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Someone says "anything over -10db in digital video is dist..

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June 22, 2005 1:00:43 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Someone made the statement that "on a digital video deck you want to keep
the levels below -10db because above that you're into distortion." Further
they state that 0db is the same as clipping. I was under the impression that
attempting to *exceed* 0db yields clipping, since there isn't anything over
0db. Soundforge definitely makes a distinction between 0db and "clipped".

What say any of you?

Thanks for all shared wisdom.
Anonymous
June 22, 2005 1:00:44 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Doc" <docsavage20@xhotmail.com> wrote in message
news:%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net...
> Someone made the statement that "on a digital video deck you want to keep
> the levels below -10db because above that you're into distortion."
> Further
> they state that 0db is the same as clipping. I was under the impression
> that
> attempting to *exceed* 0db yields clipping, since there isn't anything
> over
> 0db. Soundforge definitely makes a distinction between 0db and "clipped".
>
> What say any of you?
>
> Thanks for all shared wisdom.

You are correct. I think the intent may have been to say that metering,
being what it is, an average of -10 (or -8 or -18 or ??) is a good place to
place the average peaks, because there will often be peaks that exceed the
average by 10 db or so. Therefore, to avoid clipping set a level that will
acommodate all peaks so that none exceeds the brick wall 0 dbfs.

Steve King
June 22, 2005 1:00:44 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Martin Heffels wrote:
> On Tue, 21 Jun 2005 21:00:43 GMT, "Doc" <docsavage20@xhotmail.com>
> wrote:
>
> >Someone made the statement that "on a digital video deck you want to keep
> >the levels below -10db because above that you're into distortion."
>
> I think this person is mixing-up two things. Yes, you keep your levels
> below -10dB in the digital domain, and you do this in order to avoid
> them reaching 0dB, because that is where they clip. The area between
> -10dB and 0dB is what is called "the headroom", and is there for
> louder sounds (plosives, cough, whatever).
> -10dB is quite high actually, if you don't know exactly what you're
> doing. You can pick your choice between -12dB, -14dB, -18dB and even
> -20dB. All these values are used as safeguard against clipping, and
> each soundo has his own standard. I use generally -12dB for normal
> dialogue, and -18dB if I don't know what could happen. But you will
> have to find your own comfort-zone, because mine does not need to be
> yours :) 
>
> cheers
>
> -martin-
>

And also the short plosives or other short transient sounds may not
register fully on the meter or led's so they may be well over 0 dB but
don't register over 0 dB.

I actually thik people should pay more attention to the AVERAGE meter
reading rather that the peak. Pick a level say -15 dB and try to set
the level so that 1/2 the time the meter is above -15 and 1/2 the time
it is below. If you do it this way, the "loudness" i.e how loud it
actually sounds ... will be more consistent.

Mark


Mark
Related resources
Anonymous
June 22, 2005 1:00:44 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Doc wrote:
> Someone made the statement that "on a digital video deck you want to keep
> the levels below -10db because above that you're into distortion." Further
> they state that 0db is the same as clipping. I was under the impression that
> attempting to *exceed* 0db yields clipping, since there isn't anything over
> 0db. Soundforge definitely makes a distinction between 0db and "clipped".
>
> What say any of you?
>
> Thanks for all shared wisdom.
>
>
>

This person is mixing levels....-10dB is a line level voltage for
certain equipment, not a reference point for where clipping begins. 0dB
is the absolute ceiling of the loudest volume that can be recorded on
digital equipment (ie. 16 1 bits for that sample). It's possible for a
-15dBu signal to cause clipping on some equipment, and a +5dB u signal
to be below clipping on other equipment.

Many people and software will consider the point at which you reach, but
not exceed 0dBFS to be clipping, simply because they are both
represented the same way in binary (all 1's)

Other people, like myself, believe that clipping may occur, but only
matters when you can hear it. Some equipment handles clipping really
well, and if you're recording a thrash distorted guitar part you might
not even notice lots of clipping. On the other hand, if you're recording
clean vocals, you probably won't be fine with even minimal clipping.

But yes I think you're right, clipping occurs when they are no more
zeros to turn on to represent the extra signal, or there are no more
metal oxide particles in a tape to represent the xtra signal/information.

Jonny Durango
Anonymous
June 22, 2005 1:00:44 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Doc wrote:

> Someone made the statement that "on a digital video deck
you want to
> keep the levels below -10db because above that you're into
> distortion."

Most digital equipment clips someplace in the last 1 dB
before FS. Up until that point, digital equipment is
generally very clean.

However this begs the question - how are you measuring
levels? There's really only one totally reliable way to
measure levels in the digital domain, and that is to record
a sample and then look at it with a DAW.

What you actually record is where the rubber hits the road.
Everything else is an estimate.

> Further they state that 0db is the same as clipping.

In many cases, clipping of real world digital equipment
takes place a bit below 0 dB, IOW -0.1 dB or maybe even as
low as -0.5 dB.

Furthermore, clipping might be frequency-dependent. If
clipping is frequency-dependent, the clip point is probably
the lowest at the highest frequencies. Most equipment will
hit its highest undistorted levels at 10 KHz, but there may
be increasing losses at higher frequencies.

Most of my cautionary comments apply to the less expensive
digital equipment.

> I was under the impression that attempting to *exceed*
0db yields
> clipping, since there isn't anything over 0db.

Fact is, a lot of digital equipment clips at some point
above -1 dB, but below 0 dB.

> Soundforge definitely
> makes a distinction between 0db and "clipped".

Theoretically, 0 dB signals are not clipped.
Anonymous
June 22, 2005 1:00:44 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

In article <%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net> docsavage20@xhotmail.com writes:

> Someone made the statement that "on a digital video deck you want to keep
> the levels below -10db because above that you're into distortion."

That would be a pretty bad digital video deck, however they may want
to keep the peaks 10 dB below full scale because they want to have
10 dB of headroom in their playback chain. It could be that the deck
that they play back from produces an analog output from a -10 dBFS
digital recording that drives their system into clipping, but that's
sloppy system engineering on their part.

In any case, it's convention, and a trade organization (SMPTE)
standard, to deliver recordings that don't peak above -10 dBFS.

> Further
> they state that 0db is the same as clipping.

That's not correct.

First off, 'dB' by itself doesn't mean anything when talking about
levels. Let's assume that we're talking about dB relative to digital
full scale, which is the maximum digital level. This is what 0 dBFS
is, and it's likely that this is what your misinformant meant by "0
dB." You can have a recording with peaks that reach 0 dBFS that isn't
clipped. And you can't exceed 0 dBFS. But if you put an analog signal
into an A/D converter high enough so that it would go over 0 dBFS if
it could, then you'll get clipping.

> Soundforge definitely makes a distinction between 0db and "clipped".

Most digital devices have a "clip" indicator, but those work in
different ways. (there's no standard for it) Some will light up when
the level gets to within a couple of tenths of a dB of full scale (say
-0.2 dBFS) just to give you warning that you're about to clip if
things get much louder. Some light up as soon as it sees a sample that
reaches full scale (this is becoming common for 24-bit systems).
Others light up when they see three (or some other number greater than
one) consecutive samples at the full scale value. That's a pretty good
guess that clipping has occurred, but if you're recording square
waves, you can have the clip light on all day and still be recording
what you put in.



--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 22, 2005 1:00:45 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Mike Rivers wrote:
> In article <%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net> docsavage20@xhotmail.com writes:
>
>> Someone made the statement that "on a digital video deck you want to keep
>> the levels below -10db because above that you're into distortion."
>
>
> That would be a pretty bad digital video deck, however they may want
> to keep the peaks 10 dB below full scale because they want to have
> 10 dB of headroom in their playback chain. It could be that the deck
> that they play back from produces an analog output from a -10 dBFS
> digital recording that drives their system into clipping, but that's
> sloppy system engineering on their part.


Agreed, but it's surprisingly common in the broadcast world.




> In any case, it's convention, and a trade organization (SMPTE)
> standard, to deliver recordings that don't peak above -10 dBFS.


