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Mackie 1202 vlz pro

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Anonymous
January 13, 2005 4:57:28 PM

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

I was thinking about using a Mackie 1202 vlz pro as a mic preamp for drums.
I was wondering how many individual outputs I can get at once on this unit.

Thanks

More about : mackie 1202 vlz pro

Anonymous
January 13, 2005 4:57:29 PM

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

Sanbar from the land of sanbar@wi.rr.com wrote:

> I was thinking about using a Mackie 1202 vlz pro as a mic preamp for drums.
> I was wondering how many individual outputs I can get at once on this unit.
>
> Thanks
>

I'm sure there is someone here who can give you an exact count, but the font
of all wisdom, the seat of all things Mackie is located here:

http://www.mackie.com/support/downloads/manuals.html

Carlos
Anonymous
January 13, 2005 4:57:29 PM

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

Sanbar,

> how many individual outputs <

Four. Also see "Using a Mixer with a DAW" from EQ magazine, first in the
list on my Articles page:

www.ethanwiner.com/articles.html

It was written specifically for you!

--Ethan
Related resources
Anonymous
January 13, 2005 4:57:30 PM

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

"Carlos Alden" <calden3@msn.com> wrote in message
news:BE0BCAD8.3574F%calden3@msn.com
> Sanbar from the land of sanbar@wi.rr.com wrote:
>
>> I was thinking about using a Mackie 1202 vlz pro as a mic preamp for
>> drums. I was wondering how many individual outputs I can get at once
>> on this unit.
>>
>> Thanks
>>
>
> I'm sure there is someone here who can give you an exact count, but
> the font of all wisdom, the seat of all things Mackie is located here:
>
> http://www.mackie.com/support/downloads/manuals.html

Right, and that gets you to http://www.mackie.com/pdf/1202vlzpro_om.pdf .

Page 9 shows that there are inserts on the first four input channels, which
happily handle mics.

Page 14 shows standard insert plug wiring for the 4 insert jacks which are
TRS, tip out, ring in.

My preference for tapping insert points is to use a special cable with a TRS
plug on one end that has the tip and ring soldered together. Thus the signal
from the mic preamp is available for recording, and it's also available for
the normal use of the mixer.

Anohter approach is to use a standard insert cable with the two 1/4 plugs
inserted into a "Y" connector.

If you set the channel trims for normal mixer operation, the insert points
will have standardized signals on them. Those are your channel gain controls
for the 4 mic inputs.
Anonymous
January 13, 2005 6:16:25 PM

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

My concern after looking at the manual is that there is no way to control
the individual outputs.


"Carlos Alden" <calden3@msn.com> wrote in message
news:BE0BCAD8.3574F%calden3@msn.com...
> Sanbar from the land of sanbar@wi.rr.com wrote:
>
> > I was thinking about using a Mackie 1202 vlz pro as a mic preamp for
drums.
> > I was wondering how many individual outputs I can get at once on this
unit.
> >
> > Thanks
> >
>
> I'm sure there is someone here who can give you an exact count, but the
font
> of all wisdom, the seat of all things Mackie is located here:
>
> http://www.mackie.com/support/downloads/manuals.html
>
> Carlos
>
Anonymous
January 13, 2005 6:16:26 PM

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

Sanbar <sanbar@wi.rr.com> wrote:
>My concern after looking at the manual is that there is no way to control
>the individual outputs.

What do you mean?

Are you intending on using the mike inputs and the insert outputs? If you
do this, the trims are all the control you need.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
January 13, 2005 6:41:56 PM

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

In article <c7vFd.194603$ye4.93882@twister.rdc-kc.rr.com> sanbar@wi.rr.com writes:

> I was thinking about using a Mackie 1202 vlz pro as a mic preamp for drums.
> I was wondering how many individual outputs I can get at once on this unit.

You really need only one per mic and you can get that from the Insert
jack on each of the four mic channels. The limitation when using it as
a mic preamp for drums is not how many outputs you can get, but how
many mic inputs you have, which is four. If that's enough, you're in
good shape.


--
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
January 13, 2005 9:29:58 PM

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

Sanbar <sanbar@wi.rr.com> wrote:

> I was thinking about using a Mackie 1202 vlz pro as a mic preamp for drums.
> I was wondering how many individual outputs I can get at once on this unit.
>
> Thanks

You tap the insert outputs by using the "only to the first click"
insertion technique, and then you control the level to staorage with the
mixer's input sensitivity ("gain") control. The signal still gets
through teh bord for monitoring.

