Sign in with
Sign up | Sign in
Your question

Digital quality - is there a difference?

Last response: in Home Audio
Share
Anonymous
July 30, 2004 8:48:07 PM

Archived from groups: rec.audio.pro (More info?)

Advice please!

If you are recording a digital source onto a PC (e.g. the S/PDIF output from
a digital multitrack) does the sound card have any effect on the sound
quality?

Would something like the M-Audio Audiophile 2496 give better results than,
say, any of the Sound Blaster cards that have S/PDIF connections?

Or are they the same, as digital either works or it doesn't and there is no
D/A conversion taking place in the sound card?

Thanks
Anonymous
July 30, 2004 8:48:08 PM

Archived from groups: rec.audio.pro (More info?)

"Richard Brooks" <nobodyhere@deadspam.com> wrote in message
news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com


> If you are recording a digital source onto a PC (e.g. the S/PDIF
> output from a digital multitrack) does the sound card have any effect
> on the sound quality?

Depends on the sound card. Some resample their digital inputs. Some of those
do it badly.

> Would something like the M-Audio Audiophile 2496 give better results
> than, say, any of the Sound Blaster cards that have S/PDIF connections?

Let's put it this way - it's a well known thing that the SoundBlaster cards
resample, while none of the cards designed for audio production, including
most if not all of M-Audio's line, don't resample at all.

> Or are they the same, as digital either works or it doesn't and there
> is no D/A conversion taking place in the sound card?

Badly done, resampling done poorly can be more damaging than a mediocre
modern D/A or A/D.
Anonymous
July 30, 2004 8:48:08 PM

Archived from groups: rec.audio.pro (More info?)

In article <410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com> nobodyhere@deadspam.com writes:

> If you are recording a digital source onto a PC (e.g. the S/PDIF output from
> a digital multitrack) does the sound card have any effect on the sound
> quality?

In theory, no, as long as they work the same in theory.

> Would something like the M-Audio Audiophile 2496 give better results than,
> say, any of the Sound Blaster cards that have S/PDIF connections?

Do I remember correctly that SoundBlasters only work internally at 48
kHz, and that even a 44.1 kHz digital input gets sample-rate-converted
in the card, even though it writes a 44.1 kHz file to disk? Or is this
some abandonded form of technology abuse? Or simply brain damage
(mine)?


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Related resources
Anonymous
July 30, 2004 8:48:09 PM

Archived from groups: rec.audio.pro (More info?)

On Fri, 30 Jul 2004 13:01:17 -0400, "Arny Krueger" <arnyk@hotpop.com>
wrote:

>"Richard Brooks" <nobodyhere@deadspam.com> wrote in message
>news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com
>
>
>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>> output from a digital multitrack) does the sound card have any effect
>> on the sound quality?
>
>Depends on the sound card. Some resample their digital inputs. Some of those
>do it badly.
>

Whose cards do that? How do they mess it up? They decide that a 1
should be a 0? or a 16,000 should be a 16,001?

Or do they convert from digital to analog and back again before
writing? That seems like a long way around what should be a simple
circuit.

Perhaps the acid test would be to do the old phase inversion test on a
file from the source and the resampled S/PDIF transfer and see if any
bits got hosed or numbers re-sampled.

Except his source is a digital multitrack.




Kurt Riemann
Anonymous
July 30, 2004 8:48:09 PM

Archived from groups: rec.audio.pro (More info?)

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1091207122k@trad
> In article <410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com>
> nobodyhere@deadspam.com writes:
>
>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>> output from a digital multitrack) does the sound card have any
>> effect on the sound quality?
>
> In theory, no, as long as they work the same in theory.
>
>> Would something like the M-Audio Audiophile 2496 give better
>> results than, say, any of the Sound Blaster cards that have S/PDIF
>> connections?

> Do I remember correctly that SoundBlasters only work internally at 48
> kHz, and that even a 44.1 kHz digital input gets sample-rate-converted
> in the card, even though it writes a 44.1 kHz file to disk?

Yes.

> Or is this some abandonded form of technology abuse?

The latest Soundblasters run at sample rates up to 192 KHz, so obviously
they don't resample *everything* any more.

> Or simply brain damage (mine)?

No, no matter what Phil says...