Yup, gotta waste that last 10 dB by design. Sigh...
Anonymous
June 22, 2005 2:21:01 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Jonny Durango wrote:


>
> But yes I think you're right, clipping occurs when they are no more
> zeros to turn on to represent the extra signal, or there are no more
> metal oxide particles in a tape to represent the xtra signal/information.
>
> Jonny Durango

You'd hear the clipping at playback with D/A conversion, correct?

Then wouldn't it depend on the converter? If the converter created a
smooth peak (possible?) would that explain why some equipment handles it
better?
--


- Bill
June 22, 2005 2:23:46 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Doc" <docsavage20@xhotmail.com> wrote in message
news:%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net...
> Someone made the statement that "on a digital video deck you want to keep
> the levels below -10db because above that you're into distortion."
Further
> they state that 0db is the same as clipping. I was under the impression
that
> attempting to *exceed* 0db yields clipping, since there isn't anything
over
> 0db. Soundforge definitely makes a distinction between 0db and "clipped".

I think I should clarify, this is referring to material that's already been
recorded, not a question of where to set the levels during recording.

I was under the impression that the waveform and VU meter in Soundforge etc.
"shows all". That there are no peaks that don't show up on the wave form and
the VU meter tells you exactly what the peak levels are up to 0db, unlike
analog meters which show an approximation of the peaks, which are likely
somewhat below the actual levels.

Yes? No?
June 22, 2005 2:45:40 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Doc wrote:
> Someone made the statement that "on a digital video deck you want to keep
> the levels below -10db because above that you're into distortion." Further
> they state that 0db is the same as clipping. I was under the impression that
> attempting to *exceed* 0db yields clipping, since there isn't anything over
> 0db. Soundforge definitely makes a distinction between 0db and "clipped".
>
> What say any of you?
>
> Thanks for all shared wisdom.

In an environment with digital and analog VTRs "tone" always is set to read 0 on
an analog VU meter and -20 on a digital meter. I am referring mainly to Beta SP
(analog) and Digi-Beta (digital machines) With properly recorded audio the level
on a Beta SP deck will fluctuate around the 0 level marker but sometimes read a
couple of db over. The same audio will read maybe 10 db above the -20 marker on
the digital machines due to it's "peak reading" tendencies.
In the old days (with tape) when there were only VU meters an engineer had to
"just know" that certain sounds would read much lower on a VU meter than they
really were. An extreme example would be orchestra bells or a triangle. If you
let the level approach -10 on a VU meter the sound would be distorted. This was
especially important on the first generation of the recording. Some of the
extreme peaks would saturate the tape but in a gentle way. Later in the "mix"
you could boost these same sounds and avoid the distortion because the extreme
peaks in the overtones were now gone.
June 22, 2005 3:11:40 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:eb-dnf6r-ounDyXfRVn-tQ@comcast.com...
> Doc wrote:

> However this begs the question - how are you measuring
> levels? There's really only one totally reliable way to
> measure levels in the digital domain, and that is to record
> a sample and then look at it with a DAW.

Soundforge.

In case you haven't seen my previous posts I'll give you a quick rundown.
Made a Firewire transfer of a musical performance from a DVCam deck to my
Sony TRV-240 Digital8 camcorder. From there through a Firewire port to my
H/D. When looking at the still unprocessed file I found that the highest
peaks were either right at 0db or registered as clipped. All the highest
peaks were in one particular song which was somewhat higher in level than
the rest of the performance.

I should probably mention that the people who recorded the sound at the
performance really messed up all over the place - they showed up late and
didn't get to do a sound check. Her vocal levels were all over the place in
the mix from hot and distorted to buried. Half the time she couldn't hear
herself in the monitors. Her mic died during one song. During the opening
bars of the performance (for TV), someone apparently hit the "on" button on
the onboard reverb - suddenly the sound is awash in ridiculously overboard
reverb and a couple of seconds later it suddenly goes away (I can just see
the board guy fumbling in a panicked scramble for the fader or pot.) It was
apparent from the shape of the waveform that for some reason the first and
last songs had compression/limiting applied to them - with a lot of
accompanying distortion - I assume with an onboard compressor/limiter they
had. Why just the first and last songs I have no idea. Given everything else
that happened, I have to wonder if they even knew they were doing it. Can't
imagine why the TV studio would do it.

Anyway, the fact is there's distortion all over the place, even in places
where the sound is nowhere near clipping. It's not caused by my gear, the
VHS copy the station gave her of the original tape sounds the same way.

I used Soundforge to add reverb via the Acoustic Mirror plugin. To do this,
I had to lower the overall levels by 2db to keep many more peaks from
clipping than already were once the reverb was applied. I figured I should
put it back to where it was, i.e. with the highest peaks (maybe 2 or 3
spikes where this ocurred) at 0db, which is what I did. Saved the whole
thing with the DV format, played it back to my camcorder via firewire and
then back to their DVcam deck.

When I mentioned to the Station manager about the levels issue, he seemed
incredulous that any of them were that hot. I explained that this is where
they were on the original file. He even asked if I had "adjusted" anything
during the Firewire transfer. I'm not aware that this is even possible. I
was under the impression that a Firewire transfer is utterly unlike going
from an analog source, that a transfer simply dumps whatever data is on the
digital tape to the h/d, no user intervention possible. At any rate, all I
did was set up the cam, hook up the Firewire cable and hit "capture" on the
Pinnacle Studio 9 software.
Anonymous
June 22, 2005 3:16:48 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

On Tue, 21 Jun 2005 21:00:43 GMT, "Doc" <docsavage20@xhotmail.com>
wrote:

>Someone made the statement that "on a digital video deck you want to keep
>the levels below -10db because above that you're into distortion."

I think this person is mixing-up two things. Yes, you keep your levels
below -10dB in the digital domain, and you do this in order to avoid
them reaching 0dB, because that is where they clip. The area between
-10dB and 0dB is what is called "the headroom", and is there for
louder sounds (plosives, cough, whatever).
-10dB is quite high actually, if you don't know exactly what you're
doing. You can pick your choice between -12dB, -14dB, -18dB and even
-20dB. All these values are used as safeguard against clipping, and
each soundo has his own standard. I use generally -12dB for normal
dialogue, and -18dB if I don't know what could happen. But you will
have to find your own comfort-zone, because mine does not need to be
yours :) 

cheers

-martin-

--
"Now I want you to say it thrice daily and don't dress a bun"
Anonymous
June 22, 2005 7:00:15 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Doc wrote:


> When I mentioned to the Station manager about the levels
issue, he
> seemed incredulous that any of them were that hot. I
explained that
> this is where they were on the original file. He even
asked if I had
> "adjusted" anything during the Firewire transfer.

What's to adjust? ;-)

> I'm not aware that
> this is even possible. I was under the impression that a
Firewire
> transfer is utterly unlike going from an analog source,
that a
> transfer simply dumps whatever data is on the digital tape
to the
> h/d, no user intervention possible.

That's my experience.

> At any rate, all I did was set up
> the cam, hook up the Firewire cable and hit "capture" on
the Pinnacle
> Studio 9 software.

I've done my camcorder transfers using real basic software
and hardware - namely Windows Movie Maker (WMM) and a
generic Firewire card. It seems to work like clock - I push
some buttons, wait for a while and there's an AVI file on my
hard drive.

If I need to play with the sound track more seriously than
what WMM allows, I've used Audition/CE to pull the sound
track off of the computer file that Windows Movie Maker
created. WMM supports adding a stereo audio track back into
the movie from my hard drive.

Since the audio is always kept in the digital domain,
there's really not a lot to mess it up. It is what it is,
until I start playing with it with one of the editors that I
use.
Anonymous
June 22, 2005 11:52:37 AM

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

In article <M81ue.9648$VK4.26@newsread1.news.atl.earthlink.net> docsavage20@xhotmail.com writes:

> Made a Firewire transfer of a musical performance from a DVCam deck to my
> Sony TRV-240 Digital8 camcorder. From there through a Firewire port to my
> H/D. When looking at the still unprocessed file I found that the highest
> peaks were either right at 0db or registered as clipped.

