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Mike Rivers[_2_] Mike Rivers[_2_] is offline
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Default Who Owns the Behringer DEQ2496?

On 8/13/2014 12:14 AM, Gary Eickmeier wrote:

I am not an electronics technician but I do have a RS Sound Level Meter and
a little volt meter. But my question doesn't call for all of this
measurement that you guys are talking about.


"All this measurement" is really very basic. The reason what it was
recommended to you was to eliminate one possible cause for the
deficiencies that you're hearing with the DEQ in line with your signal
path. It isn't a straight wire, and there's no button to turn it into
one. If it changes the level to your speakers even a small amount, it
can affect what you hear.

But you have tossed that suggestion off without investigating it.

All I wanted to know was what
the translated German manual was trying to tell us about that little switch.


And you got it, in several versions, many times over. But since it
doesn't do what you want it to do, you've brushed those explanations
off. You've been pointed to at least one web article that explains the
relationship between gain and operating levels but you said you didn't
need to know that.

Some of you who own the device would probably know. I learned that it
probably pads down the output by 10 dB so that you can input more gain and
use more of the AD converter's bits.


That's bassackwards.

Paul is trying to communicate to me about the well-known level difference
problem in A/B comparisons. But this is not an A/B direct comparison, and my
supposed sound quality difference would have nothing to do with that.


But your initial complaint was that when you connect the DEQ, you heard
several things that were different from when it wasn't connected. A
direct A/B comparison is difficult to make without a "true bypass"
switch (which you could build pretty easily, by the way) but you could
make a measurement that would explain at least one difference between in
and out, and furthermore, could explain the difference you believe
you're hearing.

But long term (meaning more than about 15 seconds) aural memory isn't
very reliable. You may be hearing a problem because you THINK there is one.

More probable - if there is a
difference - is that the AD converter isn't getting a healthy enough signal
to operate with, hence my question. Learned lots here as usual.


What have you learned? If you learned anything from this aspect of the
discussion, it should have been that this "healthy enough signal"
business is not part of the problem. But still, you stick to what's
"probable" to you.

My current project is to evaluate my new speakers that I had built by a very
talented man in Indiana. If I bring in some other, more commercial speakers
to compare them with, I might want to EQ them to the same standard to take
that factor out of the comparison.


Why are you EQ-ing speakers at all if you're comparing them?




--
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Default Who Owns the Behringer DEQ2496?

On 13/08/2014 13:10, Mike Rivers wrote:
My current project is to evaluate my new speakers that I had built by
a very
talented man in Indiana. If I bring in some other, more commercial
speakers
to compare them with, I might want to EQ them to the same standard to
take
that factor out of the comparison.


Why are you EQ-ing speakers at all if you're comparing them?

And, bearing in mind a lot of the differences between speakers are in
the time domain, not the frequency domain and so can't be equalised out,
why bother?

I'll equalise an installation to get the best sound possible, but that's
only half the answer IME.


--
Tciao for Now!

John.
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Default Who Owns the Behringer DEQ2496?

None wrote:

"Gary Eickmeier" wrote in message
...
You might get a better understanding of levels and how they're
measured by reading the Meter Madness article on my web site.


I would love to Mike, but I think after doing this stuff for some 60
years I have enough understanding of levels to get along.


Whooooosh!


This could be a record setting event. Frisbies in the dark during a luny
eclipse.

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Default Who Owns the Behringer DEQ2496?

Frank Stearns wrote:

Knowing Behringer, probably not very well -- or when they copied someone's
design they cut every possible corner in power supply quality, board
quality, connector quality, caps, etc, etc. So I'd be nervous too! And
these days, I probably wouldn't worry as much about the converters as the
analog signal paths to and from those converters.


I repeat, the quality of the DEQ2496 and companion crossover DCX2496 is
very much better than the older analog kit.

The problems in this thread have not to do with the device.

--
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Default Who Owns the Behringer DEQ2496?

Scott Dorsey wrote:

Gary Eickmeier wrote:
So my main question would be: Whenever we have a digital component in the
system, especially a digital recorder, we try and make best use of available
bits. But the analog inputs to the unit from my receiver were variable IAW
the volume knob. So how did I ever know that I was using all of the bits in
the equalizer? The output meter usually barely moved. The input I doubt had
enough gain for 16 bits to be filled up.


