Computer-DAC
 

In article , David Pitt
wrote:
Rob wrote:


Anyway while googling and failing to find any actual facts, just look at
this :-

http://www.computeraudiophile.com/co...ct-digital-out

"It is well known that any lossless use will not sound as good as a flat
PCM file like AIFF or WAV." and "Just in the same way that FLAC does not
sound as good as WAV on a PC.", by one J Gordon Rankin.

Surely that is just crap, at least I hope it is.

It is possible that some hardware/software combinations are struggling to
'keep up' and buffer out the data at a decently uniform rate. But that
should not be the case. My impression is that in 'computer land' there is
often a lack of real understanding, or even basic check measurements.

Indeed, an article I read in a Linux mag a few months ago by a 'sound
expert' made me wonder what kind of results you'd get if I'd understood
correctly what he seemed to be telling people to do! But then, what he
wrote did seem to be par for a music biz where level compression and
clipping seems to be regarded as 'normal' in more ways that one! :-)

Slainte,

Jim

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

Computer-DAC
 

Jim Lesurf wrote:
In article , Rob
wrote:
On 09/02/2010 09:29, Jim Lesurf wrote:
In a.com, Rob


Though what happens if you replay audio snatched from a DVD? If you
are met with silence, you may have to occasionally reset output to
48kHz, and remember to set it back afterwards - or just leave it at
48kHz and hope that the internal circuits do a decent job of 44.1/48
resampling.

Yes, (but) I think iTunes sorts all that sort of thing out.
The DACMagic LEDs will tell you if the rate being output matches the
source material or not.


And . . . it depends. iTunes tends to sample up/down to 44.1,
regardless. I say 'tend' - if I use another application, Plex, for music
replay it changes system settings to 48/16 as if by magic - just FYI as
an example of the carnival going on behind the scenes. I can't find any
Mac software that will output the file in the recorded format.

IIUC the above shows that the Mac hardware is quite capable of providing
both 44.1 and 48. If so, your main 'challenge' (as managers say)

Yeah, the last one that said that to me didn't realise how close to death he was. :)




--
Bill Coombes

Computer-DAC
 

On 09/02/2010 18:43, Keith G wrote:

"Rob" wrote


No, that's fine, and FYI it's called 'iTunes'.


Hi Rob - my Mac Mini died suddenly after only two weeks!

Gone and refunded* now, I don't think I'll go that route again - there
are some very pleasant aspect to the Mac system, but from what little I
saw and remember of it, 'iTunes' was the least appealing thing on it! I
know you could resist without too much of a struggle, but it seemed to
want to take me 'shopping' every time I looked at it!


Lightweight! At least you gave it a go. I can see why Macs are
infuriating/not worth the bother for many - they just hit the spot for
me in terms of productivity/engineering/reliability/ease-pleasure of use.

Computer-DAC
 

On 10/02/2010 09:22, Jim Lesurf wrote:
In a.com, Rob
wrote:
On 09/02/2010 09:29, Jim Lesurf wrote:
In a.com, Rob



Though what happens if you replay audio snatched from a DVD? If you
are met with silence, you may have to occasionally reset output to
48kHz, and remember to set it back afterwards - or just leave it at
48kHz and hope that the internal circuits do a decent job of 44.1/48
resampling.


Yes, (but) I think iTunes sorts all that sort of thing out.

The DACMagic LEDs will tell you if the rate being output matches the
source material or not.


And . . . it depends. iTunes tends to sample up/down to 44.1,
regardless. I say 'tend' - if I use another application, Plex, for music
replay it changes system settings to 48/16 as if by magic - just FYI as
an example of the carnival going on behind the scenes. I can't find any
Mac software that will output the file in the recorded format.

IIUC the above shows that the Mac hardware is quite capable of providing
both 44.1 and 48. If so, your main 'challenge' (as managers say) is to find
software that simply passes out the data at the correct rates in every
case. Afraid I can't say anything about that with the Mac as I know little
about Mac software.


Yes, the Mac will 'deal' with 32 - 96Khz. Thing is, there's an OS 'gate'
that controls input and output. It's called the Audio Midi Setup
application. Now, other apps can 'call' that app, and vary the sample
rate as per the source file. None that I can see do in a consistent way.

If I get a chance I'll ask a colleague who is a Mac user to see if he
knows. However my impression is that many computer users have zero
awareness of this issue and blindly assume that if they can hear sound it
must be "working OK". They then blame any nasty noises on the source
material.


That's kind, thanks. I've put a similar query on a Mac newsgroup, and
it's not something of interest it seems. I simply want it to 'work
right'. Which it does, I think, if I set the settings to match the
source file.


However you could try asking Cambridge Audio.