That seems like it was a pretty hot recording. There was nothing in
your transfer process that would have added gain, so it is what it is.
If you were to play back the audio from the analog outputs of your
camcorder, you probably would have heard the clipping.

Unless one of the transfer options that you used automatically
normalized the recording on the way in. If peaks were formerly at
-10 dBFS, normalization would have brough the loudest peak up to
0 dBFS, but still not clipped digitally. You might want to take
another look at the settings in your transfer process. There might be
a button on there that you don't realize is pressed.

> I should probably mention that the people who recorded the sound at the
> performance really messed up all over the place - they showed up late and
> didn't get to do a sound check. Her vocal levels were all over the place in
> the mix from hot and distorted to buried.

> Anyway, the fact is there's distortion all over the place, even in places
> where the sound is nowhere near clipping.

That could explain the clipping on the recording. But depending on the
analog chain feeding the original recorder, it could have been clipped
going into the recorder. The digital level might have been OK, but
they were recording a clipped signal, perhaps at the mic input stage.

> When I mentioned to the Station manager about the levels issue, he seemed
> incredulous that any of them were that hot. I explained that this is where
> they were on the original file. He even asked if I had "adjusted" anything
> during the Firewire transfer. I'm not aware that this is even possible.

I've seen the "normalize on import" option on something that I have
around here. I don't have Sound Forge so I know that's not where I saw
it, but I know that it's an option on something. Maybe it's on
Tracktion. It's not a program that I use often enough to remember, and
it's not an option that I'd want to use.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 22, 2005 1:11:39 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

On Tue, 21 Jun 2005 23:11:40 GMT, "Doc" <docsavage20@xhotmail.com>
wrote:

>I'm not aware that this is even possible. I
>was under the impression that a Firewire transfer is utterly unlike going
>from an analog source, that a transfer simply dumps whatever data is on the
>digital tape to the h/d, no user intervention possible.

You are right.

-m-

--
"Now I want you to say it thrice daily and don't dress a bun"
Anonymous
June 22, 2005 3:39:37 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

I haven't read all the posts , but -10 digital seems to becoming a new
T.V. standard ,
From feeding cameras to the quality control inspector at discovery channel
sending shows back if there are peaks over that !

Maybe Will , will lend his experience , but it seems to combatting sloppy
field
work and over agressive transmitter chains .

regards Greg



"Doc" <docsavage20@xhotmail.com> wrote in message
news:%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net...
> Someone made the statement that "on a digital video deck you want to keep
> the levels below -10db because above that you're into distortion."
Further
> they state that 0db is the same as clipping. I was under the impression
that
> attempting to *exceed* 0db yields clipping, since there isn't anything
over
> 0db. Soundforge definitely makes a distinction between 0db and "clipped".
>
> What say any of you?
>
> Thanks for all shared wisdom.
>
>
>
Anonymous
June 22, 2005 7:43:21 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

FWIW, we set reference tone at -10 on the DV deck's meters when recording
dialog. For music, we usually use -12. But on dialog, we're nearly always
compressed at least 3:1 at the mixer, so the 10 is more like 30.

And for music, there's just too much the audio guy does to the 2-mix to
fully explain here, but for levels, there's usually a compressor on most
every input (some of them 2-stage). For critical events, there's a separate
multi-track recording for re-mix, etc.

Steve



"Steve King" <steve@TakeThisOutToReplysteveking.net> wrote in message
news:3I-dnVTKI_-LGiXfRVn-oA@comcast.com...
> "Doc" <docsavage20@xhotmail.com> wrote in message
> news:%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net...
> > Someone made the statement that "on a digital video deck you want to
keep
> > the levels below -10db because above that you're into distortion."
> > Further
> > they state that 0db is the same as clipping. I was under the impression
> > that
> > attempting to *exceed* 0db yields clipping, since there isn't anything
> > over
> > 0db. Soundforge definitely makes a distinction between 0db and
"clipped".
> >
> > What say any of you?
> >
> > Thanks for all shared wisdom.
>
> You are correct. I think the intent may have been to say that metering,
> being what it is, an average of -10 (or -8 or -18 or ??) is a good place
to
> place the average peaks, because there will often be peaks that exceed the
> average by 10 db or so. Therefore, to avoid clipping set a level that
will
> acommodate all peaks so that none exceeds the brick wall 0 dbfs.
>
> Steve King
>
>
June 24, 2005 9:52:09 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

The general standard for video in the US is (and has been for quite
some time) peaks at -10 and vu levels peaking between -18 and -24.
While this standard does fail to take advantage of the additional 10db
of headroom available in the digital realm, it allows for a seamless
signal flow between analog and digital machines.
This is not the case with DVDs, where there is not yet a clear standard
established.
Anonymous
June 24, 2005 1:42:59 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

On Tue, 21 Jun 2005 23:16:48 +0200, Martin Heffels
<safeshop77@yahoo.com> wrote:

>On Tue, 21 Jun 2005 21:00:43 GMT, "Doc" <docsavage20@xhotmail.com>
>wrote:
>
>>Someone made the statement that "on a digital video deck you want to keep
>>the levels below -10db because above that you're into distortion."
>
>I think this person is mixing-up two things. Yes, you keep your levels
>below -10dB in the digital domain, and you do this in order to avoid
>them reaching 0dB, because that is where they clip. The area between
>-10dB and 0dB is what is called "the headroom", and is there for
>louder sounds (plosives, cough, whatever).
>-10dB is quite high actually, if you don't know exactly what you're
>doing. You can pick your choice between -12dB, -14dB, -18dB and even
>-20dB. All these values are used as safeguard against clipping, and
>each soundo has his own standard. I use generally -12dB for normal
>dialogue, and -18dB if I don't know what could happen. But you will
>have to find your own comfort-zone, because mine does not need to be
>yours :) 

Over here in the BBC, we use -18dbfs as our reference level and we set
our peak to 8db higher than that - ie -10dbfs. This ensure that
there's always 10db of headroom before the onset of clipping.

Steve

The Doctor Who Restoration Team Website
http://www.restoration-team.co.uk
Anonymous
June 24, 2005 4:29:18 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

In article <1119617529.569620.242500@g47g2000cwa.googlegroups.com> dougk@musician.org writes:

> The general standard for video in the US is (and has been for quite
> some time) peaks at -10 and vu levels peaking between -18 and -24.

That's true for digital peak levels, which are not indicated by a VU
meter. Do you know where the -18 to -24 range on the VU scale is?
Let's not get our meters confused.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 26, 2005 6:09:27 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Doc" <docsavage20@xhotmail.com> wrote:
>
> Someone made the statement that "on a digital video deck you want
> to keep the levels below -10db because above that you're into
> distortion."



They *might* have meant that it causes distortion further down the
chain. Or maybe they're just unclear on the details.

Either way, the standard in digital video these days is 0VU = -20dBFS
with peaks not exceeding -10dBFS. Some facilities nudge the 0VU mark up
or down a bit, but the -10dBFS ceiling seems pretty consistent.

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

(Remove spamblock to reply)
Anonymous
June 26, 2005 6:09:28 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Lorin David Schultz" <Lorin@DAMNSPAM!v5v.ca> wrote in message
news:rGyve.60173$wr.18573@clgrps12...
> "Doc" <docsavage20@xhotmail.com> wrote:
>>
>> Someone made the statement that "on a digital video deck you want
>> to keep the levels below -10db because above that you're into
>> distortion."
>
>
>
> They *might* have meant that it causes distortion further down the chain.
> Or maybe they're just unclear on the details.
>
> Either way, the standard in digital video these days is 0VU = -20dBFS with
> peaks not exceeding -10dBFS. Some facilities nudge the 0VU mark up or
> down a bit, but the -10dBFS ceiling seems pretty consistent.
>

Of course, haven't we all seen the postings about Sony PD150/170 cameras and
their prosumer equivilents VX2000/2001 that will deliver excessively noisy
results at those settings? It is my understanding that other manufacturers
of this category of camera suffer similarly. Shortly after I switched from
BetaSP to DVCAM I had to reschedule an interview that was recorded to the
standards suggested by Lorin on a PD150. The result was simply too noisy to
use. It was an honest mistake by an experienced sound professional who
tried to apply standards that worked with analogue BetaSP to a new digital
format that simply doesn't have the capability to accommodate 20 dB of
headroom. (As a side note, the second interview was a bust. The
spontanaity of the first sitting was gone. Caution replaced candor. It
will, unfortunately, not be used in the documentary that is still an ongoing
project.)