Where did you get this idea?

What makes you think "using all the bits" is significant or even useful?

Do you always operate your power amplifier at full tilt for fear you're
not getting all the output ower it's capable of?
--scott


Well, ****, you're paying for the electricity, so why not use all of
it??

--
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Default Who Owns the Behringer DEQ2496?

PStamler wrote:

I'm not Scott, and can't speak for him, but I often make recordings where
the highest level is -20dBFS, and they sound fine. As they ought to,
since the theoretical dynamic range of a 24-bit system under those
circumstances is 124dB, meaning that whatever I'm recording is well
above the muck and mud level.


That, plus easing strain on the analog chain feeding conversion.

We need enough level throughout to avoid the noise floor, but that is
easy with 24 bits unless something up stream is broken or poorly
adjusted. I've been looking for -20dBFS average and peaks not above
-12dBFS. The results leave lots of room for mixing and mastering, and
avoids the "every last bit as loud as can be" syndrome from the gitgo.

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Default Who Owns the Behringer DEQ2496?

Gary Eickmeier wrote:

All I wanted to know was what
the translated German manual was trying to tell us about that little switch.


You posted all that crap to ask THAT? What are you thinking?

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PStamler wrote:

you're simply fumbling in the dark


That would be a step up.

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Default Who Owns the Behringer DEQ2496?

Mike Rivers wrote:

On 8/13/2014 12:14 AM, Gary Eickmeier wrote:

I am not an electronics technician but I do have a RS Sound Level Meter and
a little volt meter. But my question doesn't call for all of this
measurement that you guys are talking about.


"All this measurement" is really very basic. The reason what it was
recommended to you was to eliminate one possible cause for the
deficiencies that you're hearing with the DEQ in line with your signal
path. It isn't a straight wire, and there's no button to turn it into
one. If it changes the level to your speakers even a small amount, it
can affect what you hear.

But you have tossed that suggestion off without investigating it.


Hmmm€¦ Where have I seen that before?

All I wanted to know was what
the translated German manual was trying to tell us about that little switch.


And you got it, in several versions, many times over. But since it
doesn't do what you want it to do, you've brushed those explanations
off. You've been pointed to at least one web article that explains the
relationship between gain and operating levels but you said you didn't
need to know that.

Some of you who own the device would probably know. I learned that it
probably pads down the output by 10 dB so that you can input more gain and
use more of the AD converter's bits.


That's bassackwards.


Dude, retro is where it's at these days.

Paul is trying to communicate to me about the well-known level difference
problem in A/B comparisons. But this is not an A/B direct comparison, and my
supposed sound quality difference would have nothing to do with that.


But your initial complaint was that when you connect the DEQ, you heard
several things that were different from when it wasn't connected. A
direct A/B comparison is difficult to make without a "true bypass"
switch (which you could build pretty easily, by the way) but you could
make a measurement that would explain at least one difference between in
and out, and furthermore, could explain the difference you believe
you're hearing.

But long term (meaning more than about 15 seconds) aural memory isn't
very reliable. You may be hearing a problem because you THINK there is one.


Did the Internet report a problem with the sound of Gary's sytem?

More probable - if there is a
difference - is that the AD converter isn't getting a healthy enough signal
to operate with, hence my question. Learned lots here as usual.


What have you learned? If you learned anything from this aspect of the
discussion, it should have been that this "healthy enough signal"
business is not part of the problem. But still, you stick to what's
"probable" to you.


Amazing feats of illogic.

My current project is to evaluate my new speakers that I had built by a very
talented man in Indiana. If I bring in some other, more commercial speakers
to compare them with, I might want to EQ them to the same standard to take
that factor out of the comparison.


Why are you EQ-ing speakers at all if you're comparing them?


Oh, you just had to go there, didn't you!?!

--
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Default Who Owns the Behringer DEQ2496?

Sean Conolly wrote:

"hank alrich" wrote in message
...
Gary Eickmeier wrote:

I played with the 2496 long enough to learn all of its functions and
features, did a few EQs with it, measured my speakers etc etc, but I got
a
little nervous about having an additional A/D D/A in my system so I
yanked
it out of there and the sound seems a lot better now, tighter, more
together, larger soundstage, etc.


Man, you're just so sciency about all this. NOT!