This is @ Goole in the Group archives, covered well here many times.

--
ha
Anonymous
January 13, 2005 10:41:22 PM

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

Arny Krueger wrote:

> My preference for tapping insert points is to use a special cable with a TRS
> plug on one end that has the tip and ring soldered together. Thus the signal
> from the mic preamp is available for recording, and it's also available for
> the normal use of the mixer.

Arny,

If you think it through, you'll realize that inserting a patch cable
only to the first click bridges the insert connection and accomplishes
just what you want without a special cable. Work with TS or TRS cables.
Try it.

--
ha
Anonymous
January 13, 2005 10:41:23 PM

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

"hank alrich" <walkinay@thegrid.net> wrote in message
news:1gqbs79.1f1z5xn1dv6bulN%walkinay@thegrid.net
> Arny Krueger wrote:
>
>> My preference for tapping insert points is to use a special cable
>> with a TRS plug on one end that has the tip and ring soldered
>> together. Thus the signal from the mic preamp is available for
>> recording, and it's also available for the normal use of the mixer.
>
> Arny,
>
> If you think it through, you'll realize that inserting a patch cable
> only to the first click bridges the insert connection and accomplishes
> just what you want without a special cable. Work with TS or TRS
> cables. Try it.

Been there done that, and loved it until the first time I moved something
and the cable popped out.
Anonymous
January 13, 2005 10:41:23 PM

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

In article <1gqbs79.1f1z5xn1dv6bulN%walkinay@thegrid.net> walkinay@thegrid.net writes:

> If you think it through, you'll realize that inserting a patch cable
> only to the first click bridges the insert connection and accomplishes
> just what you want without a special cable. Work with TS or TRS cables.

But when someone bumps into that not-fully-inserted plug, off goes the
channel to the rest of the mixer. A T_R-Jumpered plug can be fully
inserted without interrupting the channel, and it's a physically safer
connection. The half-in trick is OK in the studio but dangerous for
live work.



--
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
January 13, 2005 11:19:34 PM

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

Arny Krueger wrote:

> "hank alrich" wrote:
> > Arny Krueger wrote:

> >> My preference for tapping insert points is to use a special cable
> >> with a TRS plug on one end that has the tip and ring soldered
> >> together. Thus the signal from the mic preamp is available for
> >> recording, and it's also available for the normal use of the mixer.

> > Arny,

> > If you think it through, you'll realize that inserting a patch cable
> > only to the first click bridges the insert connection and accomplishes
> > just what you want without a special cable. Work with TS or TRS
> > cables. Try it.

> Been there done that, and loved it until the first time I moved something
> and the cable popped out.

<j>

Well, don't _do_ that!

</j>

--
ha
Anonymous
January 13, 2005 11:43:12 PM

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

Thanks for all the feedback. I feel like an idiot for not fully
understanding what you all are saying. I think the bottom line here is that
I can use 4 mics and get 4 individually controlled outputs by using insert
cables and plugging them through to my DAW.

Am I correct.

Thanks again.


"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1105629081k@trad...
>
> In article <c7vFd.194603$ye4.93882@twister.rdc-kc.rr.com> sanbar@wi.rr.com
writes:
>
> > I was thinking about using a Mackie 1202 vlz pro as a mic preamp for
drums.
> > I was wondering how many individual outputs I can get at once on this
unit.
>
> You really need only one per mic and you can get that from the Insert
> jack on each of the four mic channels. The limitation when using it as
> a mic preamp for drums is not how many outputs you can get, but how
> many mic inputs you have, which is four. If that's enough, you're in
> good shape.
>
>
> --
> 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
January 13, 2005 11:43:13 PM

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

Sanbar from the land of sanbar@wi.rr.com wrote:

> Thanks for all the feedback. I feel like an idiot for not fully
> understanding what you all are saying. I think the bottom line here is that
> I can use 4 mics and get 4 individually controlled outputs by using insert
> cables and plugging them through to my DAW.
>
> Am I correct.
>
> Thanks again.

Yes.

Another way is this: Use the inputs 1-4. Pan input 1 hard left, input 2
hard right, input 3 hard left, input 4 hard right. MUTE channels 3 and 4.

Now input 1 and 2 are going out MAIN OUT left and right. Input 3 and input
4 are going out ALT 3-4 on the back. All 1/4" outs, one for each of four
input channels. This way is NOT as technically clean as the channel insert
described already, which bypasses eq and slider gain. But it also works.