Steve King
Anonymous
June 26, 2005 6:09:29 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

In article <VOOdnQR0UunuIyPfRVn-3A@comcast.com> steve@TakeThisOutToReplysteveking.net writes:

> Of course, haven't we all seen the postings about Sony PD150/170 cameras and
> their prosumer equivilents VX2000/2001 that will deliver excessively noisy
> results at those settings?

Not me, but then I don't travel in those circles.

> It is my understanding that other manufacturers
> of this category of camera suffer similarly. Shortly after I switched from
> BetaSP to DVCAM I had to reschedule an interview that was recorded to the
> standards suggested by Lorin on a PD150. The result was simply too noisy to
> use. It was an honest mistake by an experienced sound professional who
> tried to apply standards that worked with analogue BetaSP to a new digital
> format that simply doesn't have the capability to accommodate 20 dB of
> headroom.

According to the classic definition, no digital system has any
headroom. When you get to 0 dBFS, you have no place else to go. You
make your own amount of headroom based on your familiarity (or
unfamiliarity) with the program material, and adherence to the
client's wishes. At -20 dBFS, in even the crummiest of the crummy, you
should have at least 60 dB of dynamic range available. Are you
suggesting that you barely have 20 dB of dynamic range below -20 dBFS?
That's absurd.

Maybe someone told him "record beween -10 and -20" and he was watching
the VU meter on his mixer barely move off the downscale pin. That, of
course, won't word, since 0 VU is calibrated to some nominal level
which is probably in the ballpark of the nominal recording level,
analog or digital (unless there's a +4/-10 discrepancy that nobody
accounted for).


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 26, 2005 6:09:30 PM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1119798122k@trad...
>
> In article <VOOdnQR0UunuIyPfRVn-3A@comcast.com>
> steve@TakeThisOutToReplysteveking.net writes:
>
>> Of course, haven't we all seen the postings about Sony PD150/170 cameras
>> and
>> their prosumer equivilents VX2000/2001 that will deliver excessively
>> noisy
>> results at those settings?
>
> Not me, but then I don't travel in those circles.

I'm not sure what you mean by this, but the topic of audio problems with the
PD150 and VX2000 have been discussed on this newsgroup as well as more
extensively on RAMPS. I'll include a quote from Jay Rose about the PD150:

"Sony started with the inferior preamp/A>D in their VX2000, and added
balancing and phantom. Period. When the world complained about the awful
s/n, they (AFAIK) added a
software-based noise gate. No per-unit cost, those clever guys... If you
do a traditional measurement with an input signal that goes away -- what
Sound Devices did -- the gate kicks in and s/n is apparently improved.
It's simply a single-band masking phenomenon... the noise is still there,
and apparent with some signals."

>> It is my understanding that other manufacturers
>> of this category of camera suffer similarly. Shortly after I switched
>> from
>> BetaSP to DVCAM I had to reschedule an interview that was recorded to the
>> standards suggested by Lorin on a PD150. The result was simply too noisy
>> to
>> use. It was an honest mistake by an experienced sound professional who
>> tried to apply standards that worked with analogue BetaSP to a new
>> digital
>> format that simply doesn't have the capability to accommodate 20 dB of
>> headroom.
>
> According to the classic definition, no digital system has any
> headroom.

I was speaking of the 20 dB between -20 dBfs and digital full scale, but
thanks for the lecture anyway.

>When you get to 0 dBFS, you have no place else to go. You
> make your own amount of headroom based on your familiarity (or
> unfamiliarity) with the program material, and adherence to the
> client's wishes. At -20 dBFS, in even the crummiest of the crummy, you
> should have at least 60 dB of dynamic range available. Are you
> suggesting that you barely have 20 dB of dynamic range below -20 dBFS?
> That's absurd.

What I tried to say, perhaps not well, is that if the tone is set at -20dBfs
and program peaks are allowed to go no higher than -16 dBfs on a PD150 with
the internal limiters off, audio recorded in a quiet setting will have an
objectionable amount of pre-amp noise. I am simply relating my experience
in hundreds of hours of shooting with PD150s and VX2000s and making my
evaluations based on 40 years of broadcast, recording studio, and video
production experience. But, I'm still learning every day ;-)

> Maybe someone told him "record beween -10 and -20" and he was watching
> the VU meter on his mixer barely move off the downscale pin. That, of
> course, won't word, since 0 VU is calibrated to some nominal level
> which is probably in the ballpark of the nominal recording level,
> analog or digital (unless there's a +4/-10 discrepancy that nobody
> accounted for).

Unfortunately, I was the director, so if there was any telling going on it
was me doing it. It was a very early shoot with the PD150 and the first
interior in a quiet office. Previous locations had been in exterior
locations, where ambient noise masked the camera audio deficiencies.

All that said, when one knows the inherent problems of the Sony cameras,
they can deliver excellent picture results and satisfactory audio tracks. I
have since adopted a working method for run and gun using a microphone (or
wireless receiver) directly into the camera of leaving the internal limiters
on and adjusting levels to peak around -6 dbfs. The internal limiters are
not great, but they do prevent clipping. This seems to be an acceptable
compromise. When feeding the cameras from an external mixer better results
can be had by turning off the internal camera limiters and using the mixer
limiters to keep audio from exceeding digital full scale yet still peak
around the -6 dBfs point.

Steve King

> I'm really Mike Rivers (mrivers@d-and-d.com)
> However, until the spam goes away or Hell freezes over,
> lots of IP addresses are blocked from this system. If
> you e-mail me and it bounces, use your secret decoder ring
> and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 27, 2005 7:02:11 AM

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

Mike Rivers wrote:
> In article <VOOdnQR0UunuIyPfRVn-3A@comcast.com> steve@TakeThisOutToReplysteveking.net writes:
>
> > Of course, haven't we all seen the postings about Sony PD150/170 cameras and
> > their prosumer equivilents VX2000/2001 that will deliver excessively noisy
> > results at those settings?
>
> Not me, but then I don't travel in those circles.
>
> > It is my understanding that other manufacturers
> > of this category of camera suffer similarly. Shortly after I switched from
> > BetaSP to DVCAM I had to reschedule an interview that was recorded to the
> > standards suggested by Lorin on a PD150. The result was simply too noisy to
> > use. It was an honest mistake by an experienced sound professional who
> > tried to apply standards that worked with analogue BetaSP to a new digital
> > format that simply doesn't have the capability to accommodate 20 dB of
> > headroom.
>
> According to the classic definition, no digital system has any
> headroom. When you get to 0 dBFS, you have no place else to go. You
> make your own amount of headroom based on your familiarity (or
> unfamiliarity) with the program material, and adherence to the
> client's wishes. At -20 dBFS, in even the crummiest of the crummy, you
> should have at least 60 dB of dynamic range available. Are you
> suggesting that you barely have 20 dB of dynamic range below -20 dBFS?
> That's absurd.
>
> Maybe someone told him "record beween -10 and -20" and he was watching
> the VU meter on his mixer barely move off the downscale pin. That, of
> course, won't word, since 0 VU is calibrated to some nominal level
> which is probably in the ballpark of the nominal recording level,
> analog or digital (unless there's a +4/-10 discrepancy that nobody
> accounted for).

I find this hard to believe too. On the other hand I've heard some
rough sound from one of these cameras recording an interview. I think
they were using 32kHz sampling through ignorance, I don't think the
levels were too bad, but the amount of mushy noise from the lossy
compression was unimpressive.

