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Headroom on DAWs

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Anonymous
August 29, 2005 6:15:02 AM

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

Quick question...

On modern recording software like Logic, Cubase, PT etc, what headroom can I
expect on each channel? In other words, assuming the source wav isn't
clipped, can I (hypothetically... not that I'd want to) push the channel
gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
experience clipping?

I'm asking because sometimes it's so much easier to boost a snare track into
+0 territory than to bring everything else down when you're running 30 or
more tracks. Can I do this on a DAW (no external mixer) without worrying
about clipping a track like this, provided the output buss is lowered to the
point where the levels don't clip?

Thanks

TJ

More about : headroom daws

Anonymous
August 29, 2005 6:15:03 AM

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

You're no longer in the analog world when it comes to clipping. The
ultimate amount you can go to is 0. At zero you clip and it ain't
pretty.

When you record, set your mic chain so you are a good safe distance
below 0. Then, if you want to be safe, set up a compressor so anything
within -3 to -5 or so starts to effect and limit it with a hard wall
just under 0.

When you mix, you don't need to worry about it as much. You will have
a little light come on to letyou know when you reach 0+. The best way,
at least for me, to set up a mix is to start with the lead instrument
(vocals usually) and add to it. If you are recording in 24 bit, there
is so much room you don't have to think about it like you would in
the tape days. Noise hardly ever happens.
Anonymous
August 29, 2005 7:26:38 AM

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

"Danny Taddei" <palmtreedreamer@aol.com> wrote in message
news:1125279442.856964.72500@z14g2000cwz.googlegroups.com...
> You're no longer in the analog world when it comes to clipping. The
> ultimate amount you can go to is 0. At zero you clip and it ain't
> pretty.
>
> When you record, set your mic chain so you are a good safe distance
> below 0. Then, if you want to be safe, set up a compressor so anything
> within -3 to -5 or so starts to effect and limit it with a hard wall
> just under 0.
>
> When you mix, you don't need to worry about it as much. You will have
> a little light come on to letyou know when you reach 0+. The best way,
> at least for me, to set up a mix is to start with the lead instrument
> (vocals usually) and add to it. If you are recording in 24 bit, there
> is so much room you don't have to think about it like you would in
> the tape days. Noise hardly ever happens.

I'm not sure you understood my question - when I record, obviously I ensure
no clipping occurs; but I'm only referring to the mixdown stage and whether
or not individual channels can then go above 0 without clipping PROVIDED the
output buss is turned down to the point where it never exceeds 0.

Thanks

TJ
Related resources
August 29, 2005 7:55:59 AM

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

TJ Hertz ha scritto:
> Quick question...
>
> On modern recording software like Logic, Cubase, PT etc, what headroom can I
> expect on each channel? In other words, assuming the source wav isn't
> clipped, can I (hypothetically... not that I'd want to) push the channel
> gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
> experience clipping?
>
> I'm asking because sometimes it's so much easier to boost a snare track into
> +0 territory than to bring everything else down when you're running 30 or
> more tracks. Can I do this on a DAW (no external mixer) without worrying
> about clipping a track like this, provided the output buss is lowered to the
> point where the levels don't clip?
>
> Thanks
>
> TJ
>
>
yes, in cubase you can go over 0 dbfs on the individual channels without
internal clipping the waveform. in other sequencer softwares too.
This because the software internally work at a greater resolution than
the individual source files.
But, you have to control the output section during the mixdown process
to avoid clipping the output.
In other words, if you have a normalized source (top at 0 dbfs) raised
by about 6 db, you need to reduce the master setting by almost -6 db to
avoid clipping.
Usually the sequencer don't clip the waveform between stages.
Anonymous
August 29, 2005 8:31:54 AM

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

TJ Hertz wrote:

> I'm not sure you understood my question - when I record, obviously I ensure
> no clipping occurs; but I'm only referring to the mixdown stage and whether
> or not individual channels can then go above 0 without clipping PROVIDED the
> output buss is turned down to the point where it never exceeds 0.

It depends on how "0" is defined internally. If you're asking if you
can push the "fader" above zero, sure, assuming that "0" on the fader
is below full scale. The thing to watch is the mix bus meters, not the
position of the fader. And of course listen for distortion.

The concept of "headroom" in a digital system is different than in an
analog system, because there's a hard limit. The thing that you need to
understand is that you can have as much or as little headroom as you
want. Leaving 20 dB above the nomrmal level is considered good practice
for a 24-bit system. This allows sufficient headroom for normal audio
peaks. But if you've heavily limited your tracks so that there really
are no peaks more than, say 6 dB above the nominal level, you can boost
the track as much as 14 dB and still not have peaks that try to exceed
full scale.

However, understand that 24 (or even two) tracks that have frequent
peaks at or very close to full scale are likely to sum to something
greater than full scale, yielding clipping.

Pardon my shouting, but THERE IS ABSOLUTELY NO REASON TO DO THIS ! ! !
! ! !

Your time is better spent making a good mix than boosting tracks to the
point where you're risking clipping and then have to worry about that.
You can always boost your mix to clipping and beyond if you want it to
sound awful but louder.
Anonymous
August 29, 2005 9:50:53 AM

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

"TJ Hertz" <tjhertz@gmail.com> wrote in message
news:43126195$0$18643$14726298@news.sunsite.dk
> Quick question...
>
> On modern recording software like Logic, Cubase, PT etc,
> what headroom can I expect on each channel?

Depends on the configuration of the channel, whether it is
floating point or fixed point. With a floating point channel
you have gobs of headroom, 100's of dB or more. With fixed
point, you typically have no headroom at all.

I don't know exactly which of the dozens of alternative
software choices supports mixing with floating point files,
and which doesn't - you'll have to research that yourself.
I know Audition/CE and it supports either fixed or floating
point files. It also supports a fxied or floating point
mixing bus. The fix/float status of the mixing bus in
Audition/CE is up to the user.

> In other
> words, assuming the source wav isn't clipped, can I
> (hypothetically... not that I'd want to) push the channel
> gain to 10dB over 0 then reduce it by -10dB at the output
> fader, and not experience clipping?

That's a different question. The question you seem to be
asking is how much headroom can you expect on the mixing
bus. Again, the fixed point versus floating point issue is
relevant.

With a floating point mixng bus you have gobs of headroom,
100's of dB or more. With fixed point, you typically have no
headroom at all.

If you have software with floating point mixing bus (e.g.
Audition/CE), then you can push it past normal FS and bring
it down later on in the processing.
Anonymous
August 29, 2005 11:04:56 AM

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

> On modern recording software like Logic, Cubase, PT etc, what headroom can
I
> expect on each channel? In other words, assuming the source wav isn't
> clipped, can I (hypothetically... not that I'd want to) push the channel
> gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
> experience clipping?
>
> I'm asking because sometimes it's so much easier to boost a snare track
into
> +0 territory than to bring everything else down when you're running 30 or
> more tracks. Can I do this on a DAW (no external mixer) without worrying
> about clipping a track like this, provided the output buss is lowered to
the
> point where the levels don't clip?

If I or anyone else told you the clip lights indicate "it sounds bad", even
though it still sounds good to you, what good are we?

If you can't hear digital clipping you're out of a job.
Anonymous
August 29, 2005 1:43:37 PM

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

TJ Hertz <tjhertz@gmail.com> wrote:
>
>On modern recording software like Logic, Cubase, PT etc, what headroom can I
>expect on each channel? In other words, assuming the source wav isn't
>clipped, can I (hypothetically... not that I'd want to) push the channel
>gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
>experience clipping?

There is no headroom. It goes to zero, then it clips.

>I'm asking because sometimes it's so much easier to boost a snare track into
>+0 territory than to bring everything else down when you're running 30 or
>more tracks. Can I do this on a DAW (no external mixer) without worrying
>about clipping a track like this, provided the output buss is lowered to the
>point where the levels don't clip?

No, but there is probably a gain function that you can apply to drop all
channels by a certain level.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
August 29, 2005 7:16:05 PM

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

On Mon, 29 Aug 2005 09:43:37 -0400, Scott Dorsey wrote:

> TJ Hertz <tjhertz@gmail.com> wrote:
>>
>>On modern recording software like Logic, Cubase, PT etc, what headroom can I
>>expect on each channel? In other words, assuming the source wav isn't
>>clipped, can I (hypothetically... not that I'd want to) push the channel
>>gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
>>experience clipping?
>
> There is no headroom. It goes to zero, then it clips.

That's not true, on Cubase at least.
You can go as far as you like over zero, route it all into a group and
have that group peaking at +40db. As long as the very last fader in the
chain before the A/Ds reduces the gain so it does not go over zero, it
won't clip.

Insert plugins will probably clip with these very large signals, but the
headroom of the 32bit float mixer appears almost infinite.

>
>>I'm asking because sometimes it's so much easier to boost a snare track into
>>+0 territory than to bring everything else down when you're running 30 or
>>more tracks. Can I do this on a DAW (no external mixer) without worrying
>>about clipping a track like this, provided the output buss is lowered to the
>>point where the levels don't clip?
>
> No, but there is probably a gain function that you can apply to drop all
> channels by a certain level.
> --scott
Anonymous
August 29, 2005 7:22:42 PM

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

TJ,

Dont' worry about the fader position so much. You'll hear the individual
track "crackling" if you're pushing it too far - - at least it works that
way in Sonar. But if your recorded level is a bit low, you won't introduce
any problems by moving the fader indication into the +. Just watch the
individual track's output meter.

I normally don't have tracks that I've got to push the fader above '0', but
my track levels are usually pretty big to begin with just because I'm trying
to get a good level when I can.

dik

"TJ Hertz" <tjhertz@gmail.com> wrote in message
news:43126195$0$18643$14726298@news.sunsite.dk...
> Quick question...
>
> On modern recording software like Logic, Cubase, PT etc, what headroom can
I
> expect on each channel? In other words, assuming the source wav isn't
> clipped, can I (hypothetically... not that I'd want to) push the channel
> gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
> experience clipping?
>
> I'm asking because sometimes it's so much easier to boost a snare track
into
> +0 territory than to bring everything else down when you're running 30 or
> more tracks. Can I do this on a DAW (no external mixer) without worrying
> about clipping a track like this, provided the output buss is lowered to
the
> point where the levels don't clip?
>
> Thanks
>
> TJ
>
>
Anonymous
August 29, 2005 10:42:07 PM

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

In article <1125315114.583019.55530@o13g2000cwo.googlegroups.com>,
mrivers@d-and-d.com says...
>
>
>The concept of "headroom" in a digital system is different than in an
>analog system, because there's a hard limit. The thing that you need to
>understand is that you can have as much or as little headroom as you
>want. Leaving 20 dB above the nomrmal level is considered good practice
>for a 24-bit system. This allows sufficient headroom for normal audio
>peaks. But if you've heavily limited your tracks so that there really
>are no peaks more than, say 6 dB above the nominal level, you can boost
>the track as much as 14 dB and still not have peaks that try to exceed
>full scale.