There's a fair bit on t'internet, and it seems to make a pretty good job
of things using the correct settings, but apparently the USB interface
isn't up to much - sounded fine to me, have to say.

http://www.stereophile.com/digitalpr...r/index2.html#

Not yet looked at the above link. However I do get Stereophile on
subscription and remember reading the DACmagic review (and a later addition
to it IIRC). My impression is that operation with the USB will depend on
how evenly the computer sends the data. If so the real problem is initially
down to that.

FWIW When I started using my DACMagic i fed it from the computer via USB
and then ran an optical output from the DACMagic to my Meridian 563 DAC for
the actual conversion. In effect just using the DACMagic as a USB-SPDIF
convertor, and then having the Meridian reclocking the data to suppress any
jitter.

However I also experimented with using the DACMagic without the Meridian
and eventually decided that also sounded fine. Decided it sounded
indistinguishable to me. So I now just use the DACMagic and am moving the
Meridian on to other duties. Another advantage of this is that the DACMagic
can playback digital recordings at rates above 48k from my new HD P2.


'Only' 16 bit via USB IIUC, but life goes on :-)

Computer-DAC
 

On 10/02/2010 09:00, David Pitt wrote:
wrote:

On 08/02/2010 19:17, Arny Krueger wrote:
wrote in message
m
I've connected a Cambridge Dacmagic to a Mac Mini via the optical
connection, and I'm not sure which settings to select in the Apple
audio setup, for stereo playback through iTunes. It offers a variety
of 2ch settings with a variety of formats (44.1 - 96) and bits (16 -
24). They all work, but I'd like the theoretical best setting, if
there is one.

This depends on the media you are playing. Media plays best in the
format it was recorded at.

If the files you play are MP3 and MP4 and the like, 44.1 / 16 is what
you want.


Yep, thanks. As you might gather from another reply, it's a little
difficult following which software does what.

It is a tricky subject.

What would be best is "bit perfect', what comes out is what went in
unchanged in any way, this is something more than something simply coming
out at the same sampling rate.

Foobar2000, with a plugin, did just that here on Vista, foobar2000 takes
sole control of the output card bypassing the digital mixer which is the bit
that "messes" things up. (Being Vista it could not play MP3s without
glitches so despite being bit perfect it was in fact useless.) The rationale
from Microsoft was along the lines that their software manipulation of the
data was so very clever that bit rate conversion was just fine.

I now use a Mac mini, which may be no more inherently 'bit perfect' than
Windows, there is still a mixer in there to add in computer bleeps etc.,
obviously those are all turned off but the mixer will still be there but
with one input mute.

To duck the bitrate issue I use the analogue output from the Mac mini which
sounds very good to me. The thought does occur that the sound is going
through that digital mixer anyway so what I am listening to is not 'bit
perfect' after all.


It could be new clothes syndrome, but I reckon this DAC sounds a fair
bit better than the analogue out.

I did say it is tricky.

Anyway while googling and failing to find any actual facts, just look at
this :-

http://www.computeraudiophile.com/co...ct-digital-out

"It is well known that any lossless use will not sound as good as a flat PCM
file like AIFF or WAV." and "Just in the same way that FLAC does not sound
as good as WAV on a PC.", by one J Gordon Rankin.

Surely that is just crap, at least I hope it is.


If your summary is a reflection (I couldn't wade through it all!) but
surely it's nonsense? Lossless compression and decompression has nothing
to do with sound - it's simply processing to/from the original form (wav
etc), in much the same way as zipped files are compressed/decompressed.
IIUC.

On bit perfect: do you/anyone know how it's measured/tested?

Also, I think I read that if the computer software volume control works,
the digital output is not bit perfect.

Rob

Computer-DAC
 

In article , Rob
wrote:
On 10/02/2010 09:22, Jim Lesurf wrote:

And . . . it depends. iTunes tends to sample up/down to 44.1,
regardless. I say 'tend' - if I use another application, Plex, for
music replay it changes system settings to 48/16 as if by magic -
just FYI as an example of the carnival going on behind the scenes. I
can't find any Mac software that will output the file in the recorded
format.

IIUC the above shows that the Mac hardware is quite capable of
providing both 44.1 and 48. If so, your main 'challenge' (as managers
say) is to find software that simply passes out the data at the
correct rates in every case. Afraid I can't say anything about that
with the Mac as I know little about Mac software.


Yes, the Mac will 'deal' with 32 - 96Khz. Thing is, there's an OS 'gate'
that controls input and output. It's called the Audio Midi Setup
application.

Curious that it has 'midi' in the name as that is something quite different
to dealing with standard LPCM data.

Now, other apps can 'call' that app, and vary the sample
rate as per the source file. None that I can see do in a consistent way.

