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Appreachable difference in 96khz recording?

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Anonymous
September 30, 2004 10:11:12 AM

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

I'm a newbie to digital recording, but not a newbie to music or
electronics. So that being said, I'm looking for a simple DAW setup.
I'll be using a laptop to record solo guitar for the most part. I'll
either be going firewire or usb, but I digress.... In deciding on
gear, since I'll mainly be doing solo direct in guitar work I want
good quality audio. That seems fairly easy to do since at best it will
be two inputs of stereo. But since CD quality has to be 44.1khz at 16
bits, what do you experienced people hear in terms of quality when
recording at 24 bit 96khz? Since you'll have to dither that down to CD
quality do you still prefer the higher sampling? Does the dithering
process alter the sound quality? Enquiring mind wants to know.....and
thanks for the replies......


Ron B
Anonymous
September 30, 2004 1:16:21 PM

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

Ron B <thebolins@excite.com> wrote:
>I'm a newbie to digital recording, but not a newbie to music or
>electronics. So that being said, I'm looking for a simple DAW setup.
>I'll be using a laptop to record solo guitar for the most part. I'll
>either be going firewire or usb, but I digress.... In deciding on
>gear, since I'll mainly be doing solo direct in guitar work I want
>good quality audio. That seems fairly easy to do since at best it will
>be two inputs of stereo. But since CD quality has to be 44.1khz at 16
>bits, what do you experienced people hear in terms of quality when
>recording at 24 bit 96khz? Since you'll have to dither that down to CD
>quality do you still prefer the higher sampling? Does the dithering
>process alter the sound quality? Enquiring mind wants to know.....and
>thanks for the replies......

For sampling rate, it depends entirely on the converters. Some converters
sound worse at 96 ksamp/sec than they do at 44.1 ksamp/sec. Some might
sound better. Many folks record at 88.2 ksamp/sec because it's easier to
SRC down to 44.1. That becomes a question very specific to the conversion
hardware you're running.

But the wider word length is always a good idea because it gives you more room
to be sloppy about levels.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
September 30, 2004 3:40:24 PM

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

kludge@panix.com (Scott Dorsey) wrote in message news:<cjh0v5$m3r$1@panix2.panix.com>...
> Ron B <thebolins@excite.com> wrote:
> >I'm a newbie to digital recording, but not a newbie to music or
> >electronics. So that being said, I'm looking for a simple DAW setup.
> >I'll be using a laptop to record solo guitar for the most part. I'll
> >either be going firewire or usb, but I digress.... In deciding on
> >gear, since I'll mainly be doing solo direct in guitar work I want
> >good quality audio. That seems fairly easy to do since at best it will
> >be two inputs of stereo. But since CD quality has to be 44.1khz at 16
> >bits, what do you experienced people hear in terms of quality when
> >recording at 24 bit 96khz? Since you'll have to dither that down to CD
> >quality do you still prefer the higher sampling? Does the dithering
> >process alter the sound quality? Enquiring mind wants to know.....and
> >thanks for the replies......
>
> For sampling rate, it depends entirely on the converters. Some converters
> sound worse at 96 ksamp/sec than they do at 44.1 ksamp/sec. Some might
> sound better. Many folks record at 88.2 ksamp/sec because it's easier to
> SRC down to 44.1. That becomes a question very specific to the conversion
> hardware you're running.
>
> But the wider word length is always a good idea because it gives you more room
> to be sloppy about levels.
> --scott

Thanks Scott.
So I guess the next logical question is one of recommendation. is an
MBox at 48khz tops better than a M-Audio Quattro at 96 khz (both at 24
bits). Anyone with experience with these? I would ask about the
Stienberg System|4 but the posts on that say the drivers and firmware
are really screwed up, to bad since it comes with Cubase. I also like
the OmniStudio (usb) but in most cases would need to buy the software
seperately ($$).

Ron
Related resources
Anonymous
September 30, 2004 7:15:51 PM

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

"Tony Briggs" <tbriggs@oineye.com> wrote in message
news:GqV6d.181588$3l3.47932@attbi_s03
>> But the wider word length is always a good idea because it gives you
>> more room to be sloppy about levels.
>
> Craig Anderton would agree....