I wouldn't waste 20db headroom on any signal. I want to record as hot as
possible and take advantage of the robustness of my 24bit signal when it goes
through processing and track summing. If you leave 20db headroom you're only
using about 20bits for your signal. 24bits will allow you to be lazy and leave
a huge headroom and never worry about clipping and still come away with a
decent recording, and I'd consider doing that if I had $5000+ a/d converters,
but with my pro-sumer level ones, I'm looking to stretch as much quality out of
them as possible, which means recording a signal as hot as possible


>However, understand that 24 (or even two) tracks that have frequent
>peaks at or very close to full scale are likely to sum to something
>greater than full scale, yielding clipping.
>
>Pardon my shouting, but THERE IS ABSOLUTELY NO REASON TO DO THIS ! ! !
>! ! !

Of course there is, you lower the db on each track successively (hopefully your
DAW will let you gang them), and keep mixing down until you're just under
clipping. Extra work? Sure, lots. Better results? Definitely. Worth it?
Your call.


>
>Your time is better spent making a good mix than boosting tracks to the
>point where you're risking clipping and then have to worry about that.
>You can always boost your mix to clipping and beyond if you want it to
>sound awful but louder.
>

Yep, and one key to making a good mix is to fill up as many bits as you can
on each track with signal.
Anonymous
August 29, 2005 10:42:08 PM

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

"Chevdo" <Chevdo@chevdo.com> wrote in message
news:3GIQe.158818$wr.58469@clgrps12

> I wouldn't waste 20db headroom on any signal. I want to
> record as hot as possible and take advantage of the
> robustness of my 24bit signal when it goes through
> processing and track summing.

Fact is that in the real world of music recording, there are
no actual 24 bit signals. With even the most expensive
commercial DACs reproducing laboratory test signals, there
*are* 20 bit signals. Just 20 bits. The other 4 bits are
noise on a good day.

Once you move out into the real world and hook up a
microphone, there are signals with far more noise. The
widest-range commercial recording I've ever analyzed had
less than 75 dB dynamic range. That's less than 13 bits. If
you huff and puff you might be able to make one track with
85 dB dynamic range, but watch that go to San Diego in a
handbasket when you start mixing it with a bunch of other
tracks.

> If you leave 20db headroom
> you're only using about 20bits for your signal.

Not a problem given that actual 20 bit signals only exist in
the laboratory, and disappear instantly once you hook up a
microphone, a console, etc.

> 24bits
> will allow you to be lazy and leave a huge headroom

How about 10 bits of headroom?
How about 15 bits of headroom?
How about 20 bits of headroom?

> never worry about clipping and still come away with a
> decent recording, and I'd consider doing that if I had
> $5000+ a/d converters, but with my pro-sumer level ones,
> I'm looking to stretch as much quality out of them as
> possible, which means recording a signal as hot as
> possible

The better prosumer converters have maybe 17 bits of
resolution (102 dB or so). If you want to do the 20-bit
boogey you are talking about some pretty fine stuff. If you
want to dance with 24 bits, you're talking liquid nitrogen
or helium or some such.
Anonymous
August 29, 2005 10:42:08 PM

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

In article <3GIQe.158818$wr.58469@clgrps12>, Chevdo <Chevdo@chevdo.com> wrote:
>I wouldn't waste 20db headroom on any signal. I want to record as hot as
>possible and take advantage of the robustness of my 24bit signal when it goes
>through processing and track summing. If you leave 20db headroom you're only
>using about 20bits for your signal. 24bits will allow you to be lazy and leave
>a huge headroom and never worry about clipping and still come away with a
>decent recording, and I'd consider doing that if I had $5000+ a/d converters,
>but with my pro-sumer level ones, I'm looking to stretch as much quality out of
>them as possible, which means recording a signal as hot as possible

This is not a very good approach. The reason you want to have 24
bit resolution in the first place is so that you have that 20 dB of
headroom. Otherwise what's the sense?

This is not 1975 any more. You are not using some cheesy consumer
narrowtrack recorder. This is the digital age. There is no longer
any reason to obsess over pumping your levels as high as possible.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
August 29, 2005 10:54:38 PM

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

On 8/29/05 7:31 AM, in article
1125315114.583019.55530@o13g2000cwo.googlegroups.com, "Mike Rivers"
<mrivers@d-and-d.com> wrote:

> The concept of "headroom" in a digital system is different than in an
> analog system, because there's a hard limit.

I'll agree by disagreeing, more for conversational and educational clarity
than anythign else. (Mike does indeed clarify this real well in the next
lines but merely STARTING with this phrase gives me pause...)

The concept of headroom is NOT different, it is EXACTLY the same in analog
or digital systems. This is because they both DO INDEED have a hard limit.

Headroom is one of those damnably sloppily-used words that now are paying
us all back for ALLOWING them to be used sloppily for so manny decades. With
recording sound for entertainment purposes having finally dropped to the
same level as watercolor painting or home carpentry, everyman can and has
jumped into the job with little or no understanding of the simple basics of
what they;re doing. Like a sculptor who knows nothing about clay, kilns,
plaster formulations, metalurgy or carving granite, they;re lost amongst the
Cool Promises. Like the 1950's explosion of home power tools and
multi-function all-purpose home machinig/planer/lathe/whatsits ("You too Can
Turn Out Professional Master Carpentry At Home!") ernest-but-clueless folks
are wondering whay things don;t work when they just push the button.

When real engineers talked about headroom, they intuitively KNEW that it was
a relative term and were (hopefully) taking it in context in any particular
discussion... It meant a differnt KIND of thing when applied to a mixer or
other electronics vs a tape recorder (and by definition the TAPE medium
chosen for any particular job).... And ALL of that was temperes by the
defined-instrument-of-choice was the VU meter (or PPM if you preferred that)
which was DEFINED as having a certain real-level-above-nominal that you KNEW
about. As a classic example of 'A Little knowledge is indeed a dangerous
thing" we have real instantaneous-real-level meters and still have NOT
gotten over the idea that they are NOT VU meters. Otherwise the whole idea
of HEADROOJM would be an inherent understanding of what levels MEAN rather
than a simplistic 'traffic light' of whats 'OK' or 'BAD' for all purposes
all the time.


> The thing that you need to
> understand is that you can have as much or as little headroom as you
> want. Leaving 20 dB above the nomrmal level is considered good practice
> for a 24-bit system. This allows sufficient headroom for normal audio
> peaks. But if you've heavily limited your tracks so that there really
> are no peaks more than, say 6 dB above the nominal level, you can boost
> the track as much as 14 dB and still not have peaks that try to exceed
> full scale.

Unless of course it's a snare track or claves or a triangle with peaks way
up IN the 20dB range... But if you aren;t familiar with this, you don;t
know, can;t predicat dn so make ill-informed (ignorant) choices.

What you don;t know DOES hurt you.
Anonymous
August 29, 2005 11:03:01 PM

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

On 8/29/05 11:16 AM, in article
pan.2005.08.29.16.41.14.492615@localhost.com, "philicorda"
<philicorda@localhost.com> wrote:

>> There is no headroom. It goes to zero, then it clips.
>
> That's not true, on Cubase at least.

Of COURSe it's true.
0dBfs is -IT-, in cubase or anythign else....
Can;t go past there on a strip
Can't go past there in a buss
Can;t go past there on a mix
Can;t go past there any-which
Past that be dragons.

Now... what Cubase's designers CHOSE to make your -onscreen- levels ACTUALLY
mean inside the machine is something else... ie: If -0- on the screen is
ACTUALLY, say, -30dBfs inside out of 32bits, then indeed you can APPEAR to
be breaking the rule by 'pushing' things past 0, but it's a fake, a feelgood
recalibration of what -0- means. (HEADROOM!) In this case, there is Still
NOTHING above 0dBfs... But since what the onscreen meter TELLS you is
something called '0' isn;t, then you happily turn it up... And it WORKS...
Just like a console that sets meter-nominal -0- at say, -18 everywhere and
the noise floor is WAY down there (on a well designed hi-V rail pro level
console) can have 'endless headroom'. It's only because they make your
'operating level' arbitrarily way-below clipping so you can indeed push
things.
Anonymous
August 30, 2005 1:15:59 AM

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

On Mon, 29 Aug 2005 19:03:01 +0000, SSJVCmag wrote:

> On 8/29/05 11:16 AM, in article
> pan.2005.08.29.16.41.14.492615@localhost.com, "philicorda"
> <philicorda@localhost.com> wrote:
>
>>> There is no headroom. It goes to zero, then it clips.
>>
>> That's not true, on Cubase at least.
>
> Of COURSe it's true.
> 0dBfs is -IT-, in cubase or anythign else....
> Can;t go past there on a strip
> Can't go past there in a buss
> Can;t go past there on a mix
> Can;t go past there any-which
> Past that be dragons.
>
> Now... what Cubase's designers CHOSE to make your -onscreen- levels ACTUALLY
> mean inside the machine is something else... ie: If -0- on the screen is
> ACTUALLY, say, -30dBfs inside out of 32bits, then indeed you can APPEAR to
> be breaking the rule by 'pushing' things past 0, but it's a fake, a feelgood
> recalibration of what -0- means. (HEADROOM!) In this case, there is Still
> NOTHING above 0dBfs... But since what the onscreen meter TELLS you is
> something called '0' isn;t, then you happily turn it up... And it WORKS...
> Just like a console that sets meter-nominal -0- at say, -18 everywhere and
> the noise floor is WAY down there (on a well designed hi-V rail pro level
> console) can have 'endless headroom'. It's only because they make your
> 'operating level' arbitrarily way-below clipping so you can indeed push
> things.

Gotcha. So it's 0dBcubase or whatever internally. 0dBfs on the output
meter in Cubase really does mean 0dBfs as there is no headroom at all (try
to go over and it clips).