Carlos
Anonymous
January 14, 2005 9:55:02 AM

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

"Sanbar" <sanbar@wi.rr.com> wrote in message
news:o hGFd.107361$NO5.60238@twister.rdc-kc.rr.com

> Thanks for all the feedback. I feel like an idiot for not fully
> understanding what you all are saying. I think the bottom line here
> is that I can use 4 mics and get 4 individually controlled outputs by
> using insert cables and plugging them through to my DAW.
>
> Am I correct.

Yes.

You control the gain with the channel trims.
Anonymous
January 14, 2005 2:11:24 PM

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

In article <OhGFd.107361$NO5.60238@twister.rdc-kc.rr.com> sanbar@wi.rr.com writes:

> Thanks for all the feedback. I feel like an idiot for not fully
> understanding what you all are saying. I think the bottom line here is that
> I can use 4 mics and get 4 individually controlled outputs by using insert
> cables and plugging them through to my DAW.

You won't have full control. The insert jacks come before the
equalizer and channel gain knob (the knobss at the bottom) so the only
real control you have is the trim (the knobs at the top), which will
adjust the level going to the DAW sound card inputs.

You have to watch this, though. If you turn them up too far in an
attempt to record at a hotter level, you'll drive the mic preamp into
clipping and make a nice clean digital recoring of a distorted
microphone. Use the solo buttons to set the trim initially, and if you
want to turn up the level to the sound card, go ahead and do that, but
check the preamp output by pressing the solo button, make sure that
peaks aren't consistently hitting the +10 light on the 1202's VU
meter, and listen to the channel with headphones plugged into the
phones jack on the mixer, to be sure that it sounds OK.

If your sound card has a switch (hardware or software) to change the
input level from +4 to -10, you're probably better off setting it for
-10. That way you won't have to push the mic preamp too hard in order
to get a visually satisfying record level.

One other tip. The low cut filter is ahead of the insert jack, so when
recording drums, you might want to switch that in for overhead or
cymbal mics, and maybe the snare mic, too. Experiment.


--
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
January 14, 2005 7:14:25 PM

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

it has 4 preamps. the rest of the i/o's are line level in varying forms
and routing.
Anonymous
January 15, 2005 4:35:37 AM

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

On Fri, 14 Jan 2005 06:55:02 -0500, "Arny Krueger" <arnyk@hotpop.com>
wrote:

>> Thanks for all the feedback. I feel like an idiot for not fully
>> understanding what you all are saying. I think the bottom line here
>> is that I can use 4 mics and get 4 individually controlled outputs by
>> using insert cables and plugging them through to my DAW.
>>
>> Am I correct.
>
>Yes.
>
>You control the gain with the channel trims.

Sort of. Channel trims need to be set right, not used as volume
controls really. You can turn them DOWN from the optimum level,
but maybe at the expense of noise level.

But then, as you'll normally be aiming to get optimum level onto the
tracks and mix later, it doesn't really matter.

CubaseFAQ www.laurencepayne.co.uk/CubaseFAQ.htm
"Possibly the world's least impressive web site": George Perfect
Anonymous
January 15, 2005 4:35:38 AM

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

"Laurence Payne" <l@laurenceDELETEpayne.freeserve.co.uk> wrote in
message news:llsgu0dfn7fh8f4dljptcoksgatn2kr5br@4ax.com
> On Fri, 14 Jan 2005 06:55:02 -0500, "Arny Krueger" <arnyk@hotpop.com>
> wrote:
>
>>> Thanks for all the feedback. I feel like an idiot for not fully
>>> understanding what you all are saying. I think the bottom line here
>>> is that I can use 4 mics and get 4 individually controlled outputs
>>> by using insert cables and plugging them through to my DAW.

>>> Am I correct.

>> Yes.

>> You control the gain with the channel trims.

> Sort of.

No, absolutely.

> Channel trims need to be set right, not used as volume
> controls really.

Agreed. Why would one use trims as volume controls while recording? Why
would one use *anything* as a volume control while recording?

> But then, as you'll normally be aiming to get optimum level onto the
> tracks and mix later, it doesn't really matter.

Agreed.
Anonymous
January 15, 2005 9:06:48 AM

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

In article <llsgu0dfn7fh8f4dljptcoksgatn2kr5br@4ax.com> l@laurenceDELETEpayne.freeserve.co.uk writes:

> >You control the gain with the channel trims.