;-)
Anonymous
July 30, 2004 8:48:10 PM

Archived from groups: rec.audio.pro (More info?)

<Kurt Riemann> wrote in message
news:1b6lg0l4i5igp208d0rpf78e5l9ts8bfmr@4ax.com
> On Fri, 30 Jul 2004 13:01:17 -0400, "Arny Krueger" <arnyk@hotpop.com>
> wrote:
>
>> "Richard Brooks" <nobodyhere@deadspam.com> wrote in message
>> news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com
>>
>>
>>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>>> output from a digital multitrack) does the sound card have any
>>> effect on the sound quality?
>>
>> Depends on the sound card. Some resample their digital inputs. Some
>> of those do it badly.

> Whose cards do that?

I don't know the total extent of the problem, but the SBLive! that you can
listen to recording by at this web page surely has the problem:

http://www.pcabx.com/product/ct4830/index.htm

> How do they mess it up?

Resampling is arithmetic, they made some systematic errors in arithmetic.

> They decide that a 1
> should be a 0? or a 16,000 should be a 16,001?

It was a little worse that that. They missed the boat by about a random 1
part in 10,000 which would like confusing 0 with 6 or 16,006 with 16,000.
Then they also had a systematic error that varied with fequency, but peaked
out at about 1 part in 6.

> Or do they convert from digital to analog and back again before
> writing?

No, they screwed up in the digital domain. The problem with many game cards
is that their core DSPs run at 48 KHz, regardless of what the data is like.
They even resample the data when its at 48 KHz so they don't have to
synchronize their DSP with the outside world.

>That seems like a long way around what should be a simple circuit.

Game cards have all kinds of 3D effects, etc built into them. The card is
built to color sound, and they never seem to be able to turn all the color
off.

> Perhaps the acid test would be to do the old phase inversion test on a
> file from the source and the resampled S/PDIF transfer and see if any
> bits got hosed or numbers re-sampled.

Been there done that, the result is that just about everything is hosed.

> Except his source is a digital multitrack.

You still want to be able to process it cleanly when you want to process it
cleanly, right?
Anonymous
July 30, 2004 8:55:23 PM

Archived from groups: rec.audio.pro (More info?)

Richard wrote
>If you are recording a digital source onto a PC (e.g. the S/PDIF output from
>a digital multitrack) does the sound card have any effect on the sound
>quality?
>

No, but the playback thru your computers D/A converters is a whole-nother
story.
Anonymous
July 30, 2004 9:00:20 PM

Archived from groups: rec.audio.pro (More info?)

On 30 Jul 2004 16:55:23 GMT, bruwhaha58097238@aol.com (Raymond) wrote:

>Richard wrote
>>If you are recording a digital source onto a PC (e.g. the S/PDIF output from
>>a digital multitrack) does the sound card have any effect on the sound
>>quality?
>>
>
>No, but the playback thru your computers D/A converters is a whole-nother
>story.

Also, you have to watch what's going on with syncing and sample rate
conversion. If you just have a single connection -from- a good digital out
source -to- the card, and the card is not doing SRC, you're fine.
Anonymous
July 31, 2004 7:17:18 AM

Archived from groups: rec.audio.pro (More info?)

On Fri, 30 Jul 2004 11:01:54 -0800, Kurt Riemann <> wrote:

>On Fri, 30 Jul 2004 13:01:17 -0400, "Arny Krueger" <arnyk@hotpop.com>
>wrote:
>
>>"Richard Brooks" <nobodyhere@deadspam.com> wrote in message
>>news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com
>>
>>
>>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>>> output from a digital multitrack) does the sound card have any effect
>>> on the sound quality?
>>
>>Depends on the sound card. Some resample their digital inputs. Some of those
>>do it badly.
>>
>
>Whose cards do that? How do they mess it up? They decide that a 1
>should be a 0? or a 16,000 should be a 16,001?

The biggest problem is that, during resampling, it maps the incoming signal to
the internal clock, which has high jitter. That means the signal's timing has
been given a jitter exactly out of phase with the card's jitter. A perfectly
jitter free signal being resampled by a jittery card will therefore be
jittery.
Anonymous
July 31, 2004 10:52:46 PM

Archived from groups: rec.audio.pro (More info?)