So my main question would be:


Step 1: Gain staging?

"It's not about the knobs so much as it is about the person twisting
them"

Whenever we have a digital component in the
system, especially a digital recorder, we try and make best use of
available
bits. But the analog inputs to the unit from my receiver were variable
IAW
the volume knob. So how did I ever know that I was using all of the bits
in
the equalizer? The output meter usually barely moved. The input I doubt
had
enough gain for 16 bits to be filled up.


Input level sensitivity.

So would Behringer have a way of working around these problems or am I
correct in being nervous about it?


The unit is much more powerful than one might expect at first glance,
and some sectors offer significant potential newbie danger, particularly
the parametric and dynamic EQ sections. You're thinking "digital" and
"bits" but you're using the analog I/O of the equalizer. Get levels in
order and listen again.

BTW the metering is quite good. I have used the DEQ2496's for FOH and
stage monitor EQ and the peak reading 31 band meter will show me the
next potential feedback point before the system starts ringing. Pretty
cool. These shows were all acoustic sources with the occasional tiny
bass amp.

Is it a Weiss? No. The sales tax on a Weiss is more than the cost of the
Behringer.


I should feel like a dumb-ass for having one of these for years and not
knowing some of these functions, but a lot of this falls under 'problems I
don't have'.

I've never needed to switch the levels because everything I own uses +4.

I never use the auto-eq function because I don't need it at home, and it's a
time-soak at a show. I've used swept tone tests to find a good flattish
curve for the speakers I have and use that as a starting point. If I'm using
someone else's speakers I prefer to start flat and EQ by ear.

I never use the feedback eliminator because it's usually too little and too
late. I use the RTA display to look for resonant frequencies and notch those
with the parametric as needed.

I guess this makes me a happy user but not an expert.

Sean


Sean, what it makes you is one experienced enough to avoid the automatic
functions and use the device under your own control in real time. Works
great that way. That's how I use it, mostly. I had to dig into some of
the other features for installations, where memories, lockdown, feedback
suppression for dance teacher's or preacher's head mics, etc., were
important.

--
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Default Who Owns the Behringer DEQ2496?

On 14/08/2014 12:25 a.m., John Williamson wrote:
On 13/08/2014 13:10, Mike Rivers wrote:
My current project is to evaluate my new speakers that I had built by
a very
talented man in Indiana. If I bring in some other, more commercial
speakers
to compare them with, I might want to EQ them to the same standard to
take
that factor out of the comparison.


Why are you EQ-ing speakers at all if you're comparing them?

And, bearing in mind a lot of the differences between speakers are in
the time domain, not the frequency domain and so can't be equalised out,
why bother?


Yes they can - DEQX .

geoff

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Gary Eickmeier Gary Eickmeier is offline
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Default Who Owns the Behringer DEQ2496?


"Gary Eickmeier" wrote in message
...
I played with the 2496 long enough to learn all of its functions and
features, did a few EQs with it, measured my speakers etc etc, but I got a
little nervous about having an additional A/D D/A in my system so I yanked
it out of there and the sound seems a lot better now, tighter, more
together, larger soundstage, etc.

So my main question would be: Whenever we have a digital component in the
system, especially a digital recorder, we try and make best use of
available bits. But the analog inputs to the unit from my receiver were
variable IAW the volume knob. So how did I ever know that I was using all
of the bits in the equalizer? The output meter usually barely moved. The
input I doubt had enough gain for 16 bits to be filled up.


Finally had time to re-insert the equalizer and see what the switch does. As
you say, it changes the input level by 10 dB, with compensating padding of
output by the same amount. IOW, nothing changes but the input meters. With a
healthy signal coming in and played loud, my input meters are hitting about
at the -40 level with the switch in the +22 position. With the +12 position
they hit -30, with peaks getting dangerously close to clipping. So I guess
if I keep it in the system I will keep it on the +22 position. I am using
plenty of bits and no danger of clipping. As for sound quality differences,
I will have to listen some more when I am more relaxed, have more time to
evaluate things. As I said, could just be psychoacoustic. Whenever you
change something you may perceive an imaginary improvement.

Thanks,
Gary


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Default Who Owns the Behringer DEQ2496?

"Gary Eickmeier" wrote in message
...

