Who Owns the Behringer DEQ2496?

[Gary Eickmeier]

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.

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

Gary Eickmeier

--
Gary Eickmeier

[PStamler]

On Saturday, August 9, 2014 6:08:54 PM UTC-6, 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.
>
>
>
> 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.

A question and a couple of comments. First, did you measure the system's gain with and without the DEQ in circuit? If it's introducing a few dB of loss, that would account for the subjective impression of poorer sound. One way to do this is to have the DEQ set flat, play back a 1kHz test tone, and measure the voltage at the speaker terminals. Remove the DEQ from the chain and do the measurement again.

Second, the DEQ *does* have an input gain control, does it not? If that's set so low that the meter is barely moving, you indeed have a problem with gain staging; the output from a consumer hi-fi receiver is less than it's designed to operate from, and you could set the gain control so you see a little more action from the meters. On the other hand, this is a 24-bit device; you can get away with low levels and still maintain linearity and low noise.

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

Set the input gain so the system gain is the same with the DEQ in and out of the circuit, and you should still be okay. Or turn up the gain so that the meters peak at about -12dBFS; that should get you way too much signal to the power amp, but if the latter has a level control you can turn it down again.

By the way, you wrote "But the analog inputs to the unit from my receiver were variable IAW the volume knob."

What does IAW mean?

Peace,
Paul

[Sean Conolly]

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

Hey there Gary,

If you're running a *very* low signal through the unit I could see where
quantization noise could start becoming a factor. I recall your receiver is
running at -10 consumer levels? As far as I know the DEQ only runs at +4,
and although you can adjust the volume it's a digital control upstream of
the D/A converter.

Buy or build a pad for the output, so you can run the output meters close to
50% and see if that helps.

Sean

[Frank Stearns]

"Gary Eickmeier" > writes:

>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.

Not surprising at all.

>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.

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

Depends on how they designed and built it. 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've found the very best room EQ to be proper acoustic treatment -- control the bass
and reflections, add diffusion and small amounts of very carefully selected
reflection as needed; with those elements develop a reverb curve suitable for the
space, etc.

The results can be stunning. In fact, the new room here is nearly done. The mid and
top end has never sounded so good. The 20-40 hz range appears to be dead flat.
There's just a slight amount of murkiness in the 100-150 range; I'm halfway through
adding a set of traps tuned to this and will know soon whether a diaphragm
density change might be also appropriate for some of the other traps.

Maybe you need room EQ, maybe you don't. But do as much of the mechanical/phsyical
"EQ" first.

Frank
Mobile Audio
--

[geoff]

On 10/08/2014 12:08 p.m., 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.
>
> 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.
>
> So would Behringer have a way of working around these problems or am I
> correct in being nervous about it?
>
> Gary Eickmeier
>

It is unlikely the unit itself stuffed anything up - more likely your
settings o some of the effects modules, or imagination.

There was some story back in 44k1/16bit days of a panel of people set up
with a live feed versus the same source with 10 AD-DA devices in series.
And they couldn't reliably tell the difference. the 2496 is years ahead
of any of those.


geoff

[geoff]

On 10/08/2014 6:04 p.m., Frank Stearns wrote:
> "Gary Eickmeier" > writes:

>
> Depends on how they designed and built it. 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.


Not correct Frank. It it one of the Behringer 'goodies'. Even a relay
'true bypass'.

geoff

[Mike Rivers[_2_]]

On 8/9/2014 8:08 PM, 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.

So your system sounds better with the DEQ out of the signal path even
with all of its settings as bypassed as you can get? That's
disappointing. I wouldn't be surprised that there was a barely
noticeable change for no other reason than that there's another analog
component in the signal path, but you make it sound like a horror show
with it in line.

> 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.

Well, we used to, when there were only about 12 bits that we could count
on being reasonably accurate. With today's 24-bit systems, we prefer
more headroom so that we aren't using most of the bits most of the time.
The positive result is that what's converted between analog to digital
and back is more accurate. The downside is that you end up with an
analog output that may be rather low and you'll amplify the analog noise
when bringing it up to your normal listening level. This is why we have
the concept of "gain structure" - to get things working together
without, as you say, making the best use of the available bits, or
without clipping.

> 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.

Doesn't it have an input level meter? How about a switch for nominal
operating level (-10/+4)? According to the manual, there's a switch to
change the maximum input level of the main XLR inputs from +12 to +22
dBu. Since the meter is showing a low level, put the switch in the +12
position and you'll get 10 dB more gain. Do you have "tape out/in" jacks
somewhere in your system? Tape output is nearly always fixed level,
somewhere near the design center for the unit. Of course it will vary
with the input source, but as long as you aren't clipping the input or
have to turn the volume knob way up to get a usable output level, the
tape output should be around the nominal operating level of the unit.
This, for home stereo gear, is usually in the ballpark of -10 dBV.

Still, the equalizer should be a unity gain device (plus or minus
whatever you do with the equalizer) so what you put in should be what
you get out. But there's all kinds of processing in there. Make sure
none of the dynamics stuff, the compressor, limiter, or dynamic EQ
processing is switched in. You might also try using the auxiliary output
jacks and use the I/O menu to feed those outputs from somewhere earlier
in the chain than what feeds the main outputs.

Now I don't have a DEQ2496, and I've never used one, so don't take this
as defending the product. But lots of people use them effectively, lots
of people just hate everything that Behringer makes. It sounds like
you've at least listened critically to it and can be objective about
whether the overall effectiveness is to make your system sound better or
worse. But explore the options and settings fully before you give up on it.

> So would Behringer have a way of working around these problems

Yes. The input level switch, for one. RTFM

> am I correct in being nervous about it?

You shouldn't be nervous, you should look for the problem and fix it
(which may mean taking the equalizer out of your system) if it bothers you.