Richard Brooks wrote:
> Advice please!
>
> If you are recording a digital source onto a PC (e.g. the S/PDIF
> output from a digital multitrack) does the sound card have any effect
> on the sound quality?
>
> Would something like the M-Audio Audiophile 2496 give better results
> than, say, any of the Sound Blaster cards that have S/PDIF
> connections?

Depends on the driver. If there is no attenuation (or boost) going on the
the driver or associated mixer applet, there should be no effect whatsoever
on the data (unless the spdif chipset or implementation is flawed).
>
> Or are they the same, as digital either works or it doesn't and there
> is no D/A conversion taking place in the sound card?

Certainly no D/A taking place for an SPDIF output. But possibly some DSP.
Worst case being most (all ?) SoundBlasters where reputedly *everything*
gets resampled thru a 48K rate section !

geoff
Anonymous
July 31, 2004 10:54:25 PM

Archived from groups: rec.audio.pro (More info?)

Kurt Riemann wrote:
> On Fri, 30 Jul 2004 13:01:17 -0400, "Arny Krueger" <arnyk@hotpop.com>
> wrote:
>
>> "Richard Brooks" <nobodyhere@deadspam.com> wrote in message
>> news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com
>>
>>
>>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>>> output from a digital multitrack) does the sound card have any
>>> effect on the sound quality?
>>
>> Depends on the sound card. Some resample their digital inputs. Some
>> of those do it badly.
>>
>
> Whose cards do that? How do they mess it up? They decide that a 1
> should be a 0? or a 16,000 should be a 16,001?

Sample rate isn't bit depth.


> Or do they convert from digital to analog and back again before
> writing?

Outputting, you mean. No

geoff
Anonymous
July 31, 2004 10:55:45 PM

Archived from groups: rec.audio.pro (More info?)

Mike Rivers wrote:
>>
> Do I remember correctly that SoundBlasters only work internally at 48
> kHz, and that even a 44.1 kHz digital input gets sample-rate-converted
> in the card, even though it writes a 44.1 kHz file to disk? Or is this
> some abandonded form of technology abuse? Or simply brain damage
> (mine)?

Definitely brain-damage. Theirs, not yuors.

geoff
Anonymous
July 31, 2004 10:55:46 PM

Archived from groups: rec.audio.pro (More info?)

On Sat, 31 Jul 2004 18:55:45 +1200, "Geoff Wood"
<geoff@paf.co.nz-nospam> wrote:

>> Do I remember correctly that SoundBlasters only work internally at 48
>> kHz, and that even a 44.1 kHz digital input gets sample-rate-converted
>> in the card, even though it writes a 44.1 kHz file to disk? Or is this
>> some abandonded form of technology abuse? Or simply brain damage
>> (mine)?
>
>Definitely brain-damage. Theirs, not yuors.

A justification might be the SB cards' inclusion of an on-board
sample-player engine - the SoundFont system. This would require to
work at a set sample rate.

Quality cards are notable for their LACK of such on-board goodies.

CubaseFAQ www.laurencepayne.co.uk/CubaseFAQ.htm
"Possibly the world's least impressive web site": George Perfect
Anonymous
July 31, 2004 10:55:47 PM

Archived from groups: rec.audio.pro (More info?)

In article <e64ng09ir4um4qlmpbj7onjlmlae3jpqdu@4ax.com> l@laurenceDELETEpayne.freeserve.co.uk writes:

> A justification might be the SB cards' inclusion of an on-board
> sample-player engine - the SoundFont system. This would require to
> work at a set sample rate.

This is of some importance to game developers (who want their audio
production to sound as much alike on diverse systems as possible) but
it really has no place in a "pro" studio.

> Quality cards are notable for their LACK of such on-board goodies.

Yes, but keep your voice down lest you incite the "Why can't I buy a
Digital I/O Only card for less than one with analog I/O?" crowd.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 1, 2004 10:44:18 AM

Archived from groups: rec.audio.pro (More info?)

Laurence Payne wrote:
> A justification might be the SB cards' inclusion of an on-board
> sample-player engine - the SoundFont system. This would require to
> work at a set sample rate.

Call me crazy, but why couldn't the SoundFont thing work at its
required sample rate and then have the sample rate conversion
convert the SoundFont's output to the rate that the D/A is going?
It doesn't seem like this would be a lot more expensive to build
given that either way you include stuff that does sample rate
conversion...