> Watch these videos on this page:
> http://www.emplive.com/create/home_recording/index.asp

Just to pick a nit, he makes a factual mistake when he specifically says
that *all* "24 bit" converters have at least 16 bit resolution. They don't,
a leading example being the popular M-Audio AP2496.

If you check

http://audio.rightmark.org/rus/test/m-audio-audiophile2...

you'll see that the unweighted dynamic range in 24/96 mode is less than 95
dB.

This is just one example of many 24/96 converters with less than 96 dB
unweighted dynamic range.
Anonymous
September 30, 2004 7:15:52 PM

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

Arny Krueger wrote:

> "Tony Briggs" <tbriggs@oineye.com> wrote in message
> news:GqV6d.181588$3l3.47932@attbi_s03
>
>>>But the wider word length is always a good idea because it gives you
>>>more room to be sloppy about levels.
>>
>>Craig Anderton would agree....
>
>
>
>
>
>>Watch these videos on this page:
>>http://www.emplive.com/create/home_recording/index.asp
>
>
> Just to pick a nit, he makes a factual mistake when he specifically says
> that *all* "24 bit" converters have at least 16 bit resolution. They don't,
> a leading example being the popular M-Audio AP2496.
>
> If you check
>
> http://audio.rightmark.org/rus/test/m-audio-audiophile2...
>
> you'll see that the unweighted dynamic range in 24/96 mode is less than 95
> dB.
>
> This is just one example of many 24/96 converters with less than 96 dB
> unweighted dynamic range.
>
>
Just get and Echo MIA:)  It's 24/96 converters ROCK.

http://audio.rightmark.org/test/echo-mia.html

CD
Anonymous
September 30, 2004 11:29:40 PM

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

In article <roSdnU8UEuEftsHcRVn-ig@comcast.com> arnyk@hotpop.com writes:

> Just to pick a nit, he makes a factual mistake when he specifically says
> that *all* "24 bit" converters have at least 16 bit resolution. They don't,
> a leading example being the popular M-Audio AP2496.

> you'll see that the unweighted dynamic range in 24/96 mode is less than 95
> dB.
>
> This is just one example of many 24/96 converters with less than 96 dB
> unweighted dynamic range.

The converter could still have at least 16-bit resolution, but the
analog noise either on the input, the output, or both could still
raise the noise floor to above the theoretical limit for a 16-bit
device.

Or was this a digital in/out measurement? If so, then GULP!



--
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
October 1, 2004 12:42:05 AM

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

<< So I guess the next logical question is one of recommendation. is an
MBox at 48khz tops better than a M-Audio Quattro at 96 khz (both at 24
bits). >>

At doing what?



---------------------------------------
"I know enough to know I don't know enough"
Anonymous
October 1, 2004 1:12:16 AM

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

What audio codec do they use?
"Tony Briggs" <tbriggs@oineye.com> wrote in message
news:GqV6d.181588$3l3.47932@attbi_s03...
>
> > But the wider word length is always a good idea because it gives you
more
> room
> > to be sloppy about levels.
>
> Craig Anderton would agree....
> Watch these videos on this page:
> http://www.emplive.com/create/home_recording/index.asp
>
>
Anonymous
October 1, 2004 2:38:49 PM

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

Definitely. Lots and lots of preaching.
Anonymous
October 1, 2004 7:11:57 PM

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

"Ron B" <thebolins@excite.com> wrote in message
news:158b4dd.0409300511.678fbad8@posting.google.com...

> be two inputs of stereo. But since CD quality has to be 44.1khz at 16
> bits, what do you experienced people hear in terms of quality when
> recording at 24 bit 96khz? Since you'll have to dither that down to CD
> quality do you still prefer the higher sampling? Does the dithering
> process alter the sound quality?



While I applaud *any* step towards better sound, my practical experience
has been that there is very little benefit to higher sampling rates. So
little, in fact, that I decided it wasn't worth the extra processing
overhead and storage requirements. I've gone back to recording at 44.1
for CD-destined material and 48K for video stuff.

On the other hand, I found a genuine, appreciable improvement going to
24 bit from 16. I can keep levels more sensible while tracking without
a graininess penalty. Mostly, though, I noticed a significant
improvement on sessions with a lot of processing going on. Maybe the
longer digital word keeps the calculations more accurate so the result
sounds better?

I dither down to 16 bits as the very last step, and have found the
result sounds better than if I used a 16 bit session throughout.