I found this great article about 32bit floating point and headroom in
Cubase that the OP may find interesting. I assume it will be equally true
for other 32bit float DAWS.

"Cubase VST uses a nominal operating level inside this floating point
range, such that there is a more than sufficient accuracy to represent the
finest detail BUT still have a massive head room. (Someone once calculated
1500 dB headroom - but I think that gives false impression)"

http://service.steinberg.net/knowledge_pro.nsf/0/7dd303...
Anonymous
August 30, 2005 3:10:51 AM

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

I'm mixing using Cubase SX with 4 x UAD-1 cards and RME and Mytek converters
for inserting analog processors as my processing engine for centerpanned
ambient, compression and EQ, then lightpiping 48 tracks of audio that is
being processed at 32 bit float to a timeline synced Paris system where
panning, panned ambient FX and additional EQ and other DSP based processing
is being applied. . All of this is automatically delay compensated in Cubase
SX prior to being flown across to the mix bus DAW.

This accomplishes a number of things-

First of all, I can get a reading on *actual* levels of tracks that are
being processed in Cubase SX on the Paris channel faders.

Secondly, the Paris system has a 52 bit fixed point summing engine which
uses the top and bottom two bits for digital *rounding* to enable a a sound
that is less reminiscent of digital clipping and more like *tape modelling*
for lack of a better analogy. Summing in Paris is a big, open sound with a
lot of sonic options as it has 5 onboard gain satges that interact
differently with the summing architecture depending on how they are pushed
and pulled. Combine this with a signal that is being lightpiped in to the
system while being processed hot at 32 bit (if desired) and you have a few
sonic options that sound pretty nice if not totally abused. Paris is a
dinosaur that is no longer being made, but combined with the features of a
modern DAW, it still has it's uses.....which, BTW, relate to your topic
concerning headroom in DAWs.

;o)


"TJ Hertz" <tjhertz@gmail.com> wrote in message
news:43126195$0$18643$14726298@news.sunsite.dk...
> Quick question...
>
> On modern recording software like Logic, Cubase, PT etc, what headroom can
I
> expect on each channel? In other words, assuming the source wav isn't
> clipped, can I (hypothetically... not that I'd want to) push the channel
> gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
> experience clipping?
>
> I'm asking because sometimes it's so much easier to boost a snare track
into
> +0 territory than to bring everything else down when you're running 30 or
> more tracks. Can I do this on a DAW (no external mixer) without worrying
> about clipping a track like this, provided the output buss is lowered to
the
> point where the levels don't clip?
>
> Thanks
>
> TJ
>
>
Anonymous
August 30, 2005 3:14:20 AM

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

On 8/28/05 11:55 PM, in article jHvQe.113343$fm.7517617@news4.tin.it, "ale"
<sparkwest@yahooc.om> wrote:

> yes, in cubase you can go over 0 dbfs on the individual channels without
> internal clipping the waveform.

Ummm isn;t this just a totally wrong use of the term " 0dBfs" ?

> ...in other sequencer softwares too.
> This because the software internally work at a greater resolution than
> the individual source files.

No, it's because WHATEVER resolution the software uses, it sets a HEADROOM
capability above what it CALLS nominal -0-
(damn if this aint JUST like gain-stage-designing an analog console ... aint
it!)


> But, you have to control the output section during the mixdown process
> to avoid clipping the output.
> In other words, if you have a normalized source (top at 0 dbfs) raised
> by about 6 db, you need to reduce the master setting by almost -6 db to
> avoid clipping.
> Usually the sequencer don't clip the waveform between stages.
>
Anonymous
August 30, 2005 3:24:34 AM

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

On 8/29/05 5:15 PM, in article pan.2005.08.29.22.43.44.2049@localhost.com,
"philicorda" <philicorda@localhost.com> wrote:
>>... since what the onscreen meter TELLS you is
>> something called '0' isn;t, then you happily turn it up... And it WORKS...
>> Just like a console that sets meter-nominal -0- at say, -18 everywhere and
>> the noise floor is WAY down there (on a well designed hi-V rail pro level
>> console) can have 'endless headroom'. It's only because they make your
>> 'operating level' arbitrarily way-below clipping so you can indeed push
>> things.
>
> Gotcha. So it's 0dBcubase or whatever internally.

'zackly.


> 0dBfs on the output
> meter in Cubase really does mean 0dBfs as there is no headroom at all (try
> to go over and it clips).

Maybe... I don;t know Cubase) Then again maybe that output is STILL got a
safety-range...
PARIS, for one, can actually be PUSHED a bit and whatevertheheck it DOES
with that... It SOUNDS good.
SOMEBODY programmed SOMETHING in there to act like an analog system because
it's HUMAN...
Whatta concept!
Anonymous
August 30, 2005 4:30:20 AM

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

"philicorda" <philicorda@localhost.com> wrote in message
news:p an.2005.08.29.22.43.44.2049@localhost.com...
> On Mon, 29 Aug 2005 19:03:01 +0000, SSJVCmag wrote:
>
> > On 8/29/05 11:16 AM, in article
> > pan.2005.08.29.16.41.14.492615@localhost.com, "philicorda"
> > <philicorda@localhost.com> wrote:
> >
> >>> There is no headroom. It goes to zero, then it clips.
> >>
> >> That's not true, on Cubase at least.
> >
> > Of COURSe it's true.
> > 0dBfs is -IT-, in cubase or anythign else....
> > Can;t go past there on a strip
> > Can't go past there in a buss
> > Can;t go past there on a mix
> > Can;t go past there any-which
> > Past that be dragons.
> >
> > Now... what Cubase's designers CHOSE to make your -onscreen- levels
ACTUALLY
> > mean inside the machine is something else... ie: If -0- on the screen is
> > ACTUALLY, say, -30dBfs inside out of 32bits, then indeed you can APPEAR
to
> > be breaking the rule by 'pushing' things past 0, but it's a fake, a
feelgood
> > recalibration of what -0- means. (HEADROOM!) In this case, there is
Still
> > NOTHING above 0dBfs... But since what the onscreen meter TELLS you is
> > something called '0' isn;t, then you happily turn it up... And it
WORKS...
> > Just like a console that sets meter-nominal -0- at say, -18 everywhere
and
> > the noise floor is WAY down there (on a well designed hi-V rail pro
level
> > console) can have 'endless headroom'. It's only because they make your
> > 'operating level' arbitrarily way-below clipping so you can indeed push
> > things.
>
> Gotcha. So it's 0dBcubase or whatever internally. 0dBfs on the output
> meter in Cubase really does mean 0dBfs as there is no headroom at all (try
> to go over and it clips).
>
> I found this great article about 32bit floating point and headroom in
> Cubase that the OP may find interesting. I assume it will be equally true
> for other 32bit float DAWS.
>
> "Cubase VST uses a nominal operating level inside this floating point
> range, such that there is a more than sufficient accuracy to represent the
> finest detail BUT still have a massive head room. (Someone once calculated
> 1500 dB headroom - but I think that gives false impression)"
>
>
http://service.steinberg.net/knowledge_pro.nsf/0/7dd303...
3968c?OpenDocument
>

Thanks,

This document tells me *exactly* what I needed to know:

"The massive headroom is what enables you to pile on tracks, and lower the
master fader when the summed signal cannot be represented anymore by the
INTEGER bits of your sound card. Lowering the master fader is changing the
scaling value as the floating point values are converted to integer bits for
the audio card. The advantage is that you can pile on the tracks and if the
FINAL output clips you can just turn down the master fader, and not be
forced to reduce each fader in tern until the clipping at the output
disappears."

Cheers

TJ
Anonymous
August 30, 2005 7:25:20 AM

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

You seem to be forgetting that 24bits do exist inside the DAW where the mixing
will be taking place. I'll stick with my hot-as-possible signals before
mixdown, you can stick to your disappointments about affordable converters and
microphones without much dynamic range affecting your desire to make the most
of what you've got...

In article <iv-dna9nddO6xo7eRVn-qA@comcast.com>, arnyk@hotpop.com says...
>
>"Chevdo" <Chevdo@chevdo.com> wrote in message
>news:3GIQe.158818$wr.58469@clgrps12
>
>> I wouldn't waste 20db headroom on any signal. I want to
>> record as hot as possible and take advantage of the
>> robustness of my 24bit signal when it goes through
>> processing and track summing.
>
>Fact is that in the real world of music recording, there are
>no actual 24 bit signals. With even the most expensive
>commercial DACs reproducing laboratory test signals, there
>*are* 20 bit signals. Just 20 bits. The other 4 bits are
>noise on a good day.
>
>Once you move out into the real world and hook up a
>microphone, there are signals with far more noise. The
>widest-range commercial recording I've ever analyzed had
>less than 75 dB dynamic range. That's less than 13 bits. If
>you huff and puff you might be able to make one track with
>85 dB dynamic range, but watch that go to San Diego in a
>handbasket when you start mixing it with a bunch of other
>tracks.
>
>> If you leave 20db headroom
>> you're only using about 20bits for your signal.
>
>Not a problem given that actual 20 bit signals only exist in
>the laboratory, and disappear instantly once you hook up a
>microphone, a console, etc.
>
>> 24bits
>> will allow you to be lazy and leave a huge headroom
>
>How about 10 bits of headroom?
>How about 15 bits of headroom?
>How about 20 bits of headroom?
>
>> never worry about clipping and still come away with a
>> decent recording, and I'd consider doing that if I had
>> $5000+ a/d converters, but with my pro-sumer level ones,
>> I'm looking to stretch as much quality out of them as
>> possible, which means recording a signal as hot as
>> possible
>
>The better prosumer converters have maybe 17 bits of
>resolution (102 dB or so). If you want to do the 20-bit
>boogey you are talking about some pretty fine stuff. If you
>want to dance with 24 bits, you're talking liquid nitrogen
>or helium or some such.
>
>
>
Anonymous
August 30, 2005 7:42:51 AM

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

In article <devo46$2ma$1@panix2.panix.com>, kludge@panix.com says...
>
>In article <3GIQe.158818$wr.58469@clgrps12>, Chevdo <Chevdo@chevdo.com> wrote:
>>I wouldn't waste 20db headroom on any signal. I want to record as hot as
>>possible and take advantage of the robustness of my 24bit signal when it goes
>>through processing and track summing. If you leave 20db headroom you're only
>>using about 20bits for your signal. 24bits will allow you to be lazy and
leave
>>a huge headroom and never worry about clipping and still come away with a
>>decent recording, and I'd consider doing that if I had $5000+ a/d converters,
>>but with my pro-sumer level ones, I'm looking to stretch as much quality out
of
>>them as possible, which means recording a signal as hot as possible
>
>This is not a very good approach. The reason you want to have 24
>bit resolution in the first place is so that you have that 20 dB of
>headroom. Otherwise what's the sense?