- Logan
Anonymous
August 1, 2004 11:38:34 AM

Archived from groups: rec.audio.pro (More info?)

In article <6j0Pc.11906$Zm3.8751@fe2.texas.rr.com> lshaw-usenet@austin.rr.com writes:

> Call me crazy, but why couldn't the SoundFont thing work at its
> required sample rate and then have the sample rate conversion
> convert the SoundFont's output to the rate that the D/A is going?
> It doesn't seem like this would be a lot more expensive to build
> given that either way you include stuff that does sample rate
> conversion...

"It doesn't seem like this would be a lot more expensive . . " is what
people say when they don't work in the business. First off, it
probably really IS a lot more expensive, considering the change in
design concept. And even if it required only the addition of a few
five cent components, that makes enough of a cost increase in the
final product so that it gets nixed by Marketing. (This is the reason
why Mackie VLZ Pro mic preamps start to roll off the low frequencies
when you get near to full gain.)

If you're willing to spend more money to get better performance in the
areas that you really want, you need to buy a card designed for your
applications, and that costs more money if for no other reason than
that they don't sell that many of them. Wouldn't you rather use
Gigastudio than SoundFonts anyway? <g>

--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 1, 2004 11:59:28 AM

Archived from groups: rec.audio.pro (More info?)

"Laurence Payne" <l@laurenceDELETEpayne.freeserve.co.uk> wrote in
message news:e64ng09ir4um4qlmpbj7onjlmlae3jpqdu@4ax.com
> On Sat, 31 Jul 2004 18:55:45 +1200, "Geoff Wood"
> <geoff@paf.co.nz-nospam> wrote:
>
>>> Do I remember correctly that SoundBlasters only work internally at
>>> 48 kHz, and that even a 44.1 kHz digital input gets
>>> sample-rate-converted in the card, even though it writes a 44.1 kHz
>>> file to disk? Or is this some abandonded form of technology abuse?
>>> Or simply brain damage (mine)?
>>
>> Definitely brain-damage. Theirs, not yuors.
>
> A justification might be the SB cards' inclusion of an on-board
> sample-player engine - the SoundFont system. This would require to
> work at a set sample rate.
>
> Quality cards are notable for their LACK of such on-board goodies.

I think its the net complexity of the processing in the cards. There is a
lot of memory devoted to storing parameters for all these sound generation
and modification features. The designers don't want to develop and store a
seperate set of paramters for each sample rate, and they don't want to
dynamically calculate a new set for each sample rate, either.

Bottom line, these cards are designed for effects first and fidelity second.
At this time, a number of them have reasonably good sample rate conversion
algorithms, and so they don't sound all that board. I understand that the
the latest version of the card that started this all, the SBLive 5.1, has
updated sample rate algorithms, and really isn't too bad.

compare the new:

http://audio.rightmark.org/test/creative-live!5.1-1644.html

with the old:

http://mojepc.pl/documents/recenzje/0208_sound/live-liv...

Smoothing the response out and minimizing the variations where the ear is
most sensitive ( 1 Khz-6 KHz) seems to have made a bit of a difference.
However, no way does this compare to a proper card for audio production:

http://www.pcavtech.com/soundcards/delta-1010lt/1644.ht...

Note the extreme flatness where it really matters - where the ear is most
sensitive ( 1 Khz-6 KHz).
Anonymous
August 1, 2004 10:03:06 PM

Archived from groups: rec.audio.pro (More info?)

On Fri, 30 Jul 2004 16:24:39 -0400, "Arny Krueger" <arnyk@hotpop.com>
wrote:
ust about everything is hosed.
>
>> Except his source is a digital multitrack.
>
>You still want to be able to process it cleanly when you want to process it
>cleanly, right?
>

I meant that he would not be able to do the experiment without a clean
original file.




KR
Anonymous
August 1, 2004 10:15:47 PM

Archived from groups: rec.audio.pro (More info?)