> But then, as you'll normally be aiming to get optimum level onto the
> tracks and mix later, it doesn't really matter.

The problem is that the way most A/D converters are calibrated, if you
try to get maximum level on the tracks, you'll tend to turn the trims
up higher than they want to be, and will find that the preamps clip
before reaching peak record level. If the sound card can be set for
-10 dBV nominal input, that will give you enough sensitivity so that
you don't need to run the preamp gain too high. But if you're stuck at
+4 dBu nominal level, you probably won't reach full scale before the
preamp clips.


--
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
January 15, 2005 10:11:33 AM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1105754401k@trad
> In article <llsgu0dfn7fh8f4dljptcoksgatn2kr5br@4ax.com>
> l@laurenceDELETEpayne.freeserve.co.uk writes:
>
>>> You control the gain with the channel trims.
>
>> But then, as you'll normally be aiming to get optimum level onto the
>> tracks and mix later, it doesn't really matter.
>
> The problem is that the way most A/D converters are calibrated, if you
> try to get maximum level on the tracks, you'll tend to turn the trims
> up higher than they want to be, and will find that the preamps clip
> before reaching peak record level.

Somehow I've never had that problem, given that I record with adeqite
headroom.

> If the sound card can be set for
> -10 dBV nominal input, that will give you enough sensitivity so that
> you don't need to run the preamp gain too high.

In fact when I did that with my church's Mackie SR32, I had problems with
inadequate headroom.

> But if you're stuck at +4 dBu nominal level, you probably won't reach
> full scale before the
> preamp clips.

According to http://www.mackie.com/pdf/sr24vlzpro_ss.pdf , the insert outs
of a SR32 have low distortion at +14. IOW, they are not clipping. The least
sensitive pro sound card I know of clips with an input of +12, and another
with +10.
Anonymous
January 15, 2005 2:09:31 PM

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

In article <q7ednaQ9wrX1H3XcRVn-tw@comcast.com> arnyk@hotpop.com writes:

> Agreed. Why would one use trims as volume controls while recording? Why
> would one use *anything* as a volume control while recording?

To make the meters go into the red so you get all the bits you paid
for all the time.

And sometimes it helps to ride gain when recording, or at least make
changes to accommodate changing conditions. I know, with 24 bits you
have plenty of room to record at too low a level, but if you don't
have to, you don't have to.


--
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
January 15, 2005 2:09:32 PM

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

Mike Rivers from the land of mrivers@d-and-d.com wrote:

>
>> Agreed. Why would one use trims as volume controls while recording? Why
>> would one use *anything* as a volume control while recording?
>
> To make the meters go into the red so you get all the bits you paid
> for all the time.
>
> And sometimes it helps to ride gain when recording, or at least make
> changes to accommodate changing conditions. I know, with 24 bits you
> have plenty of room to record at too low a level, but if you don't
> have to, you don't have to.

I suggested to the poster to use the alt 3-4 trick to get four channels
separately out. I know, I know, this adds another layer of gain stage,
processing, and potential (well, definite) distortion to the mix as
compared to using the channel inserts and using trims as gain. But this way
I can use the main sliders for volume control and eq if necessary, depending
on the source. For most home recording into a computer applications this is
good enough quality.

Carlos

Carlos
Anonymous
January 15, 2005 4:15:36 PM

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

In article <26mdnTLAccLjlXTcRVn-ow@comcast.com> arnyk@hotpop.com writes:

> According to http://www.mackie.com/pdf/sr24vlzpro_ss.pdf , the insert outs
> of a SR32 have low distortion at +14. IOW, they are not clipping. The least
> sensitive pro sound card I know of clips with an input of +12, and another
> with +10.

The Mackie recorder with analog I/O clips at +24 dBu. I was under the
impression that most sound "professional" sound cards ran somewhere in
the ballpark of +16 to +16 dBu for 0 dBFS. I would expect "consumer"
and "-10" cards to reach full scale in the +10 to +12 dBV range. A lot
of people are unhappy with recordings that rarely go above -6 dBFS
because the waveform display looks so wimpy. You and I are not. I
regularly run the Mackie recorder below -10 dBFS because that's where
my console is comfortable.


--
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
January 15, 2005 4:15:37 PM

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

Mike Rivers wrote:
>
> The Mackie recorder with analog I/O clips at +24 dBu. I was under the
> impression that most sound "professional" sound cards ran somewhere in
> the ballpark of +16 to +16 dBu for 0 dBFS.