So that your final mix will be the highest resolution possible, particularly
if you're planning on squashing it with L3.

>
>This is not 1975 any more. You are not using some cheesy consumer
>narrowtrack recorder. This is the digital age. There is no longer
>any reason to obsess over pumping your levels as high as possible.

Nope still have to obsess because the goal now is to get as hot as possible in
the final mix with as little distortion as possible. I realize many engineers
are still kicking and screaming their way into the new paradigm but this is,
afterall, rec.audio.pro. If anyone here is producing recordings that sell in
the tens of thousands, or wants to, they're going to be part of the loudness
competition.
Anonymous
August 30, 2005 10:41:00 AM

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

"Chevdo" <chevr@doer.com> wrote in message
news:AkQQe.245240$on1.123327@clgrps13

> You seem to be forgetting that 24bits do exist inside the
> DAW where the mixing will be taking place.

No they don't.

Are you familiar with the concept of garbage-in,
garbage-out?

>I'll stick with my hot-as-possible signals before mixdown,
>you can
> stick to your disappointments about affordable converters
> and microphones without much dynamic range affecting your
> desire to make the most of what you've got...

What disappointments?
Anonymous
August 30, 2005 2:24:30 PM

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

In article <%AQQe.216669$tt5.37347@edtnps90>, Chevdo <chevdo@chev.com> wrote:
>>
>>This is not a very good approach. The reason you want to have 24
>>bit resolution in the first place is so that you have that 20 dB of
>>headroom. Otherwise what's the sense?
>
>So that your final mix will be the highest resolution possible, particularly
>if you're planning on squashing it with L3.

I think you are obsessing over "resolution" which is not actually what
you think it is. What you get by cranking the levels up is _solely_
a lower noise floor.

>>This is not 1975 any more. You are not using some cheesy consumer
>>narrowtrack recorder. This is the digital age. There is no longer
>>any reason to obsess over pumping your levels as high as possible.
>
>Nope still have to obsess because the goal now is to get as hot as possible in
>the final mix with as little distortion as possible. I realize many engineers
>are still kicking and screaming their way into the new paradigm but this is,
>afterall, rec.audio.pro. If anyone here is producing recordings that sell in
>the tens of thousands, or wants to, they're going to be part of the loudness
>competition.

This is not a new paradigm. Reducing headroom in the recording process
does _nothing_ to help you increase overall loudness anyway.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
August 30, 2005 5:00:03 PM

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

On 8/29/05 11:25 PM, in article AkQQe.245240$on1.123327@clgrps13, "Chevdo"
<chevr@doer.com> wrote:

>
> You seem to be forgetting that 24bits do exist inside the DAW where the mixing
> will be taking place.

Arny rarely forgets much, and moreso nearly NEVER 'seems' to forget anything
(to the annoyance of many). He assumes you have an attention span that'll go
more than 2 minutes and don;t need to be reminded of the context of the
discussion every time you type more than a sentence,


> I'll stick with my hot-as-possible signals before
> mixdown, you can stick to your disappointments about affordable converters and
> microphones without much dynamic range affecting your desire to make the most
> of what you've got...

Did I miss a whole thread-change here?
Anonymous
August 30, 2005 5:12:05 PM

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

On 8/29/05 11:42 PM, in article %AQQe.216669$tt5.37347@edtnps90, "Chevdo"
<chevdo@chev.com> wrote:

> In article <devo46$2ma$1@panix2.panix.com>, kludge@panix.com says...
>>
>> In article <3GIQe.158818$wr.58469@clgrps12>, Chevdo <Chevdo@chevdo.com>
>> wrote:
>>> I wouldn't waste 20db headroom on any signal. I want to record as hot as
>>> possible and take advantage of the robustness of my 24bit signal when it
>>> goes
>>> through processing and track summing. If you leave 20db headroom you're
>>> only
>>> using about 20bits for your signal. 24bits will allow you to be lazy and
> leave
>>> a huge headroom and never worry about clipping and still come away with a
>>> decent recording, and I'd consider doing that if I had $5000+ a/d
>>> converters,
>>> but with my pro-sumer level ones, I'm looking to stretch as much quality out
> of
>>> them as possible, which means recording a signal as hot as possible
>>
>> This is not a very good approach. The reason you want to have 24
>> bit resolution in the first place is so that you have that 20 dB of
>> headroom. Otherwise what's the sense?
>
> So that your final mix will be the highest resolution possible, particularly
> if you're planning on squashing it with L3.
>
>>
>> This is not 1975 any more. You are not using some cheesy consumer
>> narrowtrack recorder. This is the digital age. There is no longer
>> any reason to obsess over pumping your levels as high as possible.
>
> Nope still have to obsess because the goal now is to get as hot as possible in
> the final mix with as little distortion as possible.

Ahhh, I get it... You;re working clueless lazy-listener aesthetics aren;t
you! Shame. So many mixes coiuld be SO much better otherwise.
You'll grow out of it.

> I realize many engineers
> are still kicking and screaming their way into the new paradigm but this is,
> afterall, rec.audio.pro. If anyone here is producing recordings that sell in
> the tens of thousands, or wants to, they're going to be part of the loudness
> competition.
>

Au contraire, mixing to destructive loudness constraints is
Pointless
Uneccessary
Inexcusable in that it does a diservice to your client, whether that's the
producer, the artist or the label (and of course you're hip enough it's all
three) by
A- not leaving options in mastering for treatment appropriate for each and
any release medium.

B- Eliminating any chance of future hidef quality releases.

C- Destroying the sonics of the piece ireevocably that will make it sound
WORSE when it hits the audio chain of whatever media broadcast distributor
it has the luck of being shoved through for publicity display.

I realise many engineers are kicking and screaming their way blindly
thinking they're crossing into some new paradigm but this is after all
rec.audio.pro. If anyone here is producing recordings that sell, or wants
to, they're going to learn their basics and understand what the loudness
competition IS and triumph over it by making it work FOR them rather than
cutting their clients' noses off in spite of the damage done to their face.
Anonymous
August 30, 2005 5:13:11 PM

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

>>
>>> On 8/29/05 11:16 AM, in article
>>> pan.2005.08.29.16.41.14.492615@localhost.com, "philicorda"
>>> <philicorda@localhost.com> wrote:
>>>
>>>>> There is no headroom. It goes to zero, then it clips.
>>>>
>>>> That's not true, on Cubase at least.
>>>
>>> Of COURSe it's true.
>>> 0dBfs is -IT-, in cubase or anythign else....
>>> Can;t go past there on a strip
>>> Can't go past there in a buss
>>> Can;t go past there on a mix
>>> Can;t go past there any-which
>>> Past that be dragons.

Sorry, there IS headroom.

No, not past 0DBFS on a physical input or output concverter, but inside the
system (even the lamest shareware DAW app) internal stuff is not done at the
bit-depth of the signal. Internal stuff is done in areas of 32, 48, 56, 64
or whatever bits. Ya just got o get it back down to >0dBFS when you go to
the final bit-depth.

geoff
Anonymous
August 30, 2005 5:13:12 PM

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

On 8/29/05 9:13 PM, in article HoOQe.6977$iM2.705931@news.xtra.co.nz,
"Geoff@work" <gwood@nospam-audioproducts.co.nz> wrote:

>
>>>
>>>> On 8/29/05 11:16 AM, in article
>>>> pan.2005.08.29.16.41.14.492615@localhost.com, "philicorda"
>>>> <philicorda@localhost.com> wrote:
>>>>
>>>>>> There is no headroom. It goes to zero, then it clips.
>>>>>
>>>>> That's not true, on Cubase at least.
>>>>
>>>> Of COURSe it's true.
>>>> 0dBfs is -IT-, in cubase or anythign else....
>>>> Can;t go past there on a strip
>>>> Can't go past there in a buss
>>>> Can;t go past there on a mix
>>>> Can;t go past there any-which
>>>> Past that be dragons.
>
> Sorry, there IS headroom.

Sorry you didn;t read the rest of my post...
You would've found me saying pretty much the following:

>
> No, not past 0DBFS on a physical input or output concverter, but inside the
> system (even the lamest shareware DAW app) internal stuff is not done at the
> bit-depth of the signal. Internal stuff is done in areas of 32, 48, 56, 64
> or whatever bits. Ya just got o get it back down to >0dBFS when you go to
> the final bit-depth.
>
> geoff
>
>
and thus saved yourself some typing!
Anonymous
August 31, 2005 9:53:37 AM

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

Chevdo wrote:
> If the person says hey all
> these takes are costing me too much money, I'll tell them I could give them
> plenty of headroom so that clipping never happens, but I will also tell them
> that the quality of their recording will suffer as a result. Often when I
> tell them that it may be negligable they appear skeptical

You should ALWAYS tell them this, or better yet, put on your "good
engineer" hat, turn down the level so you don't have clipping, and just
move on. If anyone ever says "hey, that last take doesn't sound as good
as the other ones" send me the bill for the retake.

> Ultimately it's up to whoever is paying to decide how to record.

Well, I'll let them make the big decisions like whether to record to
tape or disk, or to use 48 or 96 kHz sample rate, but if I'm the
engineer, I'm going to make decisions about setting levels so that the
recording works well. I simply can't believe that you actually hear
what it is that you're preaching, unless you're doing other things to
get a hot level than just turning up the input gain until you're
running on the dangerous side of clipping.

> When I'm working on my own stuff I don't compromise, I
> always re-take clipped tracks

Better just to not make clipped tracks. Of course you're entitled to
your own style, but I think this discussion is important because
noobees shouldn't be psyched into believing that they have to record at
the highest possible level all the time, because they don't. They have
more important things to worry about and it's better to just set this
issue aside by using safe practices.

> No, that's completely false. If I record a track from an s612 with a hot
> level, I may place it very low in the mix, so suddenly the mix has greater
> dynamic range than the s612, because I am mixing tracks recorded at 24bit.

