[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 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. .

[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_]]

"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! ;-)

[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 03:47:58 -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.

[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

[VasiliY_Honda]

êòî íàõ ñêàæåòü ÷òî ÿïîíñêàÿ õîíäÿ ýòà ãàíî? àààà áëåàòü? ....

[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.

[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.

[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.

[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.

[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.

[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.

[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.

[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.

[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?

[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.

[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).

[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.

[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.

[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.

[VasiliY_Honda]

êòî íàõ ñêàæåòü ÷òî ÿïîíñêàÿ õîíäÿ ýòà ãàíî? àààà áëåàòü? ....

[ligel]

Originally Posted by 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.

I have been using Pure music for a couple of months now. They and Amarra have free trials, so you can check out any differences yourself. I don't know of others that play like this through iTunes.

I can make a lot of comments. they may come slowly while I am at work.

1. Definitely both make a positive difference. Astounding really, but you don't realize how poor iTunes is until you try one of these programs. Amarra may be a little better sounding, but I didn't try that long because Pure music has a ton of features that Amarra doesn't, unless you perhaps get the $695 version. The "junior" and "mini" versions are so limited, you may as well buy Pure Music for $165 or so.

2. There are definitely bugs in the interface. you have lots of set up options and sometimes a small mistake will make something happen like not moving to the next track or something. I have multiple instances where there is a glitch in how the itunes interface works and I gets pulsating bursts of incomplete music. Always fixed with a reboot.

3. The sound improvement is definitely worth minor inconveniences like that above and the learning curve. There are many, many featurse and options and you can add modules or use the supplied ones to digitally modify, equalize or even cross overs. you can upsample, down sample and play native high sample rates. All good, if a little confusing. I am a fan of front end digital room treatment/equalization and this does that well. It has a limitation/oversight in saving equalization files, so you can lose data, but it sounds great and is very flexible. Audio hijack Pro has a better interface for this, but doesn't have the sound quality. It distorts a lot, this doesn't.

Overall, I highly recommend it! Amarra if you are a pro, but it is too expensive for me. Pure Music makes about the same difference as going from a lousy cheap DAC to an excellent high end DAC and you should do both.

For me, money well spent, but I hope there are user interface improvements, stability improvements and more options in the future.