Alex
Anonymous
June 27, 2005 7:51:10 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Doc" <docsavage20@xhotmail.com> wrote in message
news:%d%te.9599$VK4.460@newsread1.news.atl.earthlink.net...
> Someone made the statement that "on a digital video deck you want to
> keep
> the levels below -10db because above that you're into distortion."

Sounds like they were refering to some sort of local "stanndard"
for keeping the average at -10dBFS to allow 10dB of hedroom.

10dB seems pretty risky unless you have a very predictable
source. Most of us use -20dBFS as the reference (it would
have been called the 0dB point back in analog days.)

> Further they state that 0db is the same as clipping. I was under
> the impression that attempting to *exceed* 0db yields clipping,
> since there isn't anything over 0db.

In the real world the difference between 0dB and OVER 0dB
is nonexistent. Nobody can adjust levels to keep the signal peaks
at 0 without going over. Some compressors can come very
close, but you don't want to hear what happens to your audio
when you push it that close.

> Soundforge definitely makes a distinction between 0db and "clipped".

Soundforge is working on after-the-fact files, not real-time
recording level setting.
June 27, 2005 10:50:36 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

I use Dorrough 280-D meters, which display vu and peak levels
simultaneously. Since -20 on the digital scale = 0 vu, vu levels
peaking between -18 and -24 on the digital scale = +2 to -4 in the
analog realm, so long as machines are calibrated to -20. Sorry this
wasn't clear in my last post.

Mike Rivers wrote:
> In article <1119617529.569620.242500@g47g2000cwa.googlegroups.com> dougk@musician.org writes:
>
> > The general standard for video in the US is (and has been for quite
> > some time) peaks at -10 and vu levels peaking between -18 and -24.
>
> That's true for digital peak levels, which are not indicated by a VU
> meter. Do you know where the -18 to -24 range on the VU scale is?
> Let's not get our meters confused.
June 27, 2005 5:04:39 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

I'm with ya Mike. In addition to the Dorroughs, I have an old pair of
analog vu meters I keep an eye on as well when setting levels.
Anonymous
June 27, 2005 7:27:23 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

In article <1119880236.254093.246940@g49g2000cwa.googlegroups.com> dougk@musician.org writes:

> I use Dorrough 280-D meters, which display vu and peak levels
> simultaneously. Since -20 on the digital scale = 0 vu, vu levels
> peaking between -18 and -24 on the digital scale = +2 to -4 in the
> analog realm, so long as machines are calibrated to -20. Sorry this
> wasn't clear in my last post.

That's a good meter and it tells you a lot. Being an old fuddyduddy
myself, I still use an analog console with pretty close to real VU
meters on it. I find that with my digital stuff calibrated so that
0 VU equals -20 dBFS, by watching the VU meters the way I always have,
I get perfectly satisfactory record levels.

Some people think that when they look at the waveform view on their
DAW and the squiggles don't fill up the full track width, they have a
"weak signal." I keep reminding them that -6 dBFS, which is a
reasonably hot level, only uses half the graphic height, so -10 dBFS
is just a wiggly line down the middle. That's why they have zoom
buttons on these things.




--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 27, 2005 9:03:02 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

On Sun, 26 Jun 2005 14:09:27 GMT, "Lorin David Schultz"
<Lorin@DAMNSPAM!v5v.ca> wrote:

>Either way, the standard in digital video these days is 0VU = -20dBFS
>with peaks not exceeding -10dBFS.

Avid seems to think 0VU=-14dBFS.
I know I am whinging here, but your standard does not have to be my
standard. If you want to speak about such a standard, it is wise to
add in which country you are, so people can apply your suggestions
to their situations.

cheers

-martin-

--
"Now I want you to say it thrice daily and don't dress a bun"
Anonymous
June 28, 2005 6:01:54 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

On 2005-06-27 mrivers@d-and-d.com said:
>>dougk@musician.org writes: I use Dorrough 280-D meters, which
>>display vu and peak levels simultaneously. Since -20 on the
>>digital scale = 0 vu, vu levels peaking between -18 and -24 on
>>the digital scale = +2 to -4 in the analog realm, so long as
>>machines are calibrated to -20. Sorry this wasn't clear in my
>last post. That's a good meter and it tells you a lot. Being an old
>fuddyduddy myself, I still use an analog console with pretty close
>to real VU meters on it. I find that with my digital stuff
>calibrated so that 0 VU equals -20 dBFS, by watching the VU meters
>the way I always have, I get perfectly satisfactory record levels.
Ditto here. I have a vu indicator which gives me an audible tone at
0vu and set things up with my wif's help using her eyes so that a 1
khz tone generates -20 dbfs once we've gone over to digital. I get
satisfactory recordings this way.

>Some people think that when they look at the waveform view on their
>DAW and the squiggles don't fill up the full track width, they have
>a "weak signal." I keep reminding them that -6 dBFS, which is a
>reasonably hot level, only uses half the graphic height, so -10 dBFS
>is just a wiggly line down the middle. That's why they have zoom
>buttons on these things.
I remind them that it's audio, you're supposed to use your friggin'
ears!!!
That's why we ahve another thread about why it has to be so f**ing
loud which has been hijacked into an analog vs digital thread.




Richard Webb,
Electric SPider Productions, New Orleans, La.
REplace anything before the @ symbol with elspider for real email

--



THe knobs turn in both directions". That's why it's called
mixing, otherwise we would call it adding
Anonymous
June 28, 2005 12:25:57 PM

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

In article <mc2we.9772$Zo.3833@bignews3.bellsouth.net> 0junk4me@bellsouth.net writes:

> I have a vu indicator which gives me an audible tone at
> 0vu and set things up with my wif's help using her eyes so that a 1
> khz tone generates -20 dbfs once we've gone over to digital. I get
> satisfactory recordings this way.

The problem that's becoming more and more common is that there's no
convenient way to adjust the input sensitivity of a digital recorder,
so you can't calibrate your system. You could calibrate the VU meter
so that it reads 0 at whatever level coming out of the console gives
you -20 dBFS, but even fewer devices today have VU meters that can be
calibrated by the user (other than by changing parts) to set a
different reference level than what the manufacturer offers - almost
always +4 dBu or -10 dBV (unless you're Mackie and get the bright idea
that people are confused with 0 - +4 and made their early mixers put
out 0 dBu when the meters read 0).

One of these days I'm going to write an article on all the stuff that
manufacturers leave out (and stick the user to work around) and input
level controls is going to be right at the top of my list.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 28, 2005 1:23:28 PM

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

Mike Rivers wrote:
> In article <mc2we.9772$Zo.3833@bignews3.bellsouth.net>
> 0junk4me@bellsouth.net writes:
>
>> I have a vu indicator which gives me an audible tone at
>> 0vu and set things up with my wif's help using her eyes
so that a 1
>> khz tone generates -20 dbfs once we've gone over to
digital. I get
>> satisfactory recordings this way.
>
> The problem that's becoming more and more common is that
there's no
> convenient way to adjust the input sensitivity of a
digital recorder,
> so you can't calibrate your system.

Well, when in Rome you do as the Romans.

I suspect that the most common means for setting the gain of
systems with digital recorders is to adjust something in the
analog signal chain that preceeds it. For me, that's always
the mic preamp.

My recommended procedure is to record a loud passage during
rehearsal, set gains as required to ensure that your
recording has sufficient (at least 10 dB) headroom, based on
visual inspection of the individual track recordings as
displayed in full-screen mode, perhaps with some time
expansion.

A similar means is avaiable when you record off the inserts
or direct outs of a console. If your gain staging through
the console is reasonable, you just naturally end up with
the right levels going into the digital recorder, as you set
the individual channel trims by ear.