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)
Anonymous
October 2, 2004 11:49:35 AM

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

I hear a definite difference in clarity in the high end when I go from
44.1 to 48kHz. It is subtle but there. I do 48kHz at 24 bit and then
at the end use the Waves L2 on the bounce before going to CD, but with
a flat noise shaping curve, which also seems to lend clarity to the
high end when compared to a shaped curve. There is a quantifiable
reason for that having to do with convertor design/philosophy which is
too deep for this thread.

I have never recorded at the higher rates (at present in the interest
of disk space conservation - I never had a DAT machine that would do
96k). However, my opinion is that all of the calculations made in
processing (plug-ins and such) will end up being more accurate if you
are using a higher sample rate and bit depth than your end target,
which will resort in less distortion of your final product than errors
made in calculation going from 48kHz back down to 44.1.

As mentioned before though, I think the quality of the convertor makes
a bigger difference than the nominal sample rate and bit depth. An
Apogee Rosetta 200 at 44.1/16 bit will no doubt sound "better" than an
M-Audio interface at 96/24.

All of this is from my unscientific $.02 which is probably more than
it's worth anyway :-)
Anonymous
October 2, 2004 2:53:19 PM

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

locosoundman <locosoundman@netscape.net> wrote:
>I hear a definite difference in clarity in the high end when I go from
>44.1 to 48kHz. It is subtle but there. I do 48kHz at 24 bit and then
>at the end use the Waves L2 on the bounce before going to CD, but with
>a flat noise shaping curve, which also seems to lend clarity to the
>high end when compared to a shaped curve. There is a quantifiable
>reason for that having to do with convertor design/philosophy which is
>too deep for this thread.

This is sign that you have some kind of converter issue going on, or else
the clarity improvement actually has to do with the noise shaping (which
is very common).

What sort of converters are you using?

>I have never recorded at the higher rates (at present in the interest
>of disk space conservation - I never had a DAT machine that would do
>96k). However, my opinion is that all of the calculations made in
>processing (plug-ins and such) will end up being more accurate if you
>are using a higher sample rate and bit depth than your end target,
>which will resort in less distortion of your final product than errors
>made in calculation going from 48kHz back down to 44.1.

There are some things, like noise reduction for 78s, that benefit a lot
from ultrasonic response, even if ultrasonics are inaudible. That alone
makes high sampling rates a useful thing to have around in the studio.
I'm not convinced it's something I'd want to use all the time yet, though.
But then, I'm still using the Ampex half the time.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 2, 2004 5:59:39 PM

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

"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:roSdnU8UEuEftsHcRVn-ig@comcast.com...
> Just to pick a nit, he makes a factual mistake when he specifically says
> that *all* "24 bit" converters have at least 16 bit resolution. They
don't,
> a leading example being the popular M-Audio AP2496.
> If you check
>
http://audio.rightmark.org/rus/test/m-audio-audiophile2...
tml
> you'll see that the unweighted dynamic range in 24/96 mode is less than 95
> dB.
> This is just one example of many 24/96 converters with less than 96 dB
> unweighted dynamic range.

Which is what I've been saying for years when people claim 24 bit gives much
more room for sloppy level setting.
In the case of the AP2496 there is no real extra headroom only wasted bits.
In the case of *MANY* 24 bit cards there is a gain of 1 bit extra
resolution, 2 if you're lucky, and 3 only in the case of the very best cards
available. However 95dB unweighted is quite adequate for tracking, and
better than any analog recorder anyway.

TonyP.

"Appreachable" is a good word for this group too, with SO many people
preaching their twisted versions of the audio gospel :-)
Anonymous
October 2, 2004 5:59:40 PM

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

TonyP <TonyP@optus.net.com.au> wrote:
>
>Which is what I've been saying for years when people claim 24 bit gives much
>more room for sloppy level setting.
>In the case of the AP2496 there is no real extra headroom only wasted bits.
>In the case of *MANY* 24 bit cards there is a gain of 1 bit extra
>resolution, 2 if you're lucky, and 3 only in the case of the very best cards
>available. However 95dB unweighted is quite adequate for tracking, and
>better than any analog recorder anyway.

This is not an argument against 24-bit converters. This is an argument
against calling many of those devices 24-bit unless they actually have
24 bits of real data coming out.

