives extensive
specifications, inluding maximum length, for almost all
of their optical transmitter/receiver hardware, they are
quite silent on the matter for their digital audio optical
products.

It's also curious to note that those modules they designate
for digital audio purposes, despite running at a wavelength
650nm (equivalent to a free-air frequency of 461 terahertz),
the specificied data rate is only 15 Mb/s: we were spec'ing
pulse transformers and transmitter/receiver pairs for our
project that had data rates well over an order of magnitude
higher, and this was routine. This sinmply demonstrates that,
in practice, Toslink DOES NOT have anything like a wider
bandwidth than electrical transmission.

--
+--------------------------------+
+ Dick Pierce |
+ Professional Audio Development |
+--------------------------------+

[Ken[_8_]]

On 31 Aug 2011 23:12:30 GMT, Audio Empire
wrote:

I haven't found a single piece of music with much higher frequencies
then 22k and cannot find any information about any recording studio
which advertise to do so or even have equipment to do that. ( mics )
Of course it is possible to record high frequencies with 176 kHz
sample rate but I don't know what anti alias filters they use or
what microphones.

That's because NOBODY CAN HEAR IT. Therefore it's irrelevant.

Some of us can hear high frequencies.
Now at 57 years old I can hear up to 18.5 kHz.
When I was 35 years old I could hear up to 24 kHz.
When I was younger, I don't know, but I could hear
remote controls and burglar alarm detectors then.

[Audio Empire]

On Fri, 2 Sep 2011 06:46:53 -0700, Dick Pierce wrote
(in article ):

Audio Empire wrote:
At some length, fiber requires in-line repeaters, but 30 feet is NOT
that length. Fiber. like everthing else, is lossy. Light leaks out the
sides
of the cable, there are internal reflections that can cancel or otherwise
compromise signal integrity,

Actually, in media like Toslink, it is the internal
reflections that make it work by reducing the leakage
out the sides.

Those are not the kinds of reflections I was talking about.

Mainly optical's strengths are much wider bandwidth (an optical
signal can carry much more information than wire without nearly
as much loss because light is a much higher frequency than an
electrical signal).

Well, elements of this are true, but, as a whole, it's quite
incorrect. The fact that the transmission medium uses a carrier
with a very high frequency (light) is completely irrelevant to
the system's bandwidth. The fact is the bandwidth limit is NOT
set by the freuqency of the light: is is, in this case, set by
the bandwidth of the transducers at each end of the link:
the transmitter and receiver. Their bandwidth is FAR less than
what the intervening cable might or moght not support. We could
double the frequency of the light: go into the near UV as
opposed to the near IR, and the bandwidth of the system will
not change on iota unless the transducers' bandwidth changes,
and that's almost totally independent on the frequency of the
carrier.

I said that in reality, the carrier frequency advantage was inconsequential
in this case.

Indeed, one of the big problems with Toslink is NOT the optical
cable, but the crappy electro-optics at either end: unsymmetrical
thrsehold hysteresis, rise and fall times and more in the detectors
can lead to all sorts of problems, effectively reducing the bandwidth
to far less than what might otherwise be obtained.

Very true, but again, at audio sampling frequencies, probably not of any real
consequence.

When I had done a fairly extensive amount of testing on digital
interfaces during the development of a digital audio workstation,
I explored a large number of interfaces, both electrical and optical.
Without a single exception that stands out enough to be noteworthy,
every Toslink interface had substantially less bandwidth than every
electrical interface I tested, and not by a small amount. The lone
exception to this was not one optical interface that was speedy,
but one electrical one (a S/P-DIF output form a DAT recorder) that
was abyssmal.

Optical is also electrically isolatory and can isolate
one electrical structure from another while still
transmitting data between them.

This much is very true, and can have some real benefits.

In the case of digital audio, optical has easy duty. The amount of
data is not so great (7 or 8 channels of audio at most) and the frequency
is
fairly low.

And Toslink transducers are just barely able to keep up.

For the most part, there is really little difference in the
quality of a coaxial digital connection and an optical one.

In practice, this is largely true, assuming competent
implementation, which is not a givenm in the high-end
audio realm.

I am also not sure about how light behaves in long lines perhaps going
around corners. I can imagine (again perhaps wrongly) that there might
be some loss of energy or disruption of signal in going around corners.

A "corner" that you could put in anb optical fiber without
breaking it is thousands of times larger than the wavelengths
you are dealing with. At those wavelengths, it's difficult to
actually tell the difference between a straight run and on that
has a radius of a fraction of an inch (assuming, again, you
don't break it).