--
For a good time, visit http://mikeriversaudio.wordpress.com

[Mike Rivers[_2_]]

On 8/9/2014 8:08 PM, Gary Eickmeier wrote:

> 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.

Again, RTFM and explore the controls. The Meter menu turns the display
into three pages worth of metering options, including showing the input
level.



--
For a good time, visit http://mikeriversaudio.wordpress.com

[Scott Dorsey]

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

[Frank Stearns]

geoff > writes:

>On 10/08/2014 6:04 p.m., Frank Stearns wrote:
>> "Gary Eickmeier" > writes:

>>
>> Depends on how they designed and built it. 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.


>Not correct Frank. It it one of the Behringer 'goodies'. Even a relay
>'true bypass'.

Okay, fair enough. But particularly for audio, there are long life, highly reliable,
high-performance relays that even if seldom actuated, I'd want to be on the
high-performance end just so that the signal flow through the contacts isn't
degraded intermittently. In quantity, the good ones like this are $2-$5 each.

Then there are multi-purpose cheapies that run for pennies each in quantity. Maybe
they're just for switching line AC or other applications where unlike audio,
variances in the "quality" of the close doesn't matter. But you're never sure just
how "tight" the contact close will be, and how that connection will "hold" over time
(seconds, minutes, or days).

It's likely a safe bet which class of relay was used in a device at this price
point.

YMMV.

Frank
Mobile Audio

--

[Gary Eickmeier]

"Frank Stearns" > wrote in message
acquisition...
> "Gary Eickmeier" > writes:
>
>>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.
>
> Not surprising at all.
>
>>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.
>
>>So would Behringer have a way of working around these problems or am I
>>correct in being nervous about it?
>
> Depends on how they designed and built it. 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.

Well... this is (seems to be) a very sophisticated piece of equipment. It
would be a shame if it had less than adequate components and converters in
it.
>
> I've found the very best room EQ to be proper acoustic treatment --
> control the bass
> and reflections, add diffusion and small amounts of very carefully
> selected
> reflection as needed; with those elements develop a reverb curve suitable
> for the
> space, etc.
>
> The results can be stunning. In fact, the new room here is nearly done.
> The mid and
> top end has never sounded so good. The 20-40 hz range appears to be dead
> flat.
> There's just a slight amount of murkiness in the 100-150 range; I'm
> halfway through
> adding a set of traps tuned to this and will know soon whether a diaphragm
> density change might be also appropriate for some of the other traps.
>
> Maybe you need room EQ, maybe you don't. But do as much of the
> mechanical/phsyical
> "EQ" first.
>
> Frank
> Mobile Audio

Yes, and thanks Frank. In fact, that is one of the reasons I pulled it out,
because the EQ it was calling for was barely off flat, as in not EQ'd at
all. I EQ to a room curve that has a lifted bottom and a falling high end.
My new speakers had that - along with the Velodyne - just naturally.

I need to do alot more listening to them but right now I am in hog heaven
without the EQ, so I can still use it for RTA work if I get some comparison
speakers in to test some time, but I just don't need it inserted into my
system any more.

Gary Eickmeier

[Gary Eickmeier]

PStamler wrote:
> A question and a couple of comments. First, did you measure the
> system's gain with and without the DEQ in circuit? If it's
> introducing a few dB of loss, that would account for the subjective
> impression of poorer sound. One way to do this is to have the DEQ set
> flat, play back a 1kHz test tone, and measure the voltage at the
> speaker terminals. Remove the DEQ from the chain and do the
> measurement again.
>
> I confess no I did not.
>
> Second, the DEQ *does* have an input gain control, does it not? If
> that's set so low that the meter is barely moving, you indeed have a
> problem with gain staging; the output from a consumer hi-fi receiver
> is less than it's designed to operate from, and you could set the
> gain control so you see a little more action from the meters. On the
> other hand, this is a 24-bit device; you can get away with low levels
> and still maintain linearity and low noise.

It has a little button called the MAX switch that "raises the maximum level
present at the MAIN inputs/outputs from +12 to +22 dBu." I have no idea what
that does, nor does the book tell me.

> You really don't want to be "filling up" the bits; that means you're
> pushing the edge of digital clipping, which sounds horrible.
>
> Set the input gain so the system gain is the same with the DEQ in and
> out of the circuit, and you should still be okay. Or turn up the gain
> so that the meters peak at about -12dBFS; that should get you way too
> much signal to the power amp, but if the latter has a level control
> you can turn it down again.
>
> By the way, you wrote "But the analog inputs to the unit from my
> receiver were variable IAW the volume knob."
>
> What does IAW mean?

In Accordance With

>
> Peace,
> Paul

[Gary Eickmeier]

Sean Conolly wrote:
> "Gary Eickmeier" > wrote in message
> ...

> Hey there Gary,
>
> If you're running a *very* low signal through the unit I could see
> where quantization noise could start becoming a factor. I recall your
> receiver is running at -10 consumer levels? As far as I know the DEQ
> only runs at +4, and although you can adjust the volume it's a
> digital control upstream of the D/A converter.
>
> Buy or build a pad for the output, so you can run the output meters
> close to 50% and see if that helps.
>
> Sean

That sounds like a great idea Sean. I like the equalizer, but not if it is
going to be a detriment to my sound! I will figure out how to do that and
report back.

Gary

[Scott Dorsey]

Gary Eickmeier > wrote:
>
>It has a little button called the MAX switch that "raises the maximum level
>present at the MAIN inputs/outputs from +12 to +22 dBu." I have no idea what
>that does, nor does the book tell me.

It changes the reference operating level, adding a pad on the front end
and gain on the back end. You really, really need to get the Yamaha Sound
Reinforcement Handbook and read the introduction to operating levels.

In the professional