Many of the devices advertised out there as 24-bit, though, have twenty or
so actual significant bits, and that's not shabby. They should be called
twenty-bit converters. I know my Prism 20-bit box has at least 19 real
valid bits and only one doubtful one, which I figure is pretty good. The
Lavry stuff is at least in the same league. Some of the 20-bit soundcards
don't even have 16 valid bits.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 2, 2004 5:59:41 PM

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

On Sat, 2 Oct 2004 08:20:35 -0400, Scott Dorsey wrote
(in article <cjm6ej$8og$1@panix2.panix.com>):

> TonyP <TonyP@optus.net.com.au> wrote:
>>
>> Which is what I've been saying for years when people claim 24 bit gives much
>> more room for sloppy level setting.
>> In the case of the AP2496 there is no real extra headroom only wasted bits.
>> In the case of *MANY* 24 bit cards there is a gain of 1 bit extra
>> resolution, 2 if you're lucky, and 3 only in the case of the very best cards
>> available. However 95dB unweighted is quite adequate for tracking, and
>> better than any analog recorder anyway.
>
> This is not an argument against 24-bit converters. This is an argument
> against calling many of those devices 24-bit unless they actually have
> 24 bits of real data coming out.
>
> Many of the devices advertised out there as 24-bit, though, have twenty or
> so actual significant bits, and that's not shabby. They should be called
> twenty-bit converters. I know my Prism 20-bit box has at least 19 real
> valid bits and only one doubtful one, which I figure is pretty good. The
> Lavry stuff is at least in the same league. Some of the 20-bit soundcards
> don't even have 16 valid bits.
> --scott

And at least as important, how do they actually sound?

Regards,

Ty Ford




-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
October 2, 2004 5:59:42 PM

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

Ty Ford <tyreeford@comcast.net> wrote:
>And at least as important, how do they actually sound?

I will make a bet that there is a direct corellation between monotonicity
specs and the actual sound. It would be a fun thing to test, too.

It's interesting, though, that some of the chips that have nice monotonicity
specs actually measure pretty poorly when you throw a cheap op-amp front end
on them. It's not the converter proper that has the linearity problem.

It is lots of fun to look at the noise floor on typical converters, though.
The Prism is pretty flat and even, but some of them have big spikes here
and there.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 2, 2004 6:33:22 PM

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

"Lorin David Schultz" <Lorin@DAMNSPAM!v5v.ca> wrote in message
news:1te7d.6621$223.32@edtnps89...
> On the other hand, I found a genuine, appreciable improvement going to
> 24 bit from 16. I can keep levels more sensible while tracking without
> a graininess penalty. Mostly, though, I noticed a significant
> improvement on sessions with a lot of processing going on.
> Maybe the longer digital word keeps the calculations more accurate so
> the result sounds better?

Or maybe not in the case of most modern DAW software which processes at 32
bit or even 64 bit!
Just be carefull to save any intermediate files at 32 bit, not 16 bit.
Problem solved.

> I dither down to 16 bits as the very last step, and have found the
> result sounds better than if I used a 16 bit session throughout.

Of course, but have you tried recording at 16/44 and processing at 32/44 or
64/44?
Or done a double blind listening test of a file recorded at 16/44 and one at
24/44 with the same card?
Of course hard drive space is so cheap these days, it's really a non issue
for most people now. Saving heaps of wasted bits doesn't cost you too much
as long as the hardware can handle the overhead.

TonyP.
Anonymous
October 2, 2004 7:49:37 PM

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

kludge@panix.com (Scott Dorsey) wrote in
news:cjmek8$fr5$1@panix3.panix.com:

> Ty Ford <tyreeford@comcast.net> wrote:
>>And at least as important, how do they actually sound?
>
> I will make a bet that there is a direct corellation between
> monotonicity specs and the actual sound. It would be a fun thing to
> test, too.
>
> It's interesting, though, that some of the chips that have nice
> monotonicity specs actually measure pretty poorly when you throw a
> cheap op-amp front end on them. It's not the converter proper that
> has the linearity problem.
>
> It is lots of fun to look at the noise floor on typical converters,
> though. The Prism is pretty flat and even, but some of them have big
> spikes here and there.
> --scott

You've lost me again, Scott. I know mathematically what monotonicity
means, but what are you correlating in the converter? Analog to digital?
That's a pretty tough test to measure.
Anonymous
October 2, 2004 7:49:38 PM

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

Carey Carlan <gulfjoe@hotmail.com> wrote:
>You've lost me again, Scott. I know mathematically what monotonicity
>means, but what are you correlating in the converter? Analog to digital?
>That's a pretty tough test to measure.