What I got from this response was that 30' is not an unreasonable length
in the real world, which was all that really mattered.

Neither would 50 ft or even 100 ft.

Well, again, that assumes comtentent implementation.
Toslink is NOT a lang-haul medium, indeed, Toshiba, the
company who invented and promoted it, specifies its maximum
length a whopping 6 feet.

In actual practice, the available length limits depend
heavily upon the actual transmitter and receiver modules used.
What's interesting is that while Toshiba gives extensive
specifications, inluding maximum length, for almost all
of their optical transmitter/receiver hardware, they are
quite silent on the matter for their digital audio optical
products.

It's also curious to note that those modules they designate
for digital audio purposes, despite running at a wavelength
650nm (equivalent to a free-air frequency of 461 terahertz),
the specificied data rate is only 15 Mb/s: we were spec'ing
pulse transformers and transmitter/receiver pairs for our
project that had data rates well over an order of magnitude
higher, and this was routine. This sinmply demonstrates that,
in practice, Toslink DOES NOT have anything like a wider
bandwidth than electrical transmission.

That's probably correct. My comments were about optical in general, not
TOSLINK specifically. I have no experience with TOSLINK other than as a user,
but in the Polaris Trident project, we used glass fiber interconnects to do
virtually all of the rocket guidance and internal navigation communications.
We were able to replace literally over half a ton of mil-spec cabling with
several light, thin strands of glass optical cabling carrying hundreds of
different digital signals consisting of everything from audio frequencies
for the in-engine vectoring fins to near microwave for radar and guidance
control signals, all at the same time.

[Dick Pierce[_2_]]

Audio Empire wrote:
On Fri, 2 Sep 2011 06:46:53 -0700, Dick Pierce wrote
Audio Empire wrote:

At some length, fiber requires in-line repeaters, but 30 feet is NOT
that length. Fiber. like everthing else, is lossy. Light leaks out the
sides
of the cable, there are internal reflections that can cancel or otherwise
compromise signal integrity,

Actually, in media like Toslink, it is the internal
reflections that make it work by reducing the leakage
out the sides.

Those are not the kinds of reflections I was talking about.

What kind were you talking about? End-to-end?

This introduces the same non-problem that reflections
in a cable present. Remember, stuff's happening at
the speed of light in the medium, and that's no picking
of daisies.

Let's do a little gedanke: say the ends are really badly
terminated, such that you have a 50% reflection from
the cable end. And assume you need to drop the level
60 dB to drop below the level you're generating receiving
problems. And let's further assume a 10 ft cable.

Each round trip looses 12 dB: 6 dB for each 50% loss in
reflected intensity. So you have 5 round trips, which,
assuming an RI of 1.62, and thus a propogation velocity
of 607,000,000 odd ft/sec, it takes 98 nanoseconds for
the reflected energy to drop below the 60 dB threshold.

In fact, assuming a badly terminated fiber with pure
air gap, worst-case Fresnel reflection is on the order
of 15%:

R = ((RIf -Ria) / (Rif + Ria))^2

So, that represents 16 dB per end or a 32 dB round-trip
loss, which means 2 round trips and we're below our
arbitrary 60 dB threshold, under 40 nanosecods.

And, in fact, the threshold is substantially more
generous than that.

If not end-to-end, what kind of reflections were you
talking about?

Mainly optical's strengths are much wider bandwidth (an optical

signal can carry much more information than wire without nearly
as much loss because light is a much higher frequency than an
electrical signal).

Well, elements of this are true, but, as a whole, it's quite
incorrect. The fact that the transmission medium uses a carrier
with a very high frequency (light) is completely irrelevant to
the system's bandwidth. The fact is the bandwidth limit is NOT
set by the freuqency of the light: is is, in this case, set by
the bandwidth of the transducers at each end of the link:
the transmitter and receiver. Their bandwidth is FAR less than
what the intervening cable might or moght not support. We could
double the frequency of the light: go into the near UV as
opposed to the near IR, and the bandwidth of the system will
not change on iota unless the transducers' bandwidth changes,
and that's almost totally independent on the frequency of the
carrier.

I said that in reality, the carrier frequency advantage
was inconsequential in this case.

So it's a true in some case but orrelevant here sort of
thing?

Indeed, one of the big problems with Toslink is NOT the optical
cable, but the crappy electro-optics at either end: unsymmetrical
thrsehold hysteresis, rise and fall times and more in the detectors
can lead to all sorts of problems, effectively reducing the bandwidth
to far less than what might otherwise be obtained.