My RME ADI-8DS reaches 0 dBFS at either +13 dBu when set at +4 and at +19 dBu when set for 'hi gain' (which really means hi level.)



> I would expect "consumer"
> and "-10" cards to reach full scale in the +10 to +12 dBV range.

The ADI-8DS reaches 0 dBFS at +2 dBV when set for -10.



> A lot of
> people are unhappy with recordings that rarely go above -6 dBFS
> because the waveform display looks so wimpy. You and I are not.
> I regularly run the Mackie recorder below -10 dBFS because that's
> where my console is comfortable.

Of course, you understand the concept of properly setting system levels to maximize overall headroom. I fear that in these digital days, that is no longer being drilled into the heads of newbie engineers the way it was for many of us. I remember spending *days* doing this for postproduction studios, trying to eek out every last bit of dynamic range from the room.
Anonymous
January 15, 2005 7:47:39 PM

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

<< Why
> would one use *anything* as a volume control while recording? >>



I believe it's part of the "Recording Engineer" job description.

Scott Fraser
Anonymous
January 15, 2005 8:22:37 PM

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

On Sat, 15 Jan 2005 13:11:33 +0100, Arny Krueger wrote:

>> But if you're stuck at +4 dBu nominal level, you probably won't reach
>> full scale before the
>> preamp clips.
>
> According to http://www.mackie.com/pdf/sr24vlzpro_ss.pdf , the insert
> outs of a SR32 have low distortion at +14. IOW, they are not clipping.
> The least sensitive pro sound card I know of clips with an input of +12,
> and another with +10.

The correct specification for the Mackie 1202 vlz pro is at
http://www.mackie.com/pdf/1202vlzpro_ss.pdf
They claim a 22db headroom above 0dBu.
I always use a +4 dBu setting for my soundcards with my Mackie.

--
Chel van Gennip
Visit Serg van Gennip's site http://www.serg.vangennip.com
Anonymous
January 15, 2005 8:22:38 PM

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

Chel van Gennip <chel@vangennip.nl> wrote:
>On Sat, 15 Jan 2005 13:11:33 +0100, Arny Krueger wrote:
>
>>> But if you're stuck at +4 dBu nominal level, you probably won't reach
>>> full scale before the
>>> preamp clips.
>>
>> According to http://www.mackie.com/pdf/sr24vlzpro_ss.pdf , the insert
>> outs of a SR32 have low distortion at +14. IOW, they are not clipping.
>> The least sensitive pro sound card I know of clips with an input of +12,
>> and another with +10.
>
>The correct specification for the Mackie 1202 vlz pro is at
>http://www.mackie.com/pdf/1202vlzpro_ss.pdf
>They claim a 22db headroom above 0dBu.
>I always use a +4 dBu setting for my soundcards with my Mackie.

I wouldn't believe the spec. The Mackie starts to sound funny, long before it
actually reaches clipping. It can help a lot to run it with lower operating
levels.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
January 15, 2005 8:46:11 PM

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

On Sat, 15 Jan 2005 17:09:31 +0100, Mike Rivers wrote:


> In article <q7ednaQ9wrX1H3XcRVn-tw@comcast.com> arnyk@hotpop.com writes:
>
>> Agreed. Why would one use trims as volume controls while recording? Why
>> would one use *anything* as a volume control while recording?
>
> To make the meters go into the red so you get all the bits you paid for
> all the time.
>
> And sometimes it helps to ride gain when recording, or at least make
> changes to accommodate changing conditions. I know, with 24 bits you
> have plenty of room to record at too low a level, but if you don't have
> to, you don't have to.

What is "too low a level"? It is clear what is "too hig a level. With 24
bits you get 20 bits or more, you have the room to keep it safe. Staying
6 or even 12 dB from the hard high limit will cost you 1 or 2 of these
bits you payed for. Sometimes the best performances are a bit louder than
expected in advance, so why throw those away and spend extra time for a
second take. Riding the gain during recording sessions normally is too
late and even when you are in time it is not making things much better.
Using the bits you paid for to get some headroom is using these bits well.

--
Chel van Gennip
Visit Serg van Gennip's site http://www.serg.vangennip.com
Anonymous
January 15, 2005 8:46:12 PM

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

In article <34svmjF4bfce9U1@individual.net> chel@vangennip.nl writes:

> What is "too low a level"?