Indeed, if the individual channel recordings fail the first
test above, its probable that your gain staging through the
console is off and needs adjusting.
Anonymous
June 28, 2005 2:29:56 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

Steve King wrote:
> "Lorin David Schultz" <Lorin@DAMNSPAM!v5v.ca> wrote in message
> news:rGyve.60173$wr.18573@clgrps12...
>
>
>> the standard in digital video these days is 0VU = -20dBFS with
>> peaks not exceeding -10dBFS. Some facilities nudge the 0VU mark up or
>> down a bit, but the -10dBFS ceiling seems pretty consistent.
>
>
> Of course, haven't we all seen the postings about Sony PD150/170 cameras and
> their prosumer equivilents VX2000/2001 that will deliver excessively noisy
> results at those settings? It is my understanding that other manufacturers
> of this category of camera suffer similarly.


Right, -12 dBFS is generally a better reference level for most DVcams.
Having an external preamp or mixer with a decent limiter helps quite a bit.
Anonymous
June 28, 2005 8:32:00 PM

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

In article <RIqdnUvujNDN0lzfRVn-hA@comcast.com> arnyk@hotpop.com writes:

> I suspect that the most common means for setting the gain of
> systems with digital recorders is to adjust something in the
> analog signal chain that preceeds it.

I'm sure you're right about that.

> For me, that's always the mic preamp.

But good gain management tells us that this isn't always the best way
to do it. (and of course we always want to do our best, don't we?) You
want to get all the gain you need in as early a stage as you can, and
then keep unity gain past that. So you adjust your preamp gain so that
it's plenty high but safely away from clipping, and at point, your
peaks are at +20 dBu. If your A/D converter is calibrated so that it
clips at +14 dBu ("calabrated" to -14) then your nice clean and quiet
preamp will cause your converter to clip. The right place to reduce
the level is at the output of the preamp, not the input, or at the
input of the A/D converter.

Conversely, if your converter has lower sensitivity, you might be
tempted to increase the gain of the preamp until it's going into
clipping, in an attempt to "record a hot signal." We see that a lot
around here.

> Indeed, if the individual channel recordings fail the first
> test above, its probable that your gain staging through the
> console is off and needs adjusting.

Exactly - and that's what you can't always adjust in the right place.
Most of the time, the only control you have over the direct output of
a console (or channel insert send) is the mic preamp trim control. You
can usually make the record level meters look right but you might be
compromising the signal-to-noise ratio or you might be driving your
mic preamp into clipping.

If you're smart, in the latter case, you'll recognize that it's
clipping and back it off (and complain that your preamp isn't "hot
enough." but most of the time probalby what will happen is that the
clipped signal will be reocrded and the A/D converter will get blamed.

--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
June 29, 2005 3:03:34 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Richard Crowley" <rcrowley7@xprt.net> wrote in message
news:11bvmh0ojhckf43@corp.supernews.com...

> > Soundforge definitely makes a distinction between 0db and "clipped".
>
> Soundforge is working on after-the-fact files, not real-time
> recording level setting.


Right, that's what I was referring to, material that's already been
recorded. When I got it, it had a few peaks of 0db and some that were
clipped.
Anonymous
June 29, 2005 8:48:43 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Steve King" <steve@TakeThisOutToReplysteveking.net> wrote:
>
> [...] Shortly after I switched from BetaSP to DVCAM I had to
> reschedule an interview that was recorded to the standards suggested
> by Lorin on a PD150. The result was simply too noisy to use. It was
> an honest mistake by an experienced sound professional who tried to
> apply standards that worked with analogue BetaSP to a new digital
> format that simply doesn't have the capability to accommodate 20 dB
> of headroom.



Hey, don't blame me! I just stated the facts, I didn't set the
standard! <g>

Our SX camcorders are all calibrated to 0VU = -20dBFS and we don't have
any noise problems. Of course, you could buy a truckload of PD150s for
what one of those costs though.

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

(Remove spamblock to reply)
Anonymous
June 29, 2005 8:48:43 AM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Martin Heffels" <safeshop77@yahoo.com> wrote:
>
> Avid seems to think 0VU=-14dBFS.

Which Avid? Ours is -20.



> I know I am whinging here, but your standard does not have to be my
> standard. If you want to speak about such a standard, it is wise to
> add in which country you are, so people can apply your suggestions
> to their situations.



There seems to be a whole lotta "shoot the messenger" coming from one of
the groups to which this thread is cross-posted. I didn't establish the
standard, I'm just quoting it. Whether anyone chooses to observe it or
not is no skin off my meters... like I said, I didn't come up with it so
my feelings won't be hurt if someone doesn't like it (though a committee
at Sony may take you off their Christmas card list).

Maybe it's different across the pond (I'm in North America), but I don't
think so. I think it's just a case of some people and organizations
choosing to adopt different approaches. Any piece of digital pro video
gear I've ever encountered has the zero VU mark at -20 on the meters
(note that I've not actually seen the Avid to which you refer). Maybe
other devices that I haven't seen are different.

Anyway, how does your comment relate to the original poster's question?

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

(Remove spamblock to reply)
Anonymous
June 29, 2005 10:06:06 AM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1119976835k@trad
> In article <RIqdnUvujNDN0lzfRVn-hA@comcast.com>
> arnyk@hotpop.com writes:
>
>> I suspect that the most common means for setting the gain
of
>> systems with digital recorders is to adjust something in
the
>> analog signal chain that preceeds it.
>
> I'm sure you're right about that.
>
>> For me, that's always the mic preamp.

> But good gain management tells us that this isn't always
the
> best way to do it. (and of course we always want to do our
> best, don't we?) You want to get all the gain you need in
as
> early a stage as you can, and then keep unity gain past
that.

My approach does just that.

> So you adjust your preamp gain so that it's plenty high
but
> safely away from clipping, and at point, your peaks are at
+20
> dBu. If your A/D converter is calibrated so that it clips
at
> +14 dBu ("calabrated" to -14) then your nice clean and
quiet
> preamp will cause your converter to clip.

No, that's not how I do it, nor is it how I advise people to
do it.

I tell people to adjust the gain (in this case the mic
preamp gain) so that they have the desired amount of
headroom in the digital domain - by looking at the actual
display of an actual recording they make during rehearsal.

IOW, if the user manual says that the direct outs or
inserts are +4, then I set the input sensitivity on the
computer digital audio interface to +4 and then set the
trims or mic preamp gains so that the display in the DAW
software shows that the loudest part of the loudest music is
still recording at least 10 dB below FS. If I work on mixing
the tracks and see that the headroom for some channel is
less than 10 dB, I nudge the related trim down as required
before I record the next session.

>The right place to reduce
> the level is at the output of the preamp, not the input,
or at
> the input of the A/D converter.

I think that's what I meant by: "For me, that's always the
mic preamp."

> Conversely, if your converter has lower sensitivity, you
might
> be tempted to increase the gain of the preamp until it's
going
> into clipping, in an attempt to "record a hot signal." We
see
> that a lot around here.

Hence my constant harping about maintaining about 10 dB
headroom over actual observed levels, as seen in the digital
domain.

>> Indeed, if the individual channel recordings fail the
first
>> test above, its probable that your gain staging through
the
>> console is off and needs adjusting.

> Exactly - and that's what you can't always adjust in the
right place.

If your direct outs or insert points are running at the same
nominal level as your digital recorder's input sensitivity
is set for, then adjusting the mic preamp to make the
digital recorder happy automatically ensures that the rest
of the console will be happy, too.


> Most of the time, the only control you have over the
> direct output of
> a console (or channel insert send) is the mic preamp trim
> control.

Agreed.

> You can usually make the record level meters look
> right but you might be compromising the signal-to-noise
ratio
> or you might be driving your mic preamp into clipping.

<rant on>

Where did I say *anything* about meters? I hate meters. I
never take them seriously when I am recording or mixing. I
ordered my new 02R96 without a meter bridge, and I hope to
*never* have any meters attached to it. I expect to make
minimal use of the 02R96 built-in metering. The output leds
on my Mackie SR32 are among its least-used features. BTW,
when I do look at them they look *right*, but that's a
natural consequence of good hygiene everyplace else.