Very true, but again, at audio sampling frequencies, probably not of any real
consequence.

Actually, for those DACs that depend heavily upon timing
accuracy i the individual bits to recover the sample
clock (an all-around BAD design which, unfortunately,
a number of high-end companies gleefully implemented),
it has a VERY real and VERY significant consequence,
and is one of the few verifiable causes of very large
amounts of sample jitter.

To increase the accuracy of the sattement, I might
be inclined to have said, "In equipment designed
competently to properly manage sample output clocking,
at audio sampling frequencies, probably not of any real
consequence," but this makes a lot of assumptions about
high end audio designs which are not supportable in a
practical sort of way.

It's also curious to note that those modules they designate
for digital audio purposes, despite running at a wavelength
650nm (equivalent to a free-air frequency of 461 terahertz),
the specificied data rate is only 15 Mb/s: we were spec'ing
pulse transformers and transmitter/receiver pairs for our
project that had data rates well over an order of magnitude
higher, and this was routine. This sinmply demonstrates that,
in practice, Toslink DOES NOT have anything like a wider
bandwidth than electrical transmission.

That's probably correct. My comments were about optical in
general, not TOSLINK specifically.

But the properties of things like single-mode dark fiber
are so totally removed from those of Toslink, even though
they operate under the same physical principles, whatever
is true of single-mode fiber is completely irrelevant
for Toslink. Single mode 9 micron fiber can go 60
kilometers witgout a repeater.

I have no experience
with TOSLINK other than as a user,
but in the Polaris Trident project, we used glass fiber interconnects to do
virtually all of the rocket guidance and internal navigation communications.

THat's the difference between 9 uM glass and the soda straws
used in Toslink. TOslink is cheap but at the cost of low
bandwidth and short distance. Single-mode glass can run 40 Gb/s
over kilometers, audio toslink is limited to 15 Mbs for runs
of a few meters at best.

We were able to replace literally over half a ton of mil-spec cabling with
several light, thin strands of glass optical cabling carrying hundreds of
different digital signals consisting of everything from audio frequencies
for the in-engine vectoring fins to near microwave for radar and guidance
control signals, all at the same time.

No doubt. And had you done it with Toslink, I doubt it
would even boot up.


--
+--------------------------------+
+ Dick Pierce |
+ Professional Audio Development |
+--------------------------------+

[Audio Empire]

On Fri, 2 Sep 2011 12:05:38 -0700, Ken wrote
(in article ):

On 31 Aug 2011 23:12:30 GMT, Audio Empire
wrote:

I haven't found a single piece of music with much higher frequencies
then 22k and cannot find any information about any recording studio
which advertise to do so or even have equipment to do that. ( mics )
Of course it is possible to record high frequencies with 176 kHz
sample rate but I don't know what anti alias filters they use or
what microphones.

That's because NOBODY CAN HEAR IT. Therefore it's irrelevant.

Some of us can hear high frequencies.
Now at 57 years old I can hear up to 18.5 kHz.
When I was 35 years old I could hear up to 24 kHz.
When I was younger, I don't know, but I could hear
remote controls and burglar alarm detectors then.



24 KHz is possible with DAT (48KHz sampling rate) and there are exceptions to
every rule. At 66, for instance, I can still hear 15KHz, but most people
(especially the "rock" generations) can't hear much above 9-10 KHz when they
get older.

[Robert Peirce]

Well, after much listening, I have concluded that a 25-30' length of
Toslink may be about the same as a 25-30' length of wire, at least as
far as I will be able to tell.

This raises another interesting point. How will it compare to what I
have now? The source and the DAC are constants. So everything depends
on the connection.

At present, I use ethernet over powerline because my WiFi signal isn't
strong enough to the area where my stereo is located. I use Airfoil to
transmit the signal over ethernet. It uses Apple's Core Audio. An
ethernet cable goes from my MacBook Pro to a powerline adaptor. Another
powerline adaptor feeds my AppleTV and the AppleTV feeds the DAC over a
short TosLink cable. This is a dedicated circuit. It is only used for
audio.

The proposed setup is to run a 25-30' Toslink cable directly from the
MacBook Pro's optical output to the DAC. The optical output also uses
Apple's Core Audio but possibly not in the same way. Airfoil would not
be needed.