On Sat, 31 Jul 2004 03:17:18 GMT, Steve Jorgensen
<nospam@nospam.nospam> wrote:

>On Fri, 30 Jul 2004 11:01:54 -0800, Kurt Riemann <> wrote:
>
>>On Fri, 30 Jul 2004 13:01:17 -0400, "Arny Krueger" <arnyk@hotpop.com>
>>wrote:
>>
>>>"Richard Brooks" <nobodyhere@deadspam.com> wrote in message
>>>news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com
>>>
>>>
>>>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>>>> output from a digital multitrack) does the sound card have any effect
>>>> on the sound quality?
>>>
>>>Depends on the sound card. Some resample their digital inputs. Some of those
>>>do it badly.
>>>
>>
>>Whose cards do that? How do they mess it up? They decide that a 1
>>should be a 0? or a 16,000 should be a 16,001?
>
>The biggest problem is that, during resampling, it maps the incoming signal to
>the internal clock, which has high jitter. That means the signal's timing has
>been given a jitter exactly out of phase with the card's jitter. A perfectly
>jitter free signal being resampled by a jittery card will therefore be
>jittery.

By jittery, do you mean that the errors are in the encoding and that
there is a displacement of data in the stream that is then written
incorrectly (IE later or earlier in the file by one sample) so data is
either truncated or duplicated in the actual file? Or does random data
come out? Like the crackle we all love so much.

That's how I'm reading this. I don't use cheap sound cards and always
clock to the source (with word if possible) so I'm unfamiliar with how
jitter really gets the file screwed up.

All I know is that by making something slave this way I've eliminated
clicks and pops.


I'll read the other posts and see what other illumination I get.



Kurt Riemann
Anonymous
August 2, 2004 12:04:46 AM

Archived from groups: rec.audio.pro (More info?)

On 1 Aug 2004 07:38:34 -0400, mrivers@d-and-d.com (Mike Rivers) wrote:

>(This is the reason
>why Mackie VLZ Pro mic preamps start to roll off the low frequencies
>when you get near to full gain.)

Could you elaborate on this? I understood the input trim control to
be merely an attenuator - the preamp stage was always operating at
"full gain".

CubaseFAQ www.laurencepayne.co.uk/CubaseFAQ.htm
"Possibly the world's least impressive web site": George Perfect
Anonymous
August 2, 2004 1:40:59 AM

Archived from groups: rec.audio.pro (More info?)

In article <rjfqg0p1nlc7fdotpbso7l6ke27fv67rmt@4ax.com> l@laurenceDELETEpayne.freeserve.co.uk writes:

> >why Mackie VLZ Pro mic preamps start to roll off the low frequencies
> >when you get near to full gain.)
>
> Could you elaborate on this? I understood the input trim control to
> be merely an attenuator - the preamp stage was always operating at
> "full gain".

The Trim control is a gain control. At maximum gain, the bypassing on
the gain stage is a little skimpy and the low end rolls off. This was
a decision based on cost, not engineering. The designer knew the
compromise that he was making - he could have done worse and saved
the same amount of money.

A "pad" (usually a switch) is an attenuator.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 2, 2004 2:07:25 AM

Archived from groups: rec.audio.pro (More info?)

Mike Rivers wrote:
> "It doesn't seem like this would be a lot more expensive . . " is what
> people say when they don't work in the business. First off, it
> probably really IS a lot more expensive, considering the change in
> design concept.

Well yes, given that they already went the other direction, which of
course they did. But I'm just saying I'm not convinced that having
that synth stuff on there necessarily forces them to design the card
so it does superfluous sample rate conversion. Which isn't to say
it isn't true -- just playing devil's advocate.

> And even if it required only the addition of a few
> five cent components, that makes enough of a cost increase in the
> final product so that it gets nixed by Marketing.

Indeed. I know someone who designs some of the circuits that go
into speakers you'd get if you bought a PC from one of the major
manufacturers. They sell in the millions of units, I think, so
even saving 10 cents per unit is probably more than his annual
salary. Translation: if they can hire a qualified EE for $100,000/yr
and that person can sit around and do nothing all year except figure
out how to eliminate a 10 cent part from each of two million-unit
products, they have more than paid for themselves.

> Wouldn't you rather use
> Gigastudio than SoundFonts anyway? <g>

I'd rather use all acoustic instruments and no digital synthesizers
except in rare cases, but I'm like that.