If I get a chance I'll ask a colleague who is a Mac user to see if he
knows. However my impression is that many computer users have zero
awareness of this issue and blindly assume that if they can hear sound
it must be "working OK". They then blame any nasty noises on the
source material.


That's kind, thanks. I've put a similar query on a Mac newsgroup, and
it's not something of interest it seems.

That seems consistent with the impression I've had of the level of interest
and awareness of these matters amongst most Windows and Linux and RO
users. The default assumption people seem to make is that if they can 'hear
the music' then the sound system of the computer is 'working'. They then
blame any problems on the source material being poor, or some other reason.
No thought that you need to get the rate correct for the source and obtain
bit perfect output for LPCM. But then I guess most people will be using
mp3, etc, a lot of the time without having any idea how the compressed
formats differ from LPCM.



FWIW When I started using my DACMagic i fed it from the computer via
USB and then ran an optical output from the DACMagic to my Meridian
563 DAC for the actual conversion. In effect just using the DACMagic
as a USB-SPDIF convertor, and then having the Meridian reclocking the
data to suppress any jitter.

However I also experimented with using the DACMagic without the
Meridian and eventually decided that also sounded fine. Decided it
sounded indistinguishable to me. So I now just use the DACMagic and am
moving the Meridian on to other duties. Another advantage of this is
that the DACMagic can playback digital recordings at rates above 48k
from my new HD P2.


'Only' 16 bit via USB IIUC, but life goes on :-)

I've recently been experimenting with rates above 48k. But as yet I've
simply not bothered with 24 bit at all. Although there may be some
situations where I can see a use for it. Overall, though, I am currently
happy with 44k/16 *if* the source material is of good quality and carefully
reproduced.

Slainte,

Jim

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

Computer-DAC
 

In article , Rob
wrote:

On bit perfect: do you/anyone know how it's measured/tested?

I've been though this in recent months. e.g. for

http://www.audiomisc.co.uk/Linux/Sou...Computing.html

http://www.audiomisc.co.uk/Linux/Sou...stenAgain.html

The key is that you need to have a recorder that can accept the spdif (or
USB) output and record the sample stream. Then compare that with the LPCM
source you are playing on a sample-by-sample basis.

For 44k/16 I've used a Pioneer Audio CD recorder. But am changing to the
Tascam HD P2. One reason being that it can record other sample rates and 24
bit as well as 16.

I then use my own RO and ROX/Linux apps to examine the results. I will be
making copies of the software available in due course so anyone who wishes
can use them. But the key point is to have the ability to record what
emerges from the computer.


Also, I think I read that if the computer software volume control works,
the digital output is not bit perfect.

Yes. It is necessary to set any such control to '0dB' or '100 percent'.
....and also necessary to check that this *does* mean 'gain of unity
applied'. I've encountered cases where 'volume' controls do *not* leave the
data values unchanged for such a setting! I'm afraid that the situation
with computer sound seems a right guddle to me. It seems to be aimed at
assuming ignorant users will have no interest in audio quality. :-/

BTW as you can see from the above webpages you need to set '11' on the BBC
iPlayer to stop the iPlayer software from altering the output level. i.e it
uses the 'Spinal Tap' scale. 8-}

Slainte,

Jim

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

Computer-DAC
 

Jim Lesurf wrote:

In article , Rob
wrote:
On 10/02/2010 09:22, Jim Lesurf wrote:

[snip]

Yes, the Mac will 'deal' with 32 - 96Khz. Thing is, there's an OS 'gate'
that controls input and output. It's called the Audio Midi Setup
application.

Curious that it has 'midi' in the name as that is something quite
different to dealing with standard LPCM data.

The name is just a MacFoible. The setup application is called "Audio MIDI
Setup", it should be "Audio & MIDI setup". It is where the DacMagic makes
its appearance to configure its feed.

In a fit of enthusiasm, and optimism, I have rushed out and bought a
DacMagic which is now connected via USB to the Mac mini. The Mac Mini is not
bit perfect in this mode, the DacMagic is fed at the bitrate set by the
configuration and not by the bitrate of the file being played, with iTunes.
Now does that matter, time will tell.


I did find this thread attempting to discuss bit perfection which wended its
way to "transparent resampling".

http://www.hydrogenaudio.org/forums/...hp/t60692.html

Some quotes :-

From mixminus1, "While this doesn't prove that OS X and XP are capable of
providing "bit-perfect" output to the sound output device, it seems like a
distinct possibility as I would be surprised if both OSes were performing
transparent resampling.". He actually does not have a clue, it must be bit
perfect because he cannot hear any "artefacts".

One Arnold B Krueger responded with, "Why would you be surprised if a modern
OS did transparent resampling?".

That is point, does it matter. I assume that monkeying around with the
digits is a "bad thing" but can resampling of the sort done in these
computer mixers be done sufficiently well.