That's not what dynamic range is. Mixing a track in at a low level
doesn't increase the dynamic range. Having a solo at a low level
increases the dynamic range. But adding your low level S612 to
something else that's playing louder doesn't make for more dynamic
range, it changes the tone color.

> Wow, you're learning something then, I guess?? Efficiency in this context
> refers to how often the converter converts a bit accurately.

Huh?

> They're always most accurate at the top.

Huh? Do you understand digital sampling at all?

> I have a digital scale that, like all digital
> scales, is more accurate when you put a load on it first then 'tare' it to
> zero before adding what you want to weigh.

I have stuff that doesn't work right, too. But if I really cared about
this, I'd get a better scale. My digital scale is good enough for
measuring things in the kitchen when I'm cooking, or for determining if
I need to put another postage stamp on an envelope. If I was selling
coke by the gram, I'd consider a really top grade (and accurate over
its full range) scale to be a worthwhile business investment.

Get rid of your AdLib sound card and you'll be surprised at how good a
modern 24-bit converter can sound, even when peaks don't exceed -10
dBFS.

> I suspect you are ignorant about how converters work. I'm curious as to who
> started this myth and how it perpetuated, though. Is it relegated to this
> group? Because I've never heard it before.

The "myth" is that converters only work right when they're at or near
full scale. This was true for the converters we had 25 years ago, but
it is no longer true today. It's not a myth.

> The availablility of 24bits simply reduced the
> importance, but only because we are still mixing down to a 16bit CD product.
> What are you going to do when we're mixing to 24bit in a new consumer format?
> Stick with your low levels and revel in your 'headroom'.

Produce recordings with greater dynamic range. But people won't be able
to listen to them, so it'll all get "fixed" in mastering anyway.

> Even jazz and folk bands want a loud recording these days. Your only
> sanctuary, for now, is classical. And tests have shown it's gotten louder
> and more compressed over the years.

Sadly, this is true. But any quality that might be retained in the
tracks gets lost in the process of making the final mix loud. I
understand this.
Anonymous
August 31, 2005 2:29:51 PM

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

In article <ebCdnWQqvKYgqoneRVn-vg@comcast.com>, arnyk@hotpop.com says...
>
>"Chevdo" <chevr@doer.com> wrote in message
>news:AkQQe.245240$on1.123327@clgrps13
>
>> You seem to be forgetting that 24bits do exist inside the
>> DAW where the mixing will be taking place.
>
>No they don't.
>
>Are you familiar with the concept of garbage-in,
>garbage-out?

uh huh, that's exactly my point. Get the best quality signal in, and have the
greater likelihood of getting the best quality signal out.

Are you suggesting that the quality of a 24bit signal recorded with low levels
would actually be higher than the quality of a 24bit signal recorded with
higher levels? I think your attempt at arguing that the difference would be
negligiable had more legs, but I think you've demonstrated that as usual,
lower-quality arguers tend to find it hard to resist attempting to strengthen
their arguments by creating dualities.


>
>>I'll stick with my hot-as-possible signals before mixdown,
>>you can
>> stick to your disappointments about affordable converters
>> and microphones without much dynamic range affecting your
>> desire to make the most of what you've got...
>
>What disappointments?
>
>

The ones you mentioned, which I just repeated. Dynamic range, noise floors,
etc. The 'real world' and all its less-than-perfect signals.
Anonymous
August 31, 2005 2:29:52 PM

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

"Chevdo" <chevdo@chevdo.com> wrote in message
news:zEfRe.251913$on1.216006@clgrps13
> In article <ebCdnWQqvKYgqoneRVn-vg@comcast.com>,
> arnyk@hotpop.com says...
>>
>> "Chevdo" <chevr@doer.com> wrote in message
>> news:AkQQe.245240$on1.123327@clgrps13
>>
>>> You seem to be forgetting that 24bits do exist inside
>>> the DAW where the mixing will be taking place.
>>
>> No they don't.
>>
>> Are you familiar with the concept of garbage-in,
>> garbage-out?
>
> uh huh, that's exactly my point. Get the best quality
> signal in, and have the greater likelihood of getting the
> best quality signal out.

I'm talking about recording as a determinstic process, not
rolling dice. So, liklihoods don't count.

> Are you suggesting that the quality of a 24bit signal
> recorded with low levels would actually be higher than
> the quality of a 24bit signal recorded with higher
> levels?

In the real world, that can happen. In an ideal world
recording a noisy signal is not impacted by a noisy channel
until the channel noise is just a few dB below the signal
noise.

Recording real-world audio is about something that by 24 bit
standards, is a very noisy signal.

> I think your attempt at arguing that the
> difference would be negligiable had more legs, but I
> think you've demonstrated that as usual, lower-quality
> arguers tend to find it hard to resist attempting to
> strengthen their arguments by creating dualities.

Well that's a mouthful and a bit impressive-sounding, but
let's look at a real world situation:

If I have an audio signal with a noise floor that is 75 dB
down (one heck of a clean signal!) and record it with a
noisy digital channel whose similar but lower noise floor is
85 dB down (very mediocre performance by modern standards)
the signal's noise floor is raised by less than 0.1 dB
(totally trivial) by the so-called noisy digital channel.

I conclude that if the digital channel's noise floor is 10
dB better than that of the signal, then its effects can be
safely ignored.

In the real world, I have a DAW with digital channels that
have a noise floor that is 98 dB down (average performance
for modern digital production gear). My real world signals
(performer's mics) have a measured dynamic range (from mic
on, room quiet to performer singing vigorously) of about 70
dB. I conclude that I can record with 18 dB of headroom
while sacrificing only trivial amounts (<0.1dB) of the live
sound's dynamic range.

>>> I'll stick with my hot-as-possible signals before
>>> mixdown, you can
>>> stick to your disappointments about affordable
>>> converters and microphones without much dynamic range
>>> affecting your desire to make the most of what you've
>>> got...
>>
>> What disappointments?

> The ones you mentioned, which I just repeated. Dynamic
> range, noise floors, etc. The 'real world' and all its
> less-than-perfect signals.

See my examples.
Anonymous
August 31, 2005 2:50:06 PM

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

In article <BF39D091.FF0F%ten@nozirev.gamnocssj.com>, ten@nozirev.gamnocssj.com
says...
>
>On 8/29/05 11:25 PM, in article AkQQe.245240$on1.123327@clgrps13, "Chevdo"
><chevr@doer.com> wrote:
>
>>
>> You seem to be forgetting that 24bits do exist inside the DAW where the
mixing
>> will be taking place.
>
>Arny rarely forgets much, and moreso nearly NEVER 'seems' to forget anything
>(to the annoyance of many). He assumes you have an attention span that'll go
>more than 2 minutes and don;t need to be reminded of the context of the
>discussion every time you type more than a sentence,
>

What are you, Arny's cheering squad? If Arny's memory is good, he should
be able to present his arguments in a more organized fashion than he tends to.
People who have more knowledge than sense, Arny in particular, are why I
rarely venture into this group at all. As has been pointed out by others in
this thread, Arny contradicts his own claims with inadequate explaination. In
my opinion people do this all over usenet these days, as every group has its
guru, or handful of gurus who back each other up in a social context. Private
forums stifle the challenging of resident gurus even more effectively,
hindering progress in some ways, and accelerating it in others (afterall,
there are plenty of morons who want to share their ability to waste time with
more important people). I come here occasionally merely to have a question
answered, and the wannabe know-it-alls can always be counted on for an answer.
I poked my head into this thread just to have a little fun rebuking the false
claims. Next week the same subject will come up and somebody 'important' to
this newsgroup will make the opposite claim as the consensus determined in this
one, and the consensus will end up determining that claim to be correct.

I still like the fact that people provide opinions and occasionally facts,
which are sometimes right, when I ask a question here. I always appreciate
Arny's opinion, because he is highly knowledgable, but I am usually able to
recognize when his personal prejudice or over-confidence in his knowledge
reduces the accuracy, and thus, usefulness of his responses.

>
>> I'll stick with my hot-as-possible signals before
>> mixdown, you can stick to your disappointments about affordable converters
and
>> microphones without much dynamic range affecting your desire to make the
most
>> of what you've got...
>
>Did I miss a whole thread-change here?
>
>

You seem to be trying to make a whole thread-change here to make it all about
Arny's personal defects. Personally I think he deserves more respect than
that. I prefer to deal with his tendencies to slide into BS more tactfully.
Anonymous
August 31, 2005 2:50:07 PM

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

"Chevdo" <chevdo@chdv.com> wrote in message
news:yXfRe.220691$HI.167920@edtnps84
> In article <BF39D091.FF0F%ten@nozirev.gamnocssj.com>,
> ten@nozirev.gamnocssj.com says...
>>
>> On 8/29/05 11:25 PM, in article
>> AkQQe.245240$on1.123327@clgrps13, "Chevdo"
>> <chevr@doer.com> wrote:
>>
>>>
>>> You seem to be forgetting that 24bits do exist inside
>>> the DAW where the mixing will be taking place.
>>
>> Arny rarely forgets much, and moreso nearly NEVER
>> 'seems' to forget anything (to the annoyance of many).
>> He assumes you have an attention span that'll go more
>> than 2 minutes and don;t need to be reminded of the
>> context of the discussion every time you type more than
>> a sentence,

> What are you, Arny's cheering squad?

We get along. ;-)

> If Arny's memory is
> good, he should
> be able to present his arguments in a more organized
> fashion than he tends to.

I try. I even get occasional accolades for my presentation
skills. By most accounts (even Stereophile writers) I kicked
Atkinson's ass at the HE2005 debate, and he's the editor of
Stereophile. Seems like his presentation skills should be
pretty good.

Here's a little piece of wisdom - its hard to seem like a
good presenter to people who are diametrically opposed to
what you are presenting. Not impossible, but often
difficult. When people are praised for the quality of their
presentation by people who don't agree with them, the
presentation was usually pretty spectacular - perhaps even
one that put form before function.

> People who have more knowledge
> than sense, Arny in particular, are why I rarely venture
> into this group at all.

Hey, I make my recordings, I do my live sound, I build a few
things, I write a few posts, everything seems to work pretty
well, what do I need to have to do to make you happy? ;-)

> As has been pointed out by
> others in this thread, Arny contradicts his own claims
> with inadequate explaination.

Easy to say, but AFAIK a totally unproven claim.