"Gary Eickmeier" wrote in message
...
I played with the 2496 long enough to learn all of its functions and
features, did a few EQs with it, measured my speakers etc etc, but I
got a little nervous about having an additional A/D D/A in my system
so I yanked it out of there and the sound seems a lot better now,
tighter, more together, larger soundstage, etc.

So my main question would be: Whenever we have a digital component
in the system, especially a digital recorder, we try and make best
use of available bits. But the analog inputs to the unit from my
receiver were variable IAW the volume knob. So how did I ever know
that I was using all of the bits in the equalizer? The output meter
usually barely moved. The input I doubt had enough gain for 16 bits
to be filled up.


Finally had time to re-insert the equalizer and see what the switch
does. As you say, it changes the input level by 10 dB, with
compensating padding of output by the same amount. IOW, nothing
changes but the input meters. With a healthy signal coming in and
played loud, my input meters are hitting about at the -40 level with
the switch in the +22 position. With the +12 position they hit -30,
with peaks getting dangerously close to clipping. So I guess if I
keep it in the system I will keep it on the +22 position. I am using
plenty of bits and no danger of clipping. As for sound quality
differences, I will have to listen some more when I am more relaxed,
have more time to evaluate things. As I said, could just be
psychoacoustic. Whenever you change something you may perceive an
imaginary improvement.

Thanks,
Gary


Are you responding to your own post because nobody else will?



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Default Who Owns the Behringer DEQ2496?

On 17-08-2014 03:32, Gary Eickmeier wrote:

Finally had time to re-insert the equalizer and see what the switch does. As
you say, it changes the input level by 10 dB, with compensating padding of
output by the same amount.


Thanks, confirms that the device description by someone is to the point.

IOW, nothing changes but the input meters.


IF nothing changed but the meters it would be a meter sensitivity switch.

With a
healthy signal coming in and played loud, my input meters are hitting about
at the -40 level with the switch in the +22 position. With the +12 position
they hit -30, with peaks getting dangerously close to clipping.


Some of the time one has to pick nits. This is one of those times. If
the meters hit -40 with the switch in the high line level position then
there CAN NOT exist peaks above -30 when in the household line level
position.

Also, what does "dangerously close to clipping" mean?

What is the input sensitivity of your poweramp? - are you running it as
per my recommendations, ie. with attenuated input or are you running it
flat out? - in the latter case you appear to imply that it is smallish
for the task you ask it to do.

So I guess
if I keep it in the system I will keep it on the +22 position. I am using
plenty of bits and no danger of clipping. As for sound quality differences,
I will have to listen some more when I am more relaxed, have more time to
evaluate things. As I said, could just be psychoacoustic. Whenever you
change something you may perceive an imaginary improvement.


When I get me one of those thingies I am going to run it hot and
attenuate its output. There is no reason for me to see - I may end up
standing corrected, but I want to see that correction if any - to
increase its input gain *because* doing that will also increase its
input stage distortion.

Gary


Kind regards

Peter Larsen

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Default Who Owns the Behringer DEQ2496?


"Peter Larsen" wrote in message
k...
On 17-08-2014 03:32, Gary Eickmeier wrote:

Finally had time to re-insert the equalizer and see what the switch does.
As
you say, it changes the input level by 10 dB, with compensating padding
of
output by the same amount.


Thanks, confirms that the device description by someone is to the point.

IOW, nothing changes but the input meters.


IF nothing changed but the meters it would be a meter sensitivity switch.


I'm just reporting what happens.


With a
healthy signal coming in and played loud, my input meters are hitting
about
at the -40 level with the switch in the +22 position. With the +12
position
they hit -30, with peaks getting dangerously close to clipping.


Some of the time one has to pick nits. This is one of those times. If the
meters hit -40 with the switch in the high line level position then there
CAN NOT exist peaks above -30 when in the household line level position.


I'm just reporting what happens.


Also, what does "dangerously close to clipping" mean?


In the particular recording that I was using, it hit peaks of -13. On
another recording I could foresee it clipping easily. Clipping is bad.

What is the input sensitivity of your poweramp? - are you running it as
per my recommendations, ie. with attenuated input or are you running it
flat out? - in the latter case you appear to imply that it is smallish for
the task you ask it to do.


Just some Carver M 1.5s.