- Logan
Anonymous
August 2, 2004 2:07:26 AM

Archived from groups: rec.audio.pro (More info?)

Logan Shaw <lshaw-usenet@austin.rr.com> wrote:
>
>Indeed. I know someone who designs some of the circuits that go
>into speakers you'd get if you bought a PC from one of the major
>manufacturers. They sell in the millions of units, I think, so
>even saving 10 cents per unit is probably more than his annual
>salary. Translation: if they can hire a qualified EE for $100,000/yr
>and that person can sit around and do nothing all year except figure
>out how to eliminate a 10 cent part from each of two million-unit
>products, they have more than paid for themselves.

I did this for about two weeks, right out of college. I couldn't live
with myself when I went home at the end of the day.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
August 2, 2004 2:42:25 AM

Archived from groups: rec.audio.pro (More info?)

In article <znr1091401968k@trad>, Mike Rivers <mrivers@d-and-d.com> wrote:
>In article <rjfqg0p1nlc7fdotpbso7l6ke27fv67rmt@4ax.com> l@laurenceDELETEpayne.freeserve.co.uk writes:
>
>> >why Mackie VLZ Pro mic preamps start to roll off the low frequencies
>> >when you get near to full gain.)
>>
>> Could you elaborate on this? I understood the input trim control to
>> be merely an attenuator - the preamp stage was always operating at
>> "full gain".
>
>The Trim control is a gain control. At maximum gain, the bypassing on
>the gain stage is a little skimpy and the low end rolls off. This was
>a decision based on cost, not engineering. The designer knew the
>compromise that he was making - he could have done worse and saved
>the same amount of money.

Doing it this way basically gives you more "headroom" that you'd otherwise
be able to get out of the front end; as you reduce the gain of the front
end, you aren't adding noise in the process as you would with a resistive
pad in front. The other side of the coin is that the bandwidth and the
impulse response of the front end changes as you adjust the trim.

But it's cheaper to build! A balanced attenuator that tracked well
would cost more than the whole channel strip on a Mackie....
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
August 2, 2004 9:03:00 AM

Archived from groups: rec.audio.pro (More info?)

On Sun, 01 Aug 2004 18:15:47 -0800, Kurt Riemann <> wrote:

>On Sat, 31 Jul 2004 03:17:18 GMT, Steve Jorgensen
><nospam@nospam.nospam> wrote:
>
>>On Fri, 30 Jul 2004 11:01:54 -0800, Kurt Riemann <> wrote:
>>
>>>On Fri, 30 Jul 2004 13:01:17 -0400, "Arny Krueger" <arnyk@hotpop.com>
>>>wrote:
>>>
>>>>"Richard Brooks" <nobodyhere@deadspam.com> wrote in message
>>>>news:410a6d78$0$13086$cc9e4d1f@news.dial.pipex.com
>>>>
>>>>
>>>>> If you are recording a digital source onto a PC (e.g. the S/PDIF
>>>>> output from a digital multitrack) does the sound card have any effect
>>>>> on the sound quality?
>>>>
>>>>Depends on the sound card. Some resample their digital inputs. Some of those
>>>>do it badly.
>>>>
>>>
>>>Whose cards do that? How do they mess it up? They decide that a 1
>>>should be a 0? or a 16,000 should be a 16,001?
>>
>>The biggest problem is that, during resampling, it maps the incoming signal to
>>the internal clock, which has high jitter. That means the signal's timing has
>>been given a jitter exactly out of phase with the card's jitter. A perfectly
>>jitter free signal being resampled by a jittery card will therefore be
>>jittery.
>
>By jittery, do you mean that the errors are in the encoding and that
>there is a displacement of data in the stream that is then written
>incorrectly (IE later or earlier in the file by one sample) so data is
>either truncated or duplicated in the actual file? Or does random data
>come out? Like the crackle we all love so much.
>
>That's how I'm reading this. I don't use cheap sound cards and always
>clock to the source (with word if possible) so I'm unfamiliar with how
>jitter really gets the file screwed up.
>
>All I know is that by making something slave this way I've eliminated
>clicks and pops.

If a card is neither syncing with nor doing sample rate conversion on an
incoming singl, you'll get pops because occasionally, a samle is to there yet
when the card's clock is ready, or a sample in the input is skipped because
the card's clock has gotten ahead. I think the card can also just lose the
signal for a few samples when this happens.