> In my opinion people do
> this all over usenet these days, as every group has its
> guru, or handful of gurus who back each other up in a
> social context. Private forums stifle the challenging of
> resident gurus even more effectively, hindering progress
> in some ways, and accelerating it in others (afterall,
> there are plenty of morons who want to share their
> ability to waste time with more important people). I
> come here occasionally merely to have a question
> answered, and the wannabe know-it-alls can always be
> counted on for an answer. I poked my head into this
> thread just to have a little fun rebuking the false
> claims. Next week the same subject will come up and
> somebody 'important' to this newsgroup will make the
> opposite claim as the consensus determined in this one,
> and the consensus will end up determining that claim to
> be correct.

So challenge me with some real world facts. Got any
practical examples to back your whining up with?

> I still like the fact that people provide opinions and
> occasionally facts, which are sometimes right, when I ask
> a question here. I always appreciate Arny's opinion,
> because he is highly knowledgable, but I am usually able
> to recognize when his personal prejudice or
> over-confidence in his knowledge reduces the accuracy,
> and thus, usefulness of his responses.

So where is the inaccuracy?

>>> I'll stick with my hot-as-possible signals before
>>> mixdown, you can stick to your disappointments about
>>> affordable converters and microphones without much
>>> dynamic range affecting your desire to make the most of
>>> what you've got...
>>
>> Did I miss a whole thread-change here?

> You seem to be trying to make a whole thread-change here
> to make it all about Arny's personal defects. Personally
> I think he deserves more respect than that. I prefer to
> deal with his tendencies to slide into BS more tactfully.

Where's this alleged BS?

I see a lot of baseless charges.
Anonymous
August 31, 2005 3:19:41 PM

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

In article <BF39D363.FF11%ten@nozirev.gamnocssj.com>, ten@nozirev.gamnocssj.com
says...
>
>
>>> This is not 1975 any more. You are not using some cheesy consumer
>>> narrowtrack recorder. This is the digital age. There is no longer
>>> any reason to obsess over pumping your levels as high as possible.
>>
>> Nope still have to obsess because the goal now is to get as hot as possible
in
>> the final mix with as little distortion as possible.
>
>Ahhh, I get it... You;re working clueless lazy-listener aesthetics aren;t
>you! Shame. So many mixes coiuld be SO much better otherwise.
>You'll grow out of it.

Seems like a rather poor ad hominem, but if you think you made a good point,
that's great for you!


>
>> I realize many engineers
>> are still kicking and screaming their way into the new paradigm but this is,
>> afterall, rec.audio.pro. If anyone here is producing recordings that sell
in
>> the tens of thousands, or wants to, they're going to be part of the loudness
>> competition.
>>
>
>Au contraire, mixing to destructive loudness constraints is
>Pointless
>Uneccessary
>Inexcusable

Indeed, which is why I make sure I get the best quality signals on the tracks
so that the final mix that will be squashed as much as possible without
unacceptable destructiveness to the signal. Destructiveness, in this context,
refers to the perception of the average listener.

Your fallacy was to misrepresent my argument by suggesting that I advocate
destroying signals when I've clearly been advocating all a long to get the best
signal possible, before the destructiveness that is required to make the
product marketable is applied, so that that destructiveness is minimized. You
can babble all you want about me doing a 'disservice to a client', but I
obviously believe I am doing exactly the opposite by using the methods I do.
I've also explained why, and I think my argument is the better one.


>A- not leaving options in mastering for treatment appropriate for each and
>any release medium.
>

I don't do that, and haven't said anything that would suggest I would. Just ad
hominem nonsense. If someone wants to pay me to track their recordings with
low levels, I have absolutely no problem accepting their business and doing
things the way they want. I will, however, tell them my recommendations and
the reasoning behind them. If they choose not to use my recommendations I
don't take it personally as an assault on my ego, intentional or not, as I
suspect 95% of the people reading this would, and do. This may explain why new
client invariably tell me how relieved they are to work with me, and that my
style and product is better than anything they've ever purchased before. On
the other hand, to get any work at all I have to be severely underpaid, and I
figure every engineer and producer gets smoke blown up their assess by every
client with half-a-brain. And again, I'll bet 95% of the readers of this post
who consider themselves engineers or producers let that smoke go right to their
brains, in the ego department.


>B- Eliminating any chance of future hidef quality releases.

Yeah thats it, I delete the session rather than saving it to cheap DVD media,
and filing it away in the studio archives, because I'm as dumb as you
apparently are.


>
>C- Destroying the sonics of the piece ireevocably that will make it sound
>WORSE when it hits the audio chain of whatever media broadcast distributor
>it has the luck of being shoved through for publicity display.

Preach on, oh religiously-devoted virtuous one! Just like a preacher, trying
to convince me to sabotage my intellect, you seem to want to convince me
to sabotage my career. Publicity? Yikes, wouldn't want
that. Might get sales. Might become too professional to post on this group.
Might have already happened... naaaah. Still too many dinosaurs hunkered down
at the big, long, penile-extension console for me to break-through. But onna
these days, Alice, onna these days...pow zoom, to the moon!




>
>I realise many engineers are kicking and screaming their way blindly
>thinking they're crossing into some new paradigm but this is after all
>rec.audio.pro. If anyone here is producing recordings that sell, or wants
>to, they're going to learn their basics and understand what the loudness
>competition IS and triumph over it by making it work FOR them rather than
>cutting their clients' noses off in spite of the damage done to their face.
>

Yeah and I feel particularly suited to it, but I think you've got some sacred
cows to slaughter before you'll get there. Good luck!
Anonymous
August 31, 2005 3:35:43 PM

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

In article <df1q6u$d0k$1@panix2.panix.com>, kludge@panix.com says...
>
>In article <%AQQe.216669$tt5.37347@edtnps90>, Chevdo <chevdo@chev.com> wrote:
>>>
>>>This is not a very good approach. The reason you want to have 24
>>>bit resolution in the first place is so that you have that 20 dB of
>>>headroom. Otherwise what's the sense?
>>
>>So that your final mix will be the highest resolution possible, particularly
>>if you're planning on squashing it with L3.
>
>I think you are obsessing over "resolution" which is not actually what
>you think it is. What you get by cranking the levels up is _solely_
>a lower noise floor.

Maybe you meant higher noise floor? Lower noise floor sounds desirable.

I think the people who are obsessing over resolution are those who would like
to be able to afford more and better, but since they can't, they are
cognitively conflicted. Hey, my board doesn't even go over 44.1khz sample
rate. I'd like it to go to 96khz, but I don't lose any sleep over it. I work
hard to get the best results I can out of it, and I'm sorry if you find it hard
to believe I'm able to do that using the methods that I have described here. I
recommend you do whatever you think works best for you, since that's what I do.
If my arguments don't seem compelling to you, it's no sweat off my back and it
should be no sweat off yours if your arguments don't seem compelling to me.

>
>>>This is not 1975 any more. You are not using some cheesy consumer
>>>narrowtrack recorder. This is the digital age. There is no longer
>>>any reason to obsess over pumping your levels as high as possible.
>>
>>Nope still have to obsess because the goal now is to get as hot as possible
in
>>the final mix with as little distortion as possible. I realize many
engineers
>>are still kicking and screaming their way into the new paradigm but this is,
>>afterall, rec.audio.pro. If anyone here is producing recordings that sell in
>>the tens of thousands, or wants to, they're going to be part of the loudness
>>competition.
>
>This is not a new paradigm.

Which is why it's so embarassing to see so many people still resisting it
around these parts... you can't ALL be recording orchestras playing classical
music.

>Reducing headroom in the recording process
>does _nothing_ to help you increase overall loudness anyway.

Yes it does, it allows you to later. after mixdown of all your tracks, squash
the signal with less distortion, therefore you can squash it a bit more before
distortion becomes audibly unacceptable. I'm not advocating mastering to 0db,
though, I prefer to master to -1db in order to avoid playback clipping on
inferior DACs in cheap CD players.
Anonymous
August 31, 2005 3:35:44 PM

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

In article <jCgRe.251917$on1.89551@clgrps13>, Chevdo <chevdo@chev.com> wrote:
>In article <df1q6u$d0k$1@panix2.panix.com>, kludge@panix.com says...
>>
>>In article <%AQQe.216669$tt5.37347@edtnps90>, Chevdo <chevdo@chev.com> wrote:
>>>>
>>>>This is not a very good approach. The reason you want to have 24
>>>>bit resolution in the first place is so that you have that 20 dB of
>>>>headroom. Otherwise what's the sense?
>>>
>>>So that your final mix will be the highest resolution possible, particularly
>>>if you're planning on squashing it with L3.
>>
>>I think you are obsessing over "resolution" which is not actually what
>>you think it is. What you get by cranking the levels up is _solely_
>>a lower noise floor.
>
>Maybe you meant higher noise floor? Lower noise floor sounds desirable.

It is. But it's the _only_ thing you get from cranking the levels up.
And given the sheer amount of available dynamic range, it's not that
big a deal. Back when a narrowtrack channel might have 50 dB of usable
range, that was a huge deal. Today it's not.

> If my arguments don't seem compelling to you, it's no sweat off my back and it
>should be no sweat off yours if your arguments don't seem compelling to me.

You might want to read the following paper:

Resolution Below the Least Significant Bit in Digital Systems with Dither
by Vanderkooy, John; Lipshitz, Stanley P. AES preprint 1930, from the
September 1982 convention.

Abstract:
This paper is motivated by some common misunderstandings about digital
systems. It is commonly believed that small signals or signal details
are lost if they are smaller than the quantizing step. We expand on
previous arguments showing that this is not true when the signal to be
quantized contains a wideband noise dither of amplitude approximately
the step size. The introduction traces the use of dither from video
quantization through its use in audio. Quantization error is studied
in some detail and the effort of dither is analyzed theoretically
and experimentally. We argue and show by examples of quantized signals
that the dither effectively turns signal distortion into low-level
wideband noise by linearizing the averaged quantizer staircase function,
which is as perceived by the ear.

Lipshitz does an excellent job of explaining how dither linearizes the
digital system, so that the "resolution" becomes independant of level.
The paper is old and doesn't talk about some of the newer noise shaping
technology, but the description is good and he goes through the math
about how it works.

>>Reducing headroom in the recording process
>>does _nothing_ to help you increase overall loudness anyway.
>
>Yes it does, it allows you to later. after mixdown of all your tracks, squash
>the signal with less distortion, therefore you can squash it a bit more before
>distortion becomes audibly unacceptable. I'm not advocating mastering to 0db,
>though, I prefer to master to -1db in order to avoid playback clipping on
>inferior DACs in cheap CD players.