Right. It's very difficult to measure, which is part of the problem. The
data sheet numbers are often derived from models rather than actually measured,
too.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 2, 2004 11:30:34 PM

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

First, I'd argue about the concept of sloppy level setting. If/when you
have converters of decent quality to allow for at least 16 bit recording
(i.e. -96 dB noise floor) then you have 8 bits of headroom at the max (a
theoretical impossibility), but more like 4 bits, but geez, doesn't that
equate to 24 dB of headroom, which is the typical amount of a good console's
analog output headroom?

So what does "sloppy" mean in relation to converters? I'd like to see
someone actualy define, in meaningful terms, just what this means. First,
it can't mean the same thing to two different people because no one works
exactly the same as another with exactly the same equipment. Second, there
are no methods to measure a term such as "sloppy". It's all too easy to say
someone was sloppy in their signal path setup, but in the worst of
situations, sans digital converters, even The Boss' Nebraska became a
seminal album on cassette.

--


Roger W. Norman
SirMusic Studio

"TonyP" <TonyP@optus.net.com.au> wrote in message
news:415e27d4$0$20124$afc38c87@news.optusnet.com.au...
>
> "Arny Krueger" <arnyk@hotpop.com> wrote in message
> news:roSdnU8UEuEftsHcRVn-ig@comcast.com...
> > Just to pick a nit, he makes a factual mistake when he specifically says
> > that *all* "24 bit" converters have at least 16 bit resolution. They
> don't,
> > a leading example being the popular M-Audio AP2496.
> > If you check
> >
>
http://audio.rightmark.org/rus/test/m-audio-audiophile2...
> tml
> > you'll see that the unweighted dynamic range in 24/96 mode is less than
95
> > dB.
> > This is just one example of many 24/96 converters with less than 96 dB
> > unweighted dynamic range.
>
> Which is what I've been saying for years when people claim 24 bit gives
much
> more room for sloppy level setting.
> In the case of the AP2496 there is no real extra headroom only wasted
bits.
> In the case of *MANY* 24 bit cards there is a gain of 1 bit extra
> resolution, 2 if you're lucky, and 3 only in the case of the very best
cards
> available. However 95dB unweighted is quite adequate for tracking, and
> better than any analog recorder anyway.
>
> TonyP.
>
> "Appreachable" is a good word for this group too, with SO many people
> preaching their twisted versions of the audio gospel :-)
>
>
Anonymous
October 2, 2004 11:34:43 PM

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

Glenn Meadows has often said that the quality of the sound of the conversion
is based on the analog input. Shitty input, shitty output. Marrying good
converters to a good analog front end, like perhaps Prism, Mytek or Lavry or
a number of others will produce superior results with the same converters
that perhaps MOTU uses. When I look at the input from my RME or my MOTU,
both have analog in the way. How not?

--


Roger W. Norman
SirMusic Studio

"Ty Ford" <tyreeford@comcast.net> wrote in message
news:V-WdnRHGzfziJ8PcRVn-tw@comcast.com...
> On Sat, 2 Oct 2004 08:20:35 -0400, Scott Dorsey wrote
> (in article <cjm6ej$8og$1@panix2.panix.com>):
>
> > TonyP <TonyP@optus.net.com.au> wrote:
> >>
> >> Which is what I've been saying for years when people claim 24 bit gives
much
> >> more room for sloppy level setting.
> >> In the case of the AP2496 there is no real extra headroom only wasted
bits.
> >> In the case of *MANY* 24 bit cards there is a gain of 1 bit extra
> >> resolution, 2 if you're lucky, and 3 only in the case of the very best
cards
> >> available. However 95dB unweighted is quite adequate for tracking, and
> >> better than any analog recorder anyway.
> >
> > This is not an argument against 24-bit converters. This is an argument
> > against calling many of those devices 24-bit unless they actually have
> > 24 bits of real data coming out.
> >
> > Many of the devices advertised out there as 24-bit, though, have twenty
or
> > so actual significant bits, and that's not shabby. They should be
called
> > twenty-bit converters. I know my Prism 20-bit box has at least 19 real
> > valid bits and only one doubtful one, which I figure is pretty good.
The
> > Lavry stuff is at least in the same league. Some of the 20-bit
soundcards
> > don't even have 16 valid bits.
> > --scott
>
> And at least as important, how do they actually sound?
>
> Regards,
>
> Ty Ford
>
>
>
>
> -- Ty Ford's equipment reviews, audio samples, rates and other
audiocentric
> stuff are at www.tyford.com
>
Anonymous
October 2, 2004 11:34:49 PM