With sample rate conversion, to the extent that the sender and/or receiver
have timing jitter, there will be jitter mapped onto the sound, and it will
get muddy and unclear. Jitter means the clock's rate is not perfectly
contant, so it gets a bit ahead, then a bit behind with respect to real time,
imposing a slight alternating time compression/decompression. The sample rate
converter is continually interpolating between consecutive samples, so there
are no actual glitches or pops, the signal simplay has timing inaccuracies.

>
>
>I'll read the other posts and see what other illumination I get.
>
>
>
>Kurt Riemann
>
>
>
>
>
>
Anonymous
August 2, 2004 10:53:05 AM

Archived from groups: rec.audio.pro (More info?)

"Steve Jorgensen" <nospam@nospam.nospam> wrote in message
news:3rirg0lmmf3e84u2ndk2hui6v7p10kbs8e@4ax.com

> If a card is neither syncing with nor doing sample rate conversion on
> an incoming singl, you'll get pops because occasionally, a sample is
> to there yet when the card's clock is ready, or a sample in the input
> is skipped because the card's clock has gotten ahead. I think the
> card can also just lose the signal for a few samples when this
> happens.

I dunno, when my digital inputs are out of synch they sound absolutely
horrible - like noise.

When my digital inputs are synched properly they sound absolutely
wonderful, no tics or pops ever.

> With sample rate conversion, to the extent that the sender and/or
> receiver have timing jitter, there will be jitter mapped onto the
> sound, and it will get muddy and unclear.

This would have to be restricted to real time sample rate conversion, right?

I always do sample rate conversion with PC software, and I've never seen
that kind of sample rate conversion of digital data add or subtract jitter.
I know of no theoretical reason why it would.

> Jitter means the clock's
> rate is not perfectly constant, so it gets a bit ahead, then a bit
> behind with respect to real time, imposing a slight alternating time
> compression/decompression.

That can happen at the point of analog <-> digital conversion. It can also
happen when you do real time sample rate conversion. Good reasons to avoid
both wherever you can, I guess.
Anonymous
August 2, 2004 10:59:15 AM

Archived from groups: rec.audio.pro (More info?)

"Laurence Payne" <l@laurenceDELETEpayne.freeserve.co.uk> wrote in
message news:rjfqg0p1nlc7fdotpbso7l6ke27fv67rmt@4ax.com
> On 1 Aug 2004 07:38:34 -0400, mrivers@d-and-d.com (Mike Rivers) wrote:
>
>> (This is the reason
>> why Mackie VLZ Pro mic preamps start to roll off the low frequencies
>> when you get near to full gain.)
>
> Could you elaborate on this? I understood the input trim control to
> be merely an attenuator - the preamp stage was always operating at
> "full gain".
>

It's an implementation thing. AFAIK we last discussed this in a circuit
analysis of a small Rane mono mic preamp.

This post will get you into the thick of the discussion"

http://www.google.com/groups?selm=kaZf9.496%24Pc6.406%4...

"There's another relevant RC in the form of the 470 uF cap the 2 K pot
and the 2 10 ohm resistors in parallel. At the minimum pot setting
this has a corner F of 67 Hz. That's the "gain" control..."
Anonymous
August 4, 2004 1:48:25 PM

Archived from groups: rec.audio.pro (More info?)

On or about Sun, 01 Aug 2004 06:44:18 GMT, Logan Shaw allegedly wrote:

> Laurence Payne wrote:
> > A justification might be the SB cards' inclusion of an on-board
> > sample-player engine - the SoundFont system. This would require to
> > work at a set sample rate.
>
> Call me crazy, but why couldn't the SoundFont thing work at its
> required sample rate and then have the sample rate conversion
> convert the SoundFont's output to the rate that the D/A is going?
> It doesn't seem like this would be a lot more expensive to build
> given that either way you include stuff that does sample rate
> conversion...

But if that is seen as a primary function of the card, it will sound
better if it has the short path. The digital in is probably seen as a
minor sideline that few gamers will use, and it's easier to blame the
other gear if it sounds lousy anyway.


Noel Bachelor noelbachelorAT(From:_domain)
Language Recordings Inc (Darwin Australia)
!