So I guess
if I keep it in the system I will keep it on the +22 position. I am using
plenty of bits and no danger of clipping. As for sound quality
differences,
I will have to listen some more when I am more relaxed, have more time to
evaluate things. As I said, could just be psychoacoustic. Whenever you
change something you may perceive an imaginary improvement.


When I get me one of those thingies I am going to run it hot and attenuate
its output. There is no reason for me to see - I may end up standing
corrected, but I want to see that correction if any - to increase its
input gain *because* doing that will also increase its input stage
distortion.


Then just keep it at the +22 position. Behringer thought of everyithing.

Gary


Kind regards

Peter Larsen






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Default Who Owns the Behringer DEQ2496?

"Gary Eickmeier" wrote in message
...

"Peter Larsen" wrote in message
k...
On 17-08-2014 03:32, Gary Eickmeier wrote:

Finally had time to re-insert the equalizer and see what the
switch does. As
you say, it changes the input level by 10 dB, with compensating
padding of
output by the same amount.


Thanks, confirms that the device description by someone is to the
point.

IOW, nothing changes but the input meters.


IF nothing changed but the meters it would be a meter sensitivity
switch.


I'm just reporting what happens.


You're just squawking about what you observe with your eyes and mind
closed. When anyone tries to tell you how it works (or point you to a
source such as a text), you're too busy squawking about what you
observe with your eyes and mind closed, and you insist that you have
no time to actually understand.

Do you think anyone who knows how it works has any interest in your
reporting of your limited observations?

Then just keep it at the +22 position. Behringer thought of
everyithing.


Yeah, that's certainly easier than actually learning something.


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Default Who Owns the Behringer DEQ2496?

PStamler wrote: "You really don't want to be "filling up" the bits; that means you're pushing the edge of digital clipping, which sounds horrible. "

When you say it, you get praise. When I say it, I get flamed! WTF?!

Agreed Paul, and I wish this F~~~ING "use all the bits" mentality would disappear completely from mother Earth ONCE AND FOR FRICKIN' ALL!!
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wrote:
PStamler wrote: "You really don't want to be "filling up" the bits; that means you're pushing the edge of digital clipping, which sounds horrible. "

When you say it, you get praise. When I say it, I get flamed! WTF?!


Because you're saying two totally different things. Moving the reference
level up and down does not necessarily cause clipping and is totally
independent of compression.
--scott

--
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geoff wrote: "Because you clear have no understanding that "using all the bits" has
nothing to do with loudness, and does not inherently have anything to do "

A 16bit file of a 1kHz sine at full scale is using all 16bits.

A 16bit file of a 1kHz sine peaking at -12dBfs is effectively a 14bit recording.

All things equal at playback, the former of the above two will be LOUDER. Simple common sense.

Bits determine total AMPLITUDE(absolute dynamic range) between noise floor and full scale.


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"Peter Larsen" wrote in message
k...
On 17-08-2014 03:32, Gary Eickmeier wrote:
When I get me one of those thingies I am going to run it hot and attenuate
its output. There is no reason for me to see - I may end up standing
corrected, but I want to see that correction if any - to increase its
input gain *because* doing that will also increase its input stage
distortion.


Maybe, and maybe not. A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.

You just need the right level, not too low and not too high.

Sean


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wrote in message
...
geoff wrote: "Because you clear have no understanding that "using all the
bits" has
nothing to do with loudness, and does not inherently have anything to do "

A 16bit file of a 1kHz sine at full scale is using all 16bits.

A 16bit file of a 1kHz sine peaking at -12dBfs is effectively a 14bit
recording.

All things equal at playback, the former of the above two will be LOUDER.
Simple common sense.

Bits determine total AMPLITUDE(absolute dynamic range) between noise floor
and full scale.


OK OK, this group seems to get hold of a topic - or even a sentence - like a
junkyard dog and tear it to "bits."

All I meant by fill up the bits was the commonly understood principle of
using the full dynamic range of whatever AD processor we are talking about.
If you are recording, you generally want the peaks to go as high as possible
without going over, just like in the analog days. Using 24 bit converters
maybe makes the task easier and gives us more headroom, but within reason,
we want to put a good strong signal through the processor lest the LSB get
lost in noise.

Right?