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

"TonyP" <TonyP@optus.net.com.au> wrote
>
> Of course, but have you tried recording at 16/44 and processing at
32/44 or
> 64/44?

I don't know if I have or not. What's Pro Tools' internal processing
depth?



> Or done a double blind listening test of a file recorded at 16/44 and
one at
> 24/44 with the same card?

Yeah, and there's not enough difference to care about if that's all I
was gonna do with it. The benefit of the longer word really only became
apparent to me on sessions with a fair amount of processing going on.
"Lots of processing" includes things like mixing a lot of tracks, making
any fairly healthy level changes, and/or applying plugs.



> Of course hard drive space is so cheap these days, it's really a non
issue
> for most people now. Saving heaps of wasted bits doesn't cost you too
much
> as long as the hardware can handle the overhead.

I suppose, within reason. At some point the benefits become so small
that practical realities like backup space/cost, processing time (when
the client is paying by the hour) and plug-in limits become a higher
priority than absolute limits of audiophility.

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)
Anonymous
October 3, 2004 3:04:47 AM

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

Roger W. Norman <rnorman@starpower.net> wrote:
>So what does "sloppy" mean in relation to converters? I'd like to see
>someone actualy define, in meaningful terms, just what this means. First,
>it can't mean the same thing to two different people because no one works
>exactly the same as another with exactly the same equipment. Second, there
>are no methods to measure a term such as "sloppy". It's all too easy to say
>someone was sloppy in their signal path setup, but in the worst of
>situations, sans digital converters, even The Boss' Nebraska became a
>seminal album on cassette.

It means you can record a 20 bit signal and have 16 valid bits and also
4 bits of "headroom." You can record so the signal never goes above -24dB
on the meter, and still have a full valid 16 bits for release.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 4, 2004 1:14:15 PM

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

"Carey Carlan" <gulfjoe@hotmail.com> wrote in message
news:Xns95767871C1D7Dgulfjoehotmailcom@207.69.189.194

> You've lost me again, Scott. I know mathematically what monotonicity
> means, but what are you correlating in the converter? Analog to
> digital? That's a pretty tough test to measure.

Depending on the nature of the non-monotonicity, you'll have nonlinear
distortion and/or modulation noise. On a really bad day the lack of
monotonicity will be in excess of the quantization noise that the dither is
supposed to handle, and you'll have dither failture.
Anonymous
October 4, 2004 10:08:59 PM

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

"Scott Dorsey" <kludge@panix.com> wrote in message
news:cjnq8f$chn$1@panix1.panix.com...
> It means you can record a 20 bit signal and have 16 valid bits and also
> 4 bits of "headroom." You can record so the signal never goes above -24dB
> on the meter, and still have a full valid 16 bits for release.

But you will only have 4 VALID bits for headroom, if your signal chain has a
DNR of 120dB unweighted.
***VERY*** unlikely.
If you use 4 INVALID bits for your signal (by throwing away 4 valid bits as
headroom) then you only have 12 valid bits left for release.

Some people here seem to think they can just use 4 invalid bits for
headroom, which is an impossibility. The invalid bits are at the other end.
The question remains whether you really need more than 14 valid bits (84dB
DNR), or more than 12 dB headroom though. Especially if the final result
will be highly compressed and clipped anyway, as is most pop music these
days.

TonyP.
Anonymous
October 4, 2004 10:09:00 PM

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

TonyP <TonyP@optus.net.com.au> wrote:
>"Scott Dorsey" <kludge@panix.com> wrote in message
>news:cjnq8f$chn$1@panix1.panix.com...
>> It means you can record a 20 bit signal and have 16 valid bits and also
>> 4 bits of "headroom." You can record so the signal never goes above -24dB
>> on the meter, and still have a full valid 16 bits for release.
>
>But you will only have 4 VALID bits for headroom, if your signal chain has a
>DNR of 120dB unweighted.
>***VERY*** unlikely.
>If you use 4 INVALID bits for your signal (by throwing away 4 valid bits as
>headroom) then you only have 12 valid bits left for release.