For me, it's when the meter never gets above -20 dBFS. But there are
many people who believe that if it never hits 0 dBFS, and in fact ever
drops below -10 dBFS unless the music stops, the "recording isn't hot
enough." I think it's a product of looking at waveforms. -6 dB is half
way to full scale on most waveform displays, so even a recording that
peaks at -6 looks like it's not even close to filling the window. So
many people are tempted to turn it up.

It's a hard sell, though. Too many people either remember or read
about 16-bit technology where you only had about 12 reliable bits to
start with. There, it was more important to keep close to the peak
level.

The other thing is that people stick a CD in the drive on the
computer, start it playing, and set the playback volume to that
reference, then play one of their own recordings and it's much
quieter. Of course they can't be bothered with turning up the playback
volume so they assume they need to record at a higher level.



--
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
January 15, 2005 8:46:12 PM

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

Chel van Gennip wrote:

> Using the bits you paid for to get some headroom is using these bits well.

Hear, hear!

Do you pre jocks know what the best (and the pretty darned
good and the fair to middlin) SNR that can be obtained at 48
dB of gain? I'm curious how many real bits a "24 bit" A/D
must have to be equivalent so as to eliminate the need for a
pre.


Bob
--

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

A. Einstein
Anonymous
January 16, 2005 1:29:14 AM

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

> Mike Rivers (that's me!) wrote:
> >
> > The Mackie recorder with analog I/O clips at +24 dBu. I was under the
> > impression that most sound "professional" sound cards ran somewhere in
> > the ballpark of +16 to +16 dBu for 0 dBFS.

> > I would expect "consumer"
> > and "-10" cards to reach full scale in the +10 to +12 dBV range.


I must not have been fully awake when I wrote that. How about in the
ballpark of +16 to +24 dBu for 0 dBFS for a "pro" level card and in
the ballpark of +2 to +8 dBV for a "consumer card."

Which is in the ballpark of Kurt's RME card.

> Of course, you understand the concept of properly setting system levels to
> maximize overall headroom. I fear that in these digital days, that is no
> longer being drilled into the heads of newbie engineers the way it was for
> many of us.

People talk about headroom in a digital system, but there really is
none - It's only what you make. And more often than not, you don't
have all the knobs you need in order to get the headroom you want to
have. Ideally you should use all the gain you can get at the mic
preamp without clipping, so the input gain control on the mixer or
preamp not only sets the minimum noise floor, but also sets the
headroom at the front end. So you set the gain so that when the singer
is bellowing as loud as he can, you're running at what you think is a
safe margin below clipping of the preamp, say 6 dB.

Now, if the preamp is putting out +18 dBu with that bellowing singer,
it can put out +24 dBu before it clips. But if your sound card has no
input adjustment but a +4/-10 switch, and IT reaches full scale at
+19 dBu like your RME card does, you have only 1 dB of headroom.
Ideally, you'd want to attenuate the input to the sound card a bit so
the preamp can go its full range without the recorder clipping, but
there's no knob for that. (remember when recorders used to have knobs
or trimpots for record level?) So you do the next best thing and back
off on the preamp gain to leave yourself some headroom on the digital
side, but that possibly compromises the S/N ratio.

In real life, you don't ponder these things, you do what works.


--
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
January 16, 2005 2:36:59 AM

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

On Sat, 15 Jan 2005 22:38:05 +0100, Bob Cain wrote:
> Chel van Gennip wrote:
>
>> Using the bits you paid for to get some headroom is using these bits
>> well.
>
> Hear, hear!
>
> Do you pre jocks know what the best (and the pretty darned good and the
> fair to middlin) SNR that can be obtained at 48 dB of gain? I'm curious
> how many real bits a "24 bit" A/D must have to be equivalent so as to
> eliminate the need for a pre.

The theoretical minimum EIN is about -129dBu I think, so to get to 0dBu
about 22 bits would do.

There is also a limit to the maximum achievable dynamic range for
playback. The difference between the minimal sound level of a room with
someone breezing and the sound level of permanent ear damage can be
achieved with the 16 bits of a normal CD.

But not all ears are the same, I've met engineers who claim they can
do a perfect soundmix while listening at a 60dB (unweighted) soundlevel.
I have troubles hearing low bass at that level, they don't

--
Chel van Gennip
Visit Serg van Gennip's site http://www.serg.vangennip.com
Anonymous
January 16, 2005 1:16:31 PM

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

On Sun, 16 Jan 2005 04:29:14 +0100, Mike Rivers wrote:

> So you do the next best thing and back off on the preamp gain to leave
> yourself some headroom on the digital side, but that possibly
> compromises the S/N ratio.
>
> In real life, you don't ponder these things, you do what works.