<rant off>

> If you're smart, in the latter case, you'll recognize that
it's
> clipping and back it off (and complain that your preamp
isn't
> "hot enough." but most of the time probalby what will
happen
> is that the clipped signal will be reocrded and the A/D
> converter will get blamed.

That's one reason why I tell people to set levels based on
the individual channel display(s) at full magnification in
the DAW software. Then, there's no question about how much
headroom there is between peaks and FS, and there's no
question about how meter response relates to the music you
are recording.
Anonymous
June 29, 2005 1:14:50 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

On Wed, 29 Jun 2005 04:48:43 GMT, "Lorin David Schultz"
<Lorin@DAMNSPAM!v5v.ca> wrote:

>"Martin Heffels" <safeshop77@yahoo.com> wrote:
>>
>> Avid seems to think 0VU=-14dBFS.
>
>Which Avid? Ours is -20.

Heheeh, they don't have their own standard internally. It's on the
Xpress Pro sofwtare per factory. But it's adjusteable.

>There seems to be a whole lotta "shoot the messenger" coming from one of
>the groups to which this thread is cross-posted. I didn't establish the
>standard, I'm just quoting it.

Not shooting you :)  But there is no global standard and it would be
nice to see where they keep which levels around the globe.

>Anyway, how does your comment relate to the original poster's question?

Because I don't think the OP stated where he came from. Granted, nine
out of ten times it will be the US, but suppose he is in a country
where they use -16, and he start using -20 as you said, his material
might be rejected everywhere, as levels being to high. You know,
broadcast-technicians are very sensitive when it comes to levels :) 

The other day in one of the other groups, it came out that sometimes
in one country, there are two different standards (crikey, Australia).

cheers

-martin-

--
"Now I want you to say it thrice daily and don't dress a bun"
Anonymous
June 29, 2005 1:25:48 PM

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

In article <RrydneC2nuMT71_fRVn-rw@comcast.com> arnyk@hotpop.com writes:

> IOW, if the user manual says that the direct outs or
> inserts are +4, then I set the input sensitivity on the
> computer digital audio interface to +4 . . . . .

One of my all-too-often points is that many computer digital audio
interfaces have no way to set the input sensitivity unless you do it
externally. "External" could be an output level control on the mic
preamp, or, if you have no other choice, the input gain of the preamp.
But that only works if you have to turn the level to the converter
down, not up, to achieve the desired amount of headroom.

It can all be workable, and I know that you have the understanding to
make it work. But most of the time when people are faced with this
problem, they don't have time to learn what's happening, they work on
instinct (or just turn knobs until the meters read right, not
listening to what's being recorded) and then ask on r.a.p. after the
fact what was wrong with their mic preamp.

> If your direct outs or insert points are running at the same
> nominal level as your digital recorder's input sensitivity
> is set for, then adjusting the mic preamp to make the
> digital recorder happy automatically ensures that the rest
> of the console will be happy, too.

But that "if" isn't universally true. If there was in interface
standard to which the industry adhered, we'd have an easier time with
this, but marketing pressures more often that not at least on gear
that might be somewhat lacking) cause this to be moved around for the
sake of the best advertiseable numbers.

> Where did I say *anything* about meters? I hate meters.

Meters are good, but you have to know what they're telling you. I
woudln't want to be without a meter bridge on my console because it's
a quick look at what's working and what needs some attention.

> That's one reason why I tell people to set levels based on
> the individual channel display(s) at full magnification in
> the DAW software.

I can't think of a program that allows you to do this in real time
though. It takes a test recording (or several), and faith that the
setting you've established during your test will represent what
happens in an actual take. Leaving 20 dB of headroom is usally safe,
but on a graphic display, that just looks like a skinny wiggly line. A
waveform display is really only useful when you're close to the limit.

Keeping a watchful eye on the meters allows you to make adjustments in
real time if necessary. But then this is a skill that you have to
develop, along with others.



--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 29, 2005 2:21:11 PM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1120045997k@trad
> In article <RrydneC2nuMT71_fRVn-rw@comcast.com>
> arnyk@hotpop.com writes:
>
>> IOW, if the user manual says that the direct outs or
>> inserts are +4, then I set the input sensitivity on the
>> computer digital audio interface to +4 . . . . .
>
> One of my all-too-often points is that many computer
digital
> audio interfaces have no way to set the input sensitivity
> unless you do it externally.

Well *many* is one of those vague words that seems to mean
something... ;-)

Almost all of the pro interfaces I've used (over 30) have
had at least -10 and +4 as options. One recent big nasty
surprise was the fact that the AP 24192 lacked the slectable
input sensitivity feature that graces virtually all of the
rest of the M-Audio line. But, it does a pretty credible
+4, and that is *the* pro standard, right?

> "External" could be an output
> level control on the mic preamp, or, if you have no other
> choice, the input gain of the preamp. But that only works
if
> you have to turn the level to the converter down, not up,
to
> achieve the desired amount of headroom.

The way almost every audio interface stacks up is that the
the stated sensitivity is way under FS. IOW a -10 input will
generally put FS someplace around 2 volts, and a +4 input
runs from 2.5 to about 8 volts for FS.

> It can all be workable, and I know that you have the
> understanding to make it work. But most of the time when
> people are faced with this problem, they don't have time
to
> learn what's happening, they work on instinct (or just
turn
> knobs until the meters read right, not
> listening to what's being recorded) and then ask on r.a.p.
> after the fact what was wrong with their mic preamp.

That's one reason why I keep telling people to match up the
specified numbers on the preamp and the interface, and then
*look* at what they are recording.

>> If your direct outs or insert points are running at the
same
>> nominal level as your digital recorder's input
sensitivity
>> is set for, then adjusting the mic preamp to make the
>> digital recorder happy automatically ensures that the
rest
>> of the console will be happy, too.

> But that "if" isn't universally true.

Nothing is universally true, but what do you call a console
that gets bent of shape if you take its specs at face value?
I call it unprofessional junk. Its not like the console
market has a sole source...

>If there was in interface
> standard to which the industry adhered, we'd have an
easier
> time with this, but marketing pressures more often that
not at
> least on gear
> that might be somewhat lacking) cause this to be moved
around
> for the sake of the best advertiseable numbers.

I don't see a lot of marketing grease in fudging specd
output levels.

>> Where did I say *anything* about meters? I hate meters.

> Meters are good, but you have to know what they're telling
> you.

Plan B: forget about what the meters say and trust the most
relevant and accurate empirical results. In the race between
ears and meters I'll take ears every place they work.

>I woudln't want to be without a meter bridge on my
> console because it's a quick look at what's working and
what needs some attention.

If you haven't noticed, I recommend a paradigm for recording
that really doesn't require a lot of attention during
tracking. Someplace around 16 tracks one meter per track
starts becoming more of a light show than a relevant tool.
I'll take the ability to listen to the track with headphones
(a la Mackie's grotesquely misnamed solo buttons) over a
meter, any day. Thats one reason why consoles have headphone
jacks - so you can listen! ;-)

>> That's one reason why I tell people to set levels based
on
>> the individual channel display(s) at full magnification
in
>> the DAW software.

> I can't think of a program that allows you to do this in
real
> time though.

It's very hard to do metering right in real time. So, why
make it a critical sucess factor?

> It takes a test recording (or several), and faith
> that the setting you've established during your test will
> represent what happens in an actual take.

Admittedly you have to know something about the work habits
of the talent and the equipment. Getting along with and
having a feel for talent is one of those big advantages of
using flesh and blood technical staff as opposed to relying
on machines for that.

Besides, level setting need only be very approximate during
tracking - that's one of the things that headroom is for.

> Having 20 dB of headroom is usally
> safe, but on a graphic display, that just looks like a
skinny
> wiggly line.

Hence my repeated advice that people only seriously judge
waves based a full-screen view per track. Besides 20 dB
headroom is pretty excessive in most cases. Did someone say
10 dB headroom? ;-)

> A waveform display is really only useful when
> you're close to the limit.

Agreed, so there's no reason to make the waveform display
during recording, a critical sucess factor, either.