Depends on whether there is any correlated information below the noise floor.
On some systems, where the noise floor is mostly from the analogue side,
there may be plenty of useful stuff there. On other systems there might not
be.

>Some people here seem to think they can just use 4 invalid bits for
>headroom, which is an impossibility. The invalid bits are at the other end.

How invalid are they and what's on them? That's the question.

>The question remains whether you really need more than 14 valid bits (84dB
>DNR), or more than 12 dB headroom though. Especially if the final result
>will be highly compressed and clipped anyway, as is most pop music these
>days.

I dunno, I have heard some 14-bit audio that sounded pretty good, and I have
heard some 24-bit audio that sounded really bad too. But to my mind, having
additional dynamic range isn't a bad thing. Thinking you have additional
dynamic range when you really don't is.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 5, 2004 8:50:49 PM

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

"Scott Dorsey" <kludge@panix.com> wrote in message
news:cjrk5a$dg1$1@panix2.panix.com...
> TonyP <TonyP@optus.net.com.au> wrote:
> >"Scott Dorsey" <kludge@panix.com> wrote in message
> >news:cjnq8f$chn$1@panix1.panix.com...
> >> It means you can record a 20 bit signal and have 16 valid bits and also
> >> 4 bits of "headroom." You can record so the signal never goes
above -24dB
> >> on the meter, and still have a full valid 16 bits for release.
> >
> >But you will only have 4 VALID bits for headroom, if your signal chain
has a
> >DNR of 120dB unweighted.
> >***VERY*** unlikely.
> >If you use 4 INVALID bits for your signal (by throwing away 4 valid bits
as
> >headroom) then you only have 12 valid bits left for release.
>
> Depends on whether there is any correlated information below the noise
floor.
> On some systems, where the noise floor is mostly from the analogue side,
> there may be plenty of useful stuff there.


I'm not sure what you mean here, can you tell me what you can get below the
wide band noise floor on analog that you cannot also get on Properly
Dithered digital? Wide band and narrow band noise floors are not unique to
any system.


> >Some people here seem to think they can just use 4 invalid bits for
> >headroom, which is an impossibility. The invalid bits are at the other
end.
>
> How invalid are they and what's on them? That's the question.

Noise.
If they have truly useful signal information, they aren't invalid!


> I dunno, I have heard some 14-bit audio that sounded pretty good, and I
have
> heard some 24-bit audio that sounded really bad too. But to my mind,
having
> additional dynamic range isn't a bad thing. Thinking you have additional
> dynamic range when you really don't is.

Exactly my point to begin with. So many people here think they have heaps of
head room just because the file size is 24 bits.

TonyP.
Anonymous
October 5, 2004 8:50:50 PM

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

TonyP <TonyP@optus.net.com.au> wrote:
Kludge writes:
>>
>> Depends on whether there is any correlated information below the noise
>floor.
>> On some systems, where the noise floor is mostly from the analogue side,
>> there may be plenty of useful stuff there.
>
>
>I'm not sure what you mean here, can you tell me what you can get below the
>wide band noise floor on analog that you cannot also get on Properly
>Dithered digital? Wide band and narrow band noise floors are not unique to
>any system.

Nothing. It's the same process, and with both analogue and digital there
can be useful information below the broadband noise floor. The problem
with digital systems is that they often have nonlinearity issues above the
noise floor, and the nonlinearity at low level becomes significant before
the noise does.

The point I was making is that many converter units out there have very
poor analogue front ends on them, and that a lot of the noise on inexpensive
converters is from the analogue front end rather than from the ladder itself.
With these systems, there is apt to be a high noise floor, but there might
well be usable information below the noise floor.

>> >Some people here seem to think they can just use 4 invalid bits for
>> >headroom, which is an impossibility. The invalid bits are at the other
>end.
>>
>> How invalid are they and what's on them? That's the question.
>
>Noise.
>
>If they have truly useful signal information, they aren't invalid!