Gary


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"Sean Conolly" wrote in message
...
"Peter Larsen" wrote in message
k...
On 17-08-2014 03:32, Gary Eickmeier wrote:
When I get me one of those thingies I am going to run it hot and
attenuate its output. There is no reason for me to see - I may end up
standing corrected, but I want to see that correction if any - to
increase its input gain *because* doing that will also increase its input
stage distortion.


Maybe, and maybe not. A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.

You just need the right level, not too low and not too high.

Sean


Reasonable! Sean, which position of the switch do you use? Was my
description of what it does correct? I need to play with it just a little
more, maybe this weekend. Also need to measure the loudness with it in and
out of the system, as Paul suggests. There may be a difference, even tho
there shouldn't be.

Gary


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Sean Conolly Sean Conolly is offline
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Default Who Owns the Behringer DEQ2496?

"Gary Eickmeier" wrote in message
...

"Sean Conolly" wrote in message
...
"Peter Larsen" wrote in message
k...
On 17-08-2014 03:32, Gary Eickmeier wrote:
When I get me one of those thingies I am going to run it hot and
attenuate its output. There is no reason for me to see - I may end up
standing corrected, but I want to see that correction if any - to
increase its input gain *because* doing that will also increase its
input stage distortion.


Maybe, and maybe not. A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.

You just need the right level, not too low and not too high.

Sean


Reasonable! Sean, which position of the switch do you use? Was my
description of what it does correct? I need to play with it just a little
more, maybe this weekend. Also need to measure the loudness with it in and
out of the system, as Paul suggests. There may be a difference, even tho
there shouldn't be.


I never even knew the switch wasn't for the RTA mic until this thread
started! But all of my gear runs at 'pro' level so the default setting is
fine. I can also patch it into my DAW using the AES jacks to bypass the
analog stages entirely if needed.

And no, it's not just the meters that are changing. The actual conversion
level is being shifted by 10db so maybe a little less noise on quiet
signals, and easier to clip on louder signals.

Try both, and listen carefully to the difference. I would not be suprised if
you don't hear any, and that you still hear a subtle difference if you take
it out of the chain. It's not 'audiophile' grade gear, it's a good piece of
'pro-sumer' gear - better than most home stereo equipment but still has some
tradeoffs to meet a price point.

The real value of the unit is the stuff between the converters, so if you
have a digital source like S/PDIF from a CD player that would be good to
try.

Sean


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On Tuesday, August 19, 2014 7:34:35 PM UTC-6, Sean Conolly wrote:
A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.


I expected that to be the case when I began reading Samuel Groner's exhaustive tests of opamps; I was surprised to find that it wasn't so. A few of the circuits he tested showed crossover distortion, but the majority didn't.

Peace,
Paul


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On 20-08-2014 02:34, Sean Conolly wrote:

"Peter Larsen" wrote in message
k...
On 17-08-2014 03:32, Gary Eickmeier wrote:
When I get me one of those thingies I am going to run it hot and attenuate
its output. There is no reason for me to see - I may end up standing
corrected, but I want to see that correction if any - to increase its
input gain *because* doing that will also increase its input stage
distortion.


Maybe, and maybe not. A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.


You just need the right level, not too low and not too high.


Yes. I have twice experienced more signal though a gizmo being a clear
advantage, first as previously mentioned with an electronic x-over when
it had LF356'es in it - 15 dB more signal = 15 dB less noise and cleaner
treble - and later with my SV3800, with that one it probably was/is a
converter linearity issue.

Also there are marginal circuitry out there that benefits from running
with max negative feedback, my MR8HD is an example, it is actually very
good when running with input gain low and sounds like plastic with it
set high, so using an external mic pre is a good choice with it.

Knowing that there are people out there actually making a living - or
part of a living - from selling new analog boards to the DCX - for me
makes it a good first hypothesis that the Behringer 1 rack unit DCX and
DEQ boxes should be run as hot as possible, albeit of course allowing
proper operational headroom. This also because audible noise has been
reported in home use context with household type power amplifier input
sensitivity.

Digital in and if possible out may be a good strategy, even just digital
in will take one set of mediocre opamps out of the circuit.

Having both is probably also a good idea for loudspeaker experiments.

Sean


Kind regards

Peter Larsen


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This (L)east (S)ignificant (B)it: Done some reading up on it and all I need is a simple answer: Is the lsb at the top(near full scale) or at the bottom, amplitude-wise?
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wrote in message
...
This (L)east (S)ignificant (B)it: Done some reading up on it and
all I need is a simple answer: Is the lsb at the top(near full
scale) or at the bottom, amplitude-wise?