Headroom and S/N ratio are a compromise. You should include microphone
noise, both electrical and acoustical, in your model. With current state
of technology S/N does not has to be a big problem in real life. The
upper limmit of sound in a digital system is quite hard and quite
audible, so I prefer to be on the safe side. That includes some headroom.
That won't get you a 110dB S/N recording, but why should you want that.

--
Chel van Gennip
Visit Serg van Gennip's site http://www.serg.vangennip.com
Anonymous
January 16, 2005 1:40:46 PM

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

In article <34upnfF4f9aksU1@individual.net> chel@vangennip.nl writes:

> With current state
> of technology S/N does not has to be a big problem in real life. The
> upper limmit of sound in a digital system is quite hard and quite
> audible, so I prefer to be on the safe side. That includes some headroom.
> That won't get you a 110dB S/N recording, but why should you want that.

No need for that. The preceived need is to not have to adjust the
volume control to match the last CD played through the system. Noise
doesn't enter into the minds of people who only think that way other
than the concern that "If I boost the gain digitally I'll be
amplifying the noise and I don't want to do that."

They shouldn't be worried about amplifying the noise in the mic
preamp, they should be worried about amplifying the TV set playing in
the other room that's on their recording because they don't have good
sound isolation.



--
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
January 17, 2005 9:59:16 AM

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

"Chel van Gennip" <chel@vangennip.nl> wrote in message
news:34svmjF4bfce9U1@individual.net


> What is "too low a level"?

Something that puts the noise floor of the source below or near the noise
floor of the recorder.

>It is clear what is "too hig a level.

Except with analog tape ;-)

>With 24 bits you get 20 bits or more, you have the room to keep it safe.

Letsee, the room is someplace below or near 70 dB dynamic range, the mic
might be as good as 80 dB, the console could be 90 dB, and just about every
modern audio interface is up around 100 dB.

> Staying 6 or even 12 dB from the hard high limit will cost you 1 or 2
> of these bits you payed for.

Yes, but see my real-world dynamic range budget, above.

> Sometimes the best performances are a
> bit louder than expected in advance, so why throw those away and
> spend extra time for a second take.

Agreed. Depending on where you want to set your margins, there is from 10 to
30 dB of dynamic range in the audio interface to umm, budget.

>Riding the gain during recording
> sessions normally is too late and even when you are in time it is not
> making things much better.

Totally agreed. Riding gain is why God invented gain enveloping. Do it after
the fact when you get as many chances as you need, and you are never
surprised.

>Using the bits you paid for to get some headroom is using these bits well.

Totally agreed.
Anonymous
January 17, 2005 10:01:54 AM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1105814702k@trad
> In article <34svmjF4bfce9U1@individual.net> chel@vangennip.nl writes:
>
>> What is "too low a level"?
>
> For me, it's when the meter never gets above -20 dBFS. But there are
> many people who believe that if it never hits 0 dBFS, and in fact ever
> drops below -10 dBFS unless the music stops, the "recording isn't hot
> enough." I think it's a product of looking at waveforms.

I agree. It's a sad fact of life that with alot of displays, if you are
looking at a bunch of tracks and you've got 10 dB or more of headroom, you
are looking at what appear to be flat lines.

> -6 dB is half way to full scale on most waveform displays, so even a
> recording that
> peaks at -6 looks like it's not even close to filling the window. So
> many people are tempted to turn it up.

It's like we need some kind of a logarithmic display for DAWs.

> It's a hard sell, though. Too many people either remember or read
> about 16-bit technology where you only had about 12 reliable bits to
> start with. There, it was more important to keep close to the peak
> level.

Gladly those days are gone.

> The other thing is that people stick a CD in the drive on the
> computer, start it playing, and set the playback volume to that
> reference, then play one of their own recordings and it's much
> quieter. Of course they can't be bothered with turning up the playback
> volume so they assume they need to record at a higher level.

As a practical matter, if your peaks don't get within a few dB or FS,
someone is going to complain about "...having to turn the volume up to hear
anything."
Anonymous
January 17, 2005 10:28:53 AM

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

"Bob Cain" <arcane@arcanemethods.com> wrote in message
news:csc2fv0vj4@enews2.newsguy.com
> Chel van Gennip wrote:
>
>> Using the bits you paid for to get some headroom is using these bits
>> well.
>
> Hear, hear!
>
> Do you pre jocks know what the best (and the pretty darned
> good and the fair to middlin) SNR that can be obtained at 48
> dB of gain? I'm curious how many real bits a "24 bit" A/D
> must have to be equivalent so as to eliminate the need for a
> pre.