I suspect there are audible differences between the two setups. The
important point is will I be able to hear them and if so, which will be
better? Generally, I have found the less hardware in the circuit the
less chance of audible conflicts. Nevertheless, regardless of any
theoretical difference, I suspect I won't hear any real difference.
However, I don't know.

[Dick Pierce[_2_]]

Audio Empire wrote:

24 KHz is possible with DAT (48KHz sampling rate)

No, it is not. First, the Nyquist criteria requires
the bandwidth to be limited to less than half the
sample rate, not less than or equal to half. Second,
the transition bandwidth is not infinitesimal.

practically speaking, most DAT recorders had bandwidth
similar to CD players.

--
+--------------------------------+
+ Dick Pierce |
+ Professional Audio Development |
+--------------------------------+

[Ed Seedhouse[_2_]]

On Sep 1, 3:48=A0am, Edmund wrote:

Well I admit I am a bit rusty here, but are you saying the ultrasonic sou=
nd
of a bat requires more energy to produce then the 7 Hz sound of an elepha=
nt?

Only if they are of equal amplitude, which they aren't very likely to
be.

Looking at instruments too I see the same phenomenon, low frequencies req=
uire
more air to be moved and much bigger instruments en more power to drive t=
hese
instruments. In loudspeakers too, the bass is bigger and need far more en=
ergy
then a tweeter.

But the tweeters produce much less amplitude. Try to make a tweeter
of the same efficiency of a woofer play a high frequency at the same
amplitude and you require more energy to do it.

It's proven mathematically and by measurement, but It's also
intuitively straightforward. Amplitude is how "big" the swing from
positive to negative is. If the swings are the same size the
amplitudes are the same. But higher frequencies have to swing back
and forth much faster and produce more waves than the lower ones in
the same time period, and that requires more energy. Swing a stick
slowly and then increase the speed while keeping the swings at the
same length. The faster you swing the harder you will work.

It doesn't matter if it's sound or light or electricity, the higher
the frequency the more energy an equal amplitude of the wave carries
more energy. Period. Dictated by the laws of physics. Nothing you
can do about it.

Anyway it is not a problem to deliver the energy to drive a tweeter for t=
he
very high frequencies.

Well yes it is if you want to keep the amplitude constant. What
you'll actually do in that case is burn out the tweeter rather quickly
if you have enough power available.

I read about it and also that one younger man ( boy) scored a ten out
of ten and thus he was able to tell the difference.
Then people tell a lot of stories about high end and I like to hear
it for myself.

Unless you give a reference to a properly blinded or double blinded
study where this was shown, you are merely relating an anecdote.
Anecdotes ain't evidence.

Edmund

[Arny Krueger[_4_]]

"Dick Pierce" wrote in message
...
Audio Empire wrote:

24 KHz is possible with DAT (48KHz sampling rate)

No, it is not. First, the Nyquist criteria requires
the bandwidth to be limited to less than half the
sample rate, not less than or equal to half. Second,
the transition bandwidth is not infinitesimal.

practically speaking, most DAT recorders had bandwidth
similar to CD players.

Except that CD players have a maximum sampling frequency of 44.1 KHz which
is set by the Red Book standard.

Let's not split hairs. while it is true that a 48 KHz digital signal can't
reproduce 24.000 KHz, it can reproduce 23 Khz, 23.5 kHZ, and in many cases
23.9 KHz. It depends, but many DACs don't have low pass filters that even
come close to totally obliterating the signal at 0.95 Nyquist, or at even
at the Nyquist frequency.

For example, the well-known Analog devices AD 1853 DAC

http://www.analog.com/static/importe...ets/AD1853.pdf

spec sheet says that its stop band is 26.23-358.28 KHz. 26 KHz is Nyquist +
2 KHz! According to Figure 14, its digital filter is only about 10 dB
down at 50 KHz with a 96 KHz clock. This is equivalent to being 10 dB down
at 24 KHz with a 48 KHz clock.

[Harry Lavo]

"Robert Peirce" wrote in message
...
In article ,
Audio Empire wrote:

What I have noticed is less fatigue. This is totally subjective and
probably can't be measured. It is the difference between wanting to
listen to music all day and feeling forced to turn it off after a
couple
of hours. Some people may not even notice it.

But do you know for sure that's due to extended supersonic frequency
response?

Nope. I don't know what causes it. I just know that I notice it, and
it is very subjective.

Second that. Audible from the first day of SACD playback, and continues to
this day ten years later. CD's have become excellent, but I still can only
take a few hours of listening before becoming restless. SACDs can be
playing all day without this effect (and, BTW, so can analogue tapes).