> Keeping a watchful eye on the meters allows you to make
> adjustments in real time if necessary.

Real time adjustments make mixing more confusing and more
work later on. Bad form for tracking except in dire
emergencies.

> But then this is a skill that you have to develop, along
with others.

I see real time adjustments as a skill I only practice when
I'm doing live sound.

I think I made my last real time adjustment for recording
levels during tracking about 3 months ago... That ruined the
whole day for me! ;-)
Anonymous
June 29, 2005 4:46:46 PM

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

In article <CLidnXnpVbrFM1_fRVn-pw@comcast.com> arnyk@hotpop.com writes:

> Well *many* is one of those vague words that seems to mean
> something... ;-)

That's why I used "many" rather than "most."

> Almost all of the pro interfaces I've used (over 30) have
> had at least -10 and +4 as options.

Yes, but what digital level is represented by -10 dBV or +4 dBu?
That's where you need the missing control, so you can set it for
-20, -18, -14, -12 or whatever you want.

> One recent big nasty
> surprise was the fact that the AP 24192 lacked the slectable
> input sensitivity feature that graces virtually all of the
> rest of the M-Audio line. But, it does a pretty credible
> +4, and that is *the* pro standard, right?

Right. What's its calibration? Put in +4 dBu and how many dBFS do you
get? Whatever it is, you take it or leave it (or compensate for it
elsewhere). That's my beef.

> The way almost every audio interface stacks up is that the
> the stated sensitivity is way under FS. IOW a -10 input will
> generally put FS someplace around 2 volts, and a +4 input
> runs from 2.5 to about 8 volts for FS.

Well, for the "+4 input" that's about a 10 dB range in calibration -
not very standard, is it?

> I don't see a lot of marketing grease in fudging specd
> output levels.

No, but they want to make the S/N and noise floor numbers as good as
they can. One way to do this is to eliminate any unnecessary gain or
gain control element on the input or output.

> If you haven't noticed, I recommend a paradigm for recording
> that really doesn't require a lot of attention during
> tracking.

It depends on what you're tracking. You can get away with that
sometimes. If it works for you, fine. But not everyone will be happy
with set-it-and-forget-it.

> It's very hard to do metering right in real time. So, why
> make it a critical sucess factor?

Program? Metering? I thought we were talking about real meters here.

> Besides, level setting need only be very approximate during
> tracking - that's one of the things that headroom is for.

No, headroom is so that you can accommodate dynamic range of
performance, not accommodate unpredictable performance. One good
example of wanting to ride a fader (or preamp gain) is when a
performer goes from singing to talking. There might be a 20 dB
difference that it's good to make up at the front. It's just good
engineering practice.

> Real time adjustments make mixing more confusing and more
> work later on. Bad form for tracking except in dire
> emergencies.

Good real time adjustments make mixing easier. Good tracks practically
mix themselves.

> I see real time adjustments as a skill I only practice when
> I'm doing live sound.

I'm ALWAYS doing live sound, even when recording in the studio.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
June 29, 2005 6:50:14 PM

Archived from groups: rec.video.desktop,rec.audio.pro,rec.video.production (More info?)

"Martin Heffels" <safeshop77@yahoo.com> wrote:
>
> Heheeh, they don't have their own standard internally. It's on the
> Xpress Pro sofwtare per factory. But it's adjusteable.



Of course it's adjustable, but the interface has a "default" level.
Look at the meters on the hardware box.

Maybe you don't have dedicated Avid hardware? I haven't seen the XPress
stuff.

Not that it matters in our circumstances... we're not doing a-d at the
Avid anyway. It comes in SDI, so the level will have been set at the
converters in the control room (or at the camera in the case of field
acquisition).

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

(Remove spamblock to reply)
Anonymous
June 29, 2005 8:19:01 PM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1120063521k@trad
> In article <CLidnXnpVbrFM1_fRVn-pw@comcast.com>
> arnyk@hotpop.com writes:
>
>> Well *many* is one of those vague words that seems to
mean
>> something... ;-)
>
> That's why I used "many" rather than "most."
>
>> Almost all of the pro interfaces I've used (over 30) have
>> had at least -10 and +4 as options.
>
> Yes, but what digital level is represented by -10 dBV or
+4
> dBu?

Something like 12 to 16 dB below FS.

> That's where you need the missing control, so you can set
it
> for -20, -18, -14, -12 or whatever you want.

But, we don't have that. What we do have gets us within a
few dB of a standard.

>> One recent big nasty
>> surprise was the fact that the AP 24192 lacked the
slectable
>> input sensitivity feature that graces virtually all of
the
>> rest of the M-Audio line. But, it does a pretty credible
>> +4, and that is *the* pro standard, right?

> Right. What's its calibration? Put in +4 dBu and how many
dBFS
> do you get? Whatever it is, you take it or leave it (or
> compensate for it elsewhere). That's my beef.

Nothing's perfect. OTOH, close is fine when it comes to
setting headroom.

>> The way almost every audio interface stacks up is that
the
>> the stated sensitivity is way under FS. IOW a -10 input
will
>> generally put FS someplace around 2 volts, and a +4 input
>> runs from 2.5 to about 8 volts for FS.

> Well, for the "+4 input" that's about a 10 dB range in
> calibration - not very standard, is it?

I guess we need to put this on our wish lists for audio
interface vendors.

>> I don't see a lot of marketing grease in fudging specd
>> output levels.

> No, but they want to make the S/N and noise floor numbers
as
> good as they can. One way to do this is to eliminate any
> unnecessary gain or gain control element on the input or
> output.

Agreed, and that's pretty much how the whole industry went.


>> If you haven't noticed, I recommend a paradigm for
recording
>> that really doesn't require a lot of attention during
>> tracking.

> It depends on what you're tracking. You can get away with
that
> sometimes. If it works for you, fine. But not everyone
will be
> happy with set-it-and-forget-it.


Paradigm shift is pretty well guaranteed to make *someone*
unhappy. Better, easier speaks to me.


>> It's very hard to do metering right in real time. So, why
>> make it a critical sucess factor?

> Program? Metering? I thought we were talking about real
meters
> here.

On top of everything else they do suboptimally, real meters
cost money! Bahhh!

>> Besides, level setting need only be very approximate
during
>> tracking - that's one of the things that headroom is for.

> No, headroom is so that you can accommodate dynamic range
of
> performance, not accommodate unpredictable performance.

There's plenty of evidence that headroom suits both
purposes.

> One good example of wanting to ride a fader (or preamp
gain) is when a
> performer goes from singing to talking.

Thing is, you can handle that common situation far better
after the fact.

> There might be a 20 dB
> difference that it's good to make up at the front.

If it could be done as well in real time, but it can't.

> It's just good engineering practice.

What constitutes good engineering practice changes with the
state of the art.

>> Real time adjustments make mixing more confusing and more
>> work later on. Bad form for tracking except in dire
>> emergencies.

> Good real time adjustments make mixing easier.

Due to the well-known failings of human omnisicence,
real-time mixing is very limited in terms of precision and
accuracy. Because it always has to happen in real time, it
can be far more time-consuming than is necessary.

Due to the well-known superiority of hindsight to foresight,
mixing during the mix is generally the better way to go.

> Good tracks practically mix themselves.

The mix is the best time to mix. That's why they call it the
mix. ;-)

>> I see real time adjustments as a skill I only practice
when
>> I'm doing live sound.

> I'm ALWAYS doing live sound, even when recording in the
studio.

Show me how to make foresight as accurate and reliable as
hindsight, and you've got a deal!
Anonymous
June 30, 2005 5:21:58 AM

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

Arny Krueger wrote:

> That's one reason why I tell people to set levels based on
> the individual channel display(s) at full magnification in
> the DAW software. Then, there's no question about how much
> headroom there is between peaks and FS, and there's no
> question about how meter response relates to the music you
> are recording.

It's really just a different kind of meter, a fast one with
history. I agree with you, by the way.


Bob
--

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

A. Einstein
!