No. Read Lipshitz's paper. And/or try it for yourself... record something
at -80 dB and then at 0 dB and listen to the two. If your converters are
good and your dither configuration is good, you won't hear any increased
distortion, just increased noise, on the low level track.

Again, this was a huge deal back in the early eighties when dither was not
very well understood by designers and a lot of converters had dead band
issues. But those days are over.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
August 31, 2005 3:35:44 PM

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

"Chevdo" <chevdo@chev.com> wrote in message
news:jCgRe.251917$on1.89551@clgrps13

>> Reducing headroom in the recording process
>> does _nothing_ to help you increase overall loudness
>> anyway.

> Yes it does, it allows you to later. after mixdown of all
> your tracks, squash the signal with less distortion,
> therefore you can squash it a bit more before distortion
> becomes audibly unacceptable.

Only in someone's dreams.

Digital done right is a distortionless medium. Theoretically
and practically zero-point-zero distortion of any kind. Any
linear or nonlinear distortion that you find in a good
digital audio system is either in the analog interfaces or
was put there intentionally or at least in some sense
knowingly.

Therefore contamination of the digital signal is all about
noise. Any kind of significant processing in the digital
domain will increase noise in the form of numerical errors.

As far as tracking versus mixdown goes, in modern recording
systems, the major source of noise is outside the recording
equipment. It's in the rooms and the mics. As long as the
signal coming out of the mic is handled cleanly, it won't
limit what you can do in mixdown or mastering.

In another post I irrefutably show that good modern
production equipment of average quality has about 18 dB
worth of headroom. If you want to, you can call it foot
room, and use it as such. It's excess dynamic range. My
example was based on a 16 bit system.
Anonymous
August 31, 2005 6:34:22 PM

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

On 8/31/05 6:50 AM, in article yXfRe.220691$HI.167920@edtnps84, "Chevdo"
<chevdo@chdv.com> wrote:

> In article <BF39D091.FF0F%ten@nozirev.gamnocssj.com>,
> ten@nozirev.gamnocssj.com
> says...
>>
>> On 8/29/05 11:25 PM, in article AkQQe.245240$on1.123327@clgrps13, "Chevdo"
>> <chevr@doer.com> wrote:
>>
>>>
>>> You seem to be forgetting that 24bits do exist inside the DAW where the
> mixing
>>> will be taking place.
>>
>> Arny rarely forgets much, and moreso nearly NEVER 'seems' to forget anything
>> (to the annoyance of many). He assumes you have an attention span that'll go
>> more than 2 minutes and don;t need to be reminded of the context of the
>> discussion every time you type more than a sentence,
>>
>
> What are you, Arny's cheering squad?

One little sorta kinda "Arny's Maybe Not -quite- the Idiot You Paint him"
and you get your bloomers in a bunch?

> If Arny's memory is good, he should
> be able to present his arguments in a more organized fashion than he tends to.

I'm sorry.. I'm working with the Arny in THIS universe... The only folks
that really get bent out of shape with the way Arny gets his points across
are those who think fools should be tolerated.


> People who have more knowledge than sense, Arny in particular, are why I
> rarely venture into this group at all. As has been pointed out by others in
> this thread, Arny contradicts his own
(SNIP)
Ah...
gotcha
Thanks for playing.
Pick up Your Very Own Copy of our home game on the way out!
Anonymous
September 1, 2005 1:14:30 AM

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

In article <df49ao$ba7$1@panix2.panix.com>, kludge@panix.com says...
>
>You might want to read the following paper:
>
>Resolution Below the Least Significant Bit in Digital Systems with Dither
>by Vanderkooy, John; Lipshitz, Stanley P. AES preprint 1930, from the
>September 1982 convention.
>
>Abstract:
>This paper is motivated by some common misunderstandings about digital
>systems. It is commonly believed that small signals or signal details
>are lost if they are smaller than the quantizing step. We expand on
>previous arguments showing that this is not true when the signal to be
>quantized contains a wideband noise dither of amplitude approximately
>the step size. The introduction traces the use of dither from video
>quantization through its use in audio. Quantization error is studied
>in some detail and the effort of dither is analyzed theoretically
>and experimentally. We argue and show by examples of quantized signals
>that the dither effectively turns signal distortion into low-level
>wideband noise by linearizing the averaged quantizer staircase function,
>which is as perceived by the ear.
>
>Lipshitz does an excellent job of explaining how dither linearizes the
>digital system, so that the "resolution" becomes independant of level.
>The paper is old and doesn't talk about some of the newer noise shaping
>technology, but the description is good and he goes through the math
>about how it works.

That's nice, I like dither. But can you point to anywhere in the paper where
it says that not recording hot levels on a digital system is a good idea? The
paragraph you quoted was about an apparent misunderstanding about signals being
lost if they are smaller than the quantizing step. That refers to signal
information that falls 'between' the bit and sample rate and doesn't seem to
have anything to do with what I've been talking about.


>No. Read Lipshitz's paper. And/or try it for yourself... record something
>at -80 dB and then at 0 dB and listen to the two. If your converters are
>good and your dither configuration is good, you won't hear any increased
>distortion, just increased noise, on the low level track.

You're not reading my argument, hopefully not deliberately. I was referring to
master mixdowns from DAWs after they have been squashed by
compressor/limiters. I certainly would end up with an inferior product if I
were to record every track at -80db before mixing, and normalizing the signal
way up in order to get squashed by the master limiting. I hope Lipshitz would
agree, but if not, I'd like to see his argument about that since I haven't seen
it yet.


>
>Again, this was a huge deal back in the early eighties when dither was not
>very well understood by designers and a lot of converters had dead band
>issues. But those days are over.

It seems that the days we're in now are filled with even more digitally
confused engineers. Which is why I prefer to lurk and why you likely won't see
me here again for quite some time.
Anonymous
September 1, 2005 1:38:38 AM

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

On Mon, 29 Aug 2005 03:26:38 +0100, "TJ Hertz" <tjhertz@gmail.com>
wrote:

>I'm not sure you understood my question - when I record, obviously I ensure
>no clipping occurs; but I'm only referring to the mixdown stage and whether
>or not individual channels can then go above 0 without clipping PROVIDED the
>output buss is turned down to the point where it never exceeds 0.

Programs such as Cubase mix in a "32-bit floating point environment"
Whatever this technically means, the practical result is that it's
practically impossible to overload within the mixer channels.
Anonymous
September 1, 2005 1:40:00 AM

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

On 29 Aug 2005 09:43:37 -0400, kludge@panix.com (Scott Dorsey) wrote:

>>On modern recording software like Logic, Cubase, PT etc, what headroom can I
>>expect on each channel? In other words, assuming the source wav isn't
>>clipped, can I (hypothetically... not that I'd want to) push the channel
>>gain to 10dB over 0 then reduce it by -10dB at the output fader, and not
>>experience clipping?
>
>There is no headroom. It goes to zero, then it clips.

I suspect you have misread the question.
Anonymous
September 1, 2005 1:43:14 AM

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

In article <tPGdnRe3DqqoMYjeRVn-gQ@comcast.com>, arnyk@hotpop.com says...
>

>Recording real-world audio is about something that by 24 bit
>standards, is a very noisy signal.

Do sentences like this mean anything? These kinds of statements are your
downfall, Arny.

>If I have an audio signal with a noise floor that is 75 dB
>down (one heck of a clean signal!) and record it with a
>noisy digital channel whose similar but lower noise floor is
>85 dB down (very mediocre performance by modern standards)
>the signal's noise floor is raised by less than 0.1 dB
>(totally trivial) by the so-called noisy digital channel.
>
>I conclude that if the digital channel's noise floor is 10
>dB better than that of the signal, then its effects can be
>safely ignored.

Yeah that sounds right, but does not seem to be addressing what I've been
talking about at all regarding recording digital signals as hot as possible.


>
>In the real world, I have a DAW with digital channels that
>have a noise floor that is 98 dB down (average performance
>for modern digital production gear). My real world signals
>(performer's mics) have a measured dynamic range (from mic
>on, room quiet to performer singing vigorously) of about 70
>dB. I conclude that I can record with 18 dB of headroom
>while sacrificing only trivial amounts (<0.1dB) of the live
>sound's dynamic range.

I'm sure you'll do fine with that, and maybe you are sacrificing only trivial
amounts of signal, but it's hard to say since the 'dynamic range' spec of
analog equipment only refers to the region of power output in which the signal
is at it's best, it does not refer to any kind of cut-off as it does in the
digital world as bit resolution. With my huge 24bit resolution I can capture
not only the dynamic range of my equipment, but whatever signal is beyond spec,
some of which might end up being very acceptable, depending on how modest the
manufacturer is with his specs. Reducing the dynamic range of my recording
device won't reduce the dynamic range of what I'm recording, though, unless I
reduce the bit resolution drastically. Once the recording is squashed the
least significant bits will be dither anyway. But in the meantime I want even
the under-spec signals to go through all my reverbs, delays, eqs, and
every other wacky plugin I plugin and mix. They can only add
to my resolution, not detract from it.


Yours is a better argument than the others have presented, though.
Anonymous
September 1, 2005 3:31:19 AM

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

On Wed, 31 Aug 2005 10:50:06 GMT, chevdo@chdv.com (Chevdo) wrote:

> If Arny's memory is good, he should
>be able to present his arguments in a more organized fashion than he tends to.

Now, *that's* funny.

> Next week the same subject will come up and somebody 'important' to
>this newsgroup will make the opposite claim as the consensus determined in this
>one, and the consensus will end up determining that claim to be correct.

Not in this case, Mon. Really.

Good fortune,

Chris Hornbeck
Anonymous
September 1, 2005 4:06:57 AM

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

Chevdo wrote:

>>Reducing headroom in the recording process
>>does _nothing_ to help you increase overall loudness anyway.
>
>
> Yes it does, it allows you to later. after mixdown of all your tracks, squash
> the signal with less distortion, therefore you can squash it a bit more before
> distortion becomes audibly unacceptable.

Can you explain, technically and without waving of hands,
why this would be true? As an EE and DSP specialist I can
see no reason why higher recording levels can affect the
distortion from dynamic compression after all is mixed.