The DacMagic has sorted another oddity here. I had doubts about the
Squeezebox being bit perfect, that must be bit perfect because it does not
have a mixer in it in the way that a computer does but that did not stop my
Quad kit from flagging everything as 44.1kHz. The DacMagic gets it right.

--
David Pitt

MessengerPro on iMac, Snow Leopard

Computer-DAC
 

On 11/02/2010 12:14, David Pitt wrote:
Jim wrote:

In a.com, Rob
wrote:
On 10/02/2010 09:22, Jim Lesurf wrote:

[snip]

Yes, the Mac will 'deal' with 32 - 96Khz. Thing is, there's an OS 'gate'
that controls input and output. It's called the Audio Midi Setup
application.

Curious that it has 'midi' in the name as that is something quite
different to dealing with standard LPCM data.

The name is just a MacFoible. The setup application is called "Audio MIDI
Setup", it should be "Audio& MIDI setup". It is where the DacMagic makes
its appearance to configure its feed.


Yep.

In a fit of enthusiasm, and optimism, I have rushed out and bought a
DacMagic which is now connected via USB to the Mac mini. The Mac Mini is not
bit perfect in this mode, the DacMagic is fed at the bitrate set by the
configuration and not by the bitrate of the file being played, with iTunes.
Now does that matter, time will tell.


Well, if the file is the same as the 'iTunes forced mode' (44.1/16),
then I would hope that all's well. Except to say the Dacmagic
*upsamples* for what the bumpf maintains are Very Good Reasons. Just
have to let that one go ;-)

I did find this thread attempting to discuss bit perfection which wended its
way to "transparent resampling".

http://www.hydrogenaudio.org/forums/...hp/t60692.html

Some quotes :-

From mixminus1, "While this doesn't prove that OS X and XP are capable of
providing "bit-perfect" output to the sound output device, it seems like a
distinct possibility as I would be surprised if both OSes were performing
transparent resampling.". He actually does not have a clue, it must be bit
perfect because he cannot hear any "artefacts".

One Arnold B Krueger responded with, "Why would you be surprised if a modern
OS did transparent resampling?".

That is point, does it matter. I assume that monkeying around with the
digits is a "bad thing" but can resampling of the sort done in these
computer mixers be done sufficiently well.


I think I'll just use USB after all. Sounds very good to me. And
something about velvet glove/sow's ear springs to mind.


The DacMagic has sorted another oddity here. I had doubts about the
Squeezebox being bit perfect, that must be bit perfect because it does not
have a mixer in it in the way that a computer does but that did not stop my
Quad kit from flagging everything as 44.1kHz. The DacMagic gets it right.


Yes, no mucking about using it in that application.

Rob

Computer-DAC
 

On 11/02/2010 09:16, Jim Lesurf wrote:
In a.com, Rob
wrote:

On bit perfect: do you/anyone know how it's measured/tested?

I've been though this in recent months. e.g. for

http://www.audiomisc.co.uk/Linux/Sou...Computing.html

http://www.audiomisc.co.uk/Linux/Sou...stenAgain.html


You did, you did, saw that, but it was more a 'which bit of wire do I
connect to what and how do I interpret the data' type thing enquiry.

The key is that you need to have a recorder that can accept the spdif (or
USB) output and record the sample stream. Then compare that with the LPCM
source you are playing on a sample-by-sample basis.

For 44k/16 I've used a Pioneer Audio CD recorder. But am changing to the
Tascam HD P2. One reason being that it can record other sample rates and 24
bit as well as 16.

I then use my own RO and ROX/Linux apps to examine the results. I will be
making copies of the software available in due course so anyone who wishes
can use them. But the key point is to have the ability to record what
emerges from the computer.


Excellent, I'll watch out for that.


Also, I think I read that if the computer software volume control works,
the digital output is not bit perfect.

Yes. It is necessary to set any such control to '0dB' or '100 percent'.
...and also necessary to check that this *does* mean 'gain of unity
applied'. I've encountered cases where 'volume' controls do *not* leave the
data values unchanged for such a setting! I'm afraid that the situation
with computer sound seems a right guddle to me. It seems to be aimed at
assuming ignorant users will have no interest in audio quality. :-/


A guddle indeed. I actually meant that if the software volume *works* -
is active - it's a rough and ready and mainly accurate indication that
the digital stream is being manipulated. The only time this has happened
on the two 'layers' of volume control for the Mac (system and app) was
optical feed to an AV amplifier - so I'd guess and say hgetting the
'pure' signal out is not impossible. But I'd be guessing.

BTW as you can see from the above webpages you need to set '11' on the BBC
iPlayer to stop the iPlayer software from altering the output level. i.e it
uses the 'Spinal Tap' scale. 8-}


Jocular lot the BBC.