No.

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Sean Conolly wrote:
"Peter Larsen" wrote in message
. dk...
On 17-08-2014 03:32, Gary Eickmeier wrote:
When I get me one of those thingies I am going to run it hot and attenuate
its output. There is no reason for me to see - I may end up standing
corrected, but I want to see that correction if any - to increase its
input gain *because* doing that will also increase its input stage
distortion.


Maybe, and maybe not. A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.


There aren't so many of these left any more. This was a very popular
distortion source with early op-amps, and it's still the primary distortion
source of the INA103. This is why some of the popular Walt Jung modifications
involved adding DC bias to the output to turn off one of the output
transistors and shift the whole thing into class A single-transistor operation
at the expense of voltage swing and current drive.

These days this is a much more rare thing, although it can be still found
now and then. The good news is that most datasheets will have a plot of
output swing vs. distortion so you can be forewarned.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."


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Gary Eickmeier wrote:
All I meant by fill up the bits was the commonly understood principle of
using the full dynamic range of whatever AD processor we are talking about.
If you are recording, you generally want the peaks to go as high as possible
without going over, just like in the analog days. Using 24 bit converters
maybe makes the task easier and gives us more headroom, but within reason,
we want to put a good strong signal through the processor lest the LSB get
lost in noise.


Correct. What you're talking about has nothing to do with linearity, only
with gain reference level, and there is one unfortunately frequent poster
who does not understand the difference between these.

However, we do live in the 21st century where it's common for the noise floor
to be well below -100dBFS, so the need to worry about working as close as
possible to the limit no longer exists except for the final release. It is
very common for people to be recording with peaks at -20dBFS in order to
have safety margin today. You could not do that with 1/4", even with Dolby A.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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PStamler wrote:
On Tuesday, August 19, 2014 7:34:35 PM UTC-6, Sean Conolly wrote:
A lot of opamps with will have a tiny amount of
crossover distortion on the output, but at a fixed level. The distortion
percentage goes down as the output signal increases, until some other
mechanism dominates the distortion.


I expected that to be the case when I began reading Samuel Groner's exhaustive tests of opamps; I was surprised to find that it wasn't so. A few of the circuits he tested showed crossover distortion, but the majority didn't.


And let me guess, the ones that showed a lot of crossover distortion were
video op-amps designed for driving 75 ohm loads?
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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Scott Dorsey wrote:

wrote: PStamler wrote: "You really don't want
to be "filling up" the bits; that means you're pushing the edge of digital
clipping, which sounds horrible. "

When you say it, you get praise. When I say it, I get flamed! WTF?!


WTF = Way Too Funny!

Because you're saying two totally different things. Moving the reference
level up and down does not necessarily cause clipping and is totally
independent of compression.
--scott


Did you hear the one about the pinhead who uses Vise-Grips for hatbands?

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic
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On Wednesday, August 20, 2014 8:46:49 AM UTC-6, Scott Dorsey wrote:
PStamler wrote:

On Tuesday, August 19, 2014 7:34:35 PM UTC-6, Sean Conolly wrote:


A lot of opamps with will have a tiny amount of


crossover distortion on the output, but at a fixed level. The distortion


percentage goes down as the output signal increases, until some other


mechanism dominates the distortion.




I expected that to be the case when I began reading Samuel Groner's exhaustive tests of opamps; I was surprised to find that it wasn't so. A few of the circuits he tested showed crossover distortion, but the majority didn't.




And let me guess, the ones that showed a lot of crossover distortion were
video op-amps designed for driving 75 ohm loads?


Actually no. The two chips that showed crossover distortion significant enough to mention were the LT1007 and LT1632, neither of which seems to be designed for video use.

BTW, if anyone wants to download this extremely useful set of tests, it can be found at:

http://tinyurl.com/opamptests

Peace,
Paul
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On 20/08/2014 11:33 p.m., wrote:
This (L)east (S)ignificant (B)it: Done some reading up on it and all I need is a simple answer: Is the lsb at the top(near full scale) or at the bottom, amplitude-wise?


You needed to read up on it ?!!!


geoff


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