Usually I end up giving dyamic mics about 30 dB gain, and dynamics more like
20. 30 dB gain is like 5-6 bits and 20 is like 3-4. If you line up 14 bits
ideally, you have pretty good audio. You need 10-20 dB headroom which is 3-4
bits. Therfore, a converter with 14+4+6 or 26 bits might work pretty well
in general without a mic preamp.

For reference purposes
http://www.neumann.com/infopool/mics/produkte_sol_d.php...
claims

"The result is an internal digital 28-bit signal with a dynamic range of
more than 130 dB (A-weighted, incl. microphone capsule)."
Anonymous
January 17, 2005 10:38:59 AM

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

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1105805769k@trad
> In article <26mdnTLAccLjlXTcRVn-ow@comcast.com> arnyk@hotpop.com
> writes:
>
>> According to http://www.mackie.com/pdf/sr24vlzpro_ss.pdf , the
>> insert outs of a SR32 have low distortion at +14. IOW, they are not
>> clipping. The least sensitive pro sound card I know of clips with
>> an input of +12, and another with +10.

Brain fart time. Let me amend that to 10-12 dBu over +4 dBu for +14 +16 dBu
for peak output.

> The Mackie recorder with analog I/O clips at +24 dBu. I was under the
> impression that most sound "professional" sound cards ran somewhere in
> the ballpark of +16 to +16 dBu for 0 dBFS.

Your 16 dBu meets my 16 dBu

> I would expect "consumer" and "-10" cards to reach full scale in the +10
> to +12 dBV range.

True consumer sound cards put FS at 1 volt (AC97) or 2 volts (Creative Labs
until lately)

I would expect a -10 dBv card to have 10-12 dB headroom at most which puts
FS at 0 to 2 dBv. There's about 2 dB between 0 dBu and 0 dBv, so that would
be +2 to +4 dBu.

> A lot of people are unhappy with recordings that rarely go above -6 dBFS
> because the waveform display looks so wimpy.

Agreed, we just covered that in another thread.

> You and I are not.

Something about hatred of the sound of clipping. ;-)

> I regularly run the Mackie recorder below -10 dBFS because that's where
> my console is comfortable.

I think that the levels through my church's SR32 are pretty close to
standardized. In that context I record with Delta cards on the inserts,
running in +4 mode. Most of my recordings come into editing with at least 10
dB worth of headroom unless someone gets *really* excited or things wander
out of control, technically. It's been along time since I've seen any
clipping. The noise floor of finished recordings is dominated by
environmental acoustical noises. I think that is about as good as it gets
in my recording context until I get the audience and musicians to stop
breathing and pumping blood through their bodies! ;-)
Anonymous
January 18, 2005 9:25:45 AM

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

Arny Krueger wrote:

> Why would one use *anything* as a volume control while recording?

Well, if you have the chart in front of you and the singer on the other
side of the glass, you can ride gain more intelligently than any
compressor or limiter. The skill to get it right _on the way in_ is one
not well understood by the casual recordist or anyone who has been
relying on fixing it after the fact. But once upon a time that was a
sign of serious recording chops.

--
ha
Anonymous
January 18, 2005 9:25:46 AM

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

Scott Dorsey wrote:

> Chel van Gennip wrote:
> > Arny Krueger wrote:

> >>> But if you're stuck at +4 dBu nominal level, you probably won't reach
> >>> full scale before the
> >>> preamp clips.

> >> According to http://www.mackie.com/pdf/sr24vlzpro_ss.pdf , the insert
> >> outs of a SR32 have low distortion at +14. IOW, they are not clipping.
> >> The least sensitive pro sound card I know of clips with an input of +12,
> >> and another with +10.

> >The correct specification for the Mackie 1202 vlz pro is at
> >http://www.mackie.com/pdf/1202vlzpro_ss.pdf
> >They claim a 22db headroom above 0dBu.
> >I always use a +4 dBu setting for my soundcards with my Mackie.

> I wouldn't believe the spec. The Mackie starts to sound funny, long before it
> actually reaches clipping. It can help a lot to run it with lower operating
> levels.

One must keep in mind that Mackie's ")dB" is actually that instead of
the oft used ) dB = + 4 dB.

--
ha
!