It can lower quantization noise but that's all, and if
working with (truly) wider converters that becomes less and
less relevant. All of which has been pointed out by others.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein
Anonymous
September 1, 2005 7:53:49 AM

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

"Chevdo" <chevdo@chevdont.com> wrote in message
news:SvpRe.225972$tt5.38080@edtnps90
> In article <tPGdnRe3DqqoMYjeRVn-gQ@comcast.com>,
> arnyk@hotpop.com says...
>>
>
>> Recording real-world audio is about something that by 24
>> bit standards, is a very noisy signal.
>
> Do sentences like this mean anything?

Another reader seems to think so.

> These kinds of
> statements are your downfall, Arny.

Not my downfall, yours.

>> If I have an audio signal with a noise floor that is 75
>> dB down (one heck of a clean signal!) and record it with
>> a noisy digital channel whose similar but lower noise
>> floor is 85 dB down (very mediocre performance by modern
>> standards) the signal's noise floor is raised by less
>> than 0.1 dB (totally trivial) by the so-called noisy
>> digital channel.
>>
>> I conclude that if the digital channel's noise floor is
>> 10 dB better than that of the signal, then its effects
>> can be safely ignored.
>
> Yeah that sounds right, but does not seem to be
> addressing what I've been talking about at all regarding
> recording digital signals as hot as possible.

That was the prepartory example, not the final explanation.

>> In the real world, I have a DAW with digital channels
>> that have a noise floor that is 98 dB down (average
>> performance for modern digital production gear). My
>> real world signals (performer's mics) have a measured
>> dynamic range (from mic on, room quiet to performer
>> singing vigorously) of about 70 dB. I conclude that I
>> can record with 18 dB of headroom while sacrificing only
>> trivial amounts (<0.1dB) of the live sound's dynamic
>> range.

>I'm sure you'll do fine with that, and maybe you are
> sacrificing only trivial amounts of signal,



> but it's hard
> to say since the 'dynamic range' spec of analog equipment
> only refers to the region of power output in which the
> signal is at it's best,

The normal audio range is 20-20K whether the equipment is
analog or digital. A-weighting is the same for analog or
digital equipment.

> t does not refer to any kind of
> cut-off as it does in the digital world as bit
> resolution.

Sure it does. Any noise measurement, whether of analog or
digtital equipment, is meaningless without specifying the
measurement bandwidth.


> With my huge 24bit resolution I can capture
> not only the dynamic range of my equipment, but whatever
> signal is beyond spec, some of which might end up being
> very acceptable, depending on how modest the manufacturer
> is with his specs.

You still think there is 24 bit resolution someplace in the
real world?

> Reducing the dynamic range of my
> recording device won't reduce the dynamic range of what
> I'm recording, though, unless I reduce the bit resolution
> drastically.

That's my argument.

> Once the recording is squashed the least
> significant bits will be dither anyway.

Any proper digital signal has a LSB that is either dither or
strongly influenced by dither, anyway.

> But in the
> meantime I want even the under-spec signals to go through
> all my reverbs, delays, eqs, and
> every other wacky plugin I plugin and mix. They can only
> add
> to my resolution, not detract from it.

Huh? There is no way to add to the resolution of a signal.
That would violate information theory.

> Yours is a better argument than the others have
> presented, though.

Maybe I'm not so dumb after all. ;-)
Anonymous
September 1, 2005 8:07:36 AM

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

Chevdo wrote:

> In article <tPGdnRe3DqqoMYjeRVn-gQ@comcast.com>, arnyk@hotpop.com says...
> >
>
> >Recording real-world audio is about something that by 24 bit
> >standards, is a very noisy signal.
>
> Do sentences like this mean anything? These kinds of statements are your
> downfall, Arny.
>
> >If I have an audio signal with a noise floor that is 75 dB
> >down (one heck of a clean signal!) and record it with a
> >noisy digital channel whose similar but lower noise floor is
> >85 dB down (very mediocre performance by modern standards)
> >the signal's noise floor is raised by less than 0.1 dB
> >(totally trivial) by the so-called noisy digital channel.
> >
> >I conclude that if the digital channel's noise floor is 10
> >dB better than that of the signal, then its effects can be
> >safely ignored.
>
> Yeah that sounds right, but does not seem to be addressing what I've been
> talking about at all regarding recording digital signals as hot as possible.
>
>
> >
> >In the real world, I have a DAW with digital channels that
> >have a noise floor that is 98 dB down (average performance
> >for modern digital production gear). My real world signals
> >(performer's mics) have a measured dynamic range (from mic
> >on, room quiet to performer singing vigorously) of about 70
> >dB. I conclude that I can record with 18 dB of headroom
> >while sacrificing only trivial amounts (<0.1dB) of the live
> >sound's dynamic range.
>
> I'm sure you'll do fine with that, and maybe you are sacrificing only trivial
> amounts of signal, but it's hard to say since the 'dynamic range' spec of
> analog equipment only refers to the region of power output in which the signal
> is at it's best, it does not refer to any kind of cut-off as it does in the
> digital world as bit resolution. With my huge 24bit resolution I can capture
> not only the dynamic range of my equipment, but whatever signal is beyond spec,
> some of which might end up being very acceptable, depending on how modest the
> manufacturer is with his specs. Reducing the dynamic range of my recording
> device won't reduce the dynamic range of what I'm recording, though, unless I
> reduce the bit resolution drastically. Once the recording is squashed the
> least significant bits will be dither anyway. But in the meantime I want even
> the under-spec signals to go through all my reverbs, delays, eqs, and
> every other wacky plugin I plugin and mix. They can only add
> to my resolution, not detract from it.
>
>
> Yours is a better argument than the others have presented, though.

Is this the same poster who, in another thread,
was connecting two 8ohm spakers in series to
get a 4ohm load ?
I sensed a troll at the outset.

rd
Anonymous
September 1, 2005 8:52:59 AM

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

Arny Krueger wrote:
> By most accounts (even Stereophile writers) I kicked
> [John] Atkinson's ass at the HE2005 debate, and he's
> the editor of Stereophile.

Noted without comment, other to point out that "most
accounts" have been posted by Mr. Krueger :-)

For an independent account of the debate, including a recording
please visit http://www.stereophile.com/news/050905debate. For
my own discussion of the issues I raised at the debate, please
visit http://www.stereophile.com/asweseeit/705awsi.

On the subject of this thread, A/D converters perform increasingly
less well in the top 3dB of their dynamic range. But once the
signal is in the digital domain, why is there any need for headroom,
unless further gain is going to be applied down the road?

John Atkinson
Editor, Stereophile
Anonymous
September 1, 2005 9:45:00 AM

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

Chevdo wrote:
> In article <1125315114.583019.55530@o13g2000cwo.googlegroups.com>,
> mrivers@d-and-d.com says...
>
>>
>>The concept of "headroom" in a digital system is different than in an
>>analog system, because there's a hard limit. The thing that you need to
>>understand is that you can have as much or as little headroom as you
>>want. Leaving 20 dB above the nomrmal level is considered good practice
>>for a 24-bit system. This allows sufficient headroom for normal audio
>>peaks. But if you've heavily limited your tracks so that there really
>>are no peaks more than, say 6 dB above the nominal level, you can boost
>>the track as much as 14 dB and still not have peaks that try to exceed
>>full scale.
>
>
>
> I wouldn't waste 20db headroom on any signal. I want to record as hot as
> possible and take advantage of the robustness of my 24bit signal when it goes
> through processing and track summing.

Nuh uh. It's fine. You have headroom; use it.

> If you leave 20db headroom you're only
> using about 20bits for your signal.

This is Not A Problem.

> 24bits will allow you to be lazy and leave
> a huge headroom and never worry about clipping and still come away with a
> decent recording, and I'd consider doing that if I had $5000+ a/d converters,
> but with my pro-sumer level ones, I'm looking to stretch as much quality out of
> them as possible, which means recording a signal as hot as possible
>
>

Ain't tape. Relax the levels.

>
>>However, understand that 24 (or even two) tracks that have frequent
>>peaks at or very close to full scale are likely to sum to something
>>greater than full scale, yielding clipping.
>>
>>Pardon my shouting, but THERE IS ABSOLUTELY NO REASON TO DO THIS ! ! !
>>! ! !
>
>
> Of course there is, you lower the db on each track successively (hopefully your
> DAW will let you gang them), and keep mixing down until you're just under
> clipping. Extra work? Sure, lots. Better results? Definitely. Worth it?
> Your call.
>

Suppose you print a mix @ -20 RMS. It peaks at -10dB. Post
processing said mix by adding 10dB will result in exactly the same
waveform as printing the mix at -10dB RMS, 0dBFS peak ( absent
dithering ). Try it.

The same principle applies to tracking....

>
>
>>Your time is better spent making a good mix than boosting tracks to the
>>point where you're risking clipping and then have to worry about that.
>>You can always boost your mix to clipping and beyond if you want it to
>>sound awful but louder.
>>
>
>
> Yep, and one key to making a good mix is to fill up as many bits as you can
> on each track with signal.
>

Oy. No, this is not true.

--
Les Cargill
Anonymous
September 1, 2005 9:51:37 AM

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

SSJVCmag wrote:

> On 8/28/05 11:55 PM, in article jHvQe.113343$fm.7517617@news4.tin.it, "ale"
> <sparkwest@yahooc.om> wrote:
>
>
>>yes, in cubase you can go over 0 dbfs on the individual channels without
>>internal clipping the waveform.
>
>
> Ummm isn;t this just a totally wrong use of the term " 0dBfs" ?
>
>

Sorta. With floating point, you can cheerfully represent +10 dbFS.

It's what? 2.0, where 1.0 is odBFS?

It won't *print* worth much. But that might not be the last
processing stage...

<snip>

--
Les Cargill
Anonymous
September 1, 2005 11:44:42 AM

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

Arny Krueger wrote:

> OK, so we do a recording of the Saturn V pad on launch day.
> Yes, it gets loud at launch time, but how quiet does it get
> in the 24 hours before?

For what it's worth, a few years back Bob Katz reported on a field
recording he made of a launch from the official "press" site. He used a
Masterlink powered by a UPS. I don't recall what mics and preamps he
used, but there may still be a writeup of the experiment (and the
experience) on his web site http://www.digido.com He said it was
mighty loud, but that he got a mucho satisfactory recording with the
dynamic range he had available.

Knowing the respect with which Bob treats dynamic range, I don't think
I'd want to play this at authentic volume in my house.
!