Okay, let's say you have an early Flying Cow A/D, that has a huge spike at
60 Hz. It's basically a ten bit converter... the bottom ten bits just have
60 Hz noise on them. But let's say you take the output of that and put it
into a notch filter and take the 60 Hz noise out... then you have something
like fifteen or sixteen valid bits. If the noise is correlated and removable,
the data isn't necessarily invalid.

Likewise if you have a 60 dB broadband hiss, and you add to it music that
is 3 dB below the noise level, you'll still be able to make the words out,
because there is useful information still below the noise. In this case,
it's the signal that is correlated and not the noise.

>> I dunno, I have heard some 14-bit audio that sounded pretty good, and I
>have
>> heard some 24-bit audio that sounded really bad too. But to my mind,
>having
>> additional dynamic range isn't a bad thing. Thinking you have additional
>> dynamic range when you really don't is.
>
>Exactly my point to begin with. So many people here think they have heaps of
>head room just because the file size is 24 bits.

That's true. But the split between conversion and storage word length is
ANOTHER whole can of worms.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
October 5, 2004 8:50:50 PM

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

In article <41624471$0$23894$afc38c87@news.optusnet.com.au> TonyP@optus.net.com.au writes:

> Exactly my point to begin with. So many people here think they have heaps of
> head room just because the file size is 24 bits.

You don't need heaps of headroom, 20 dB is almost always enough. You
just have to figure out which portion of the 140 dB of theoretical
dynamic range is most useful to get the headroom and noise floor that
you find acceptable.

Also, you need to accept the practical. Just because you can measure
noise 95 dB below your nominal signal level doesn't mean that someone
listening to your CD in their car (or even on the audiophile system in
your living room) will say "geez, what a noisy recording." And if they
do, so what? You know better.


--
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
October 7, 2004 6:58:41 PM

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

"Scott Dorsey" <kludge@panix.com> wrote in message
news:cju7pf$g0r$1@panix2.panix.com...
> TonyP <TonyP@optus.net.com.au> wrote:
> >I'm not sure what you mean here, can you tell me what you can get below
the
> >wide band noise floor on analog that you cannot also get on Properly
> >Dithered digital? Wide band and narrow band noise floors are not unique
to
> >any system.
>
> Nothing. It's the same process, and with both analogue and digital there
> can be useful information below the broadband noise floor.

Exactly.

>The problem
> with digital systems is that they often have nonlinearity issues above the
> noise floor, and the nonlinearity at low level becomes significant before
> the noise does.

I've not seen a digital system in over ten years that was worse than analog
systems in this regard. But I'll accept that you may have.

> The point I was making is that many converter units out there have very
> poor analogue front ends on them, and that a lot of the noise on
inexpensive
> converters is from the analogue front end rather than from the ladder
itself.

Yep, it's analog once again that is the problem.

> With these systems, there is apt to be a high noise floor, but there might
> well be usable information below the noise floor.

As with all systems.

> >If they have truly useful signal information, they aren't invalid!
>
> Okay, let's say you have an early Flying Cow A/D, that has a huge spike at
> 60 Hz. It's basically a ten bit converter... the bottom ten bits just
have
> 60 Hz noise on them. But let's say you take the output of that and put it
> into a notch filter and take the 60 Hz noise out... then you have
something
> like fifteen or sixteen valid bits. If the noise is correlated and
removable,
> the data isn't necessarily invalid.

Agreed, that's what I said.

> Likewise if you have a 60 dB broadband hiss, and you add to it music that
> is 3 dB below the noise level, you'll still be able to make the words out,
> because there is useful information still below the noise. In this case,
> it's the signal that is correlated and not the noise.

Yep with any system. It's all a matter of knowing what the measurements
really mean.

> >Exactly my point to begin with. So many people here think they have heaps
of
> >head room just because the file size is 24 bits.
>
> That's true. But the split between conversion and storage word length is
> ANOTHER whole can of worms.

Actually it was the point I was making all along, including processing word
length, which can be, and usually is higher again.

TonyP.
Anonymous
October 7, 2004 6:58:42 PM

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

In article <4164cd28$0$10347$afc38c87@news.optusnet.com.au> TonyP@optus.net.com.au writes:

> Yep, it's analog once again that is the problem.

Tony, please go argue against analog somewhere else and don't come
back without a direct digital injection system to your brain. If it
isn't analog somewhere, I can't hear it and you can't either, and the
first and last links of the chain are far worse than any electronics,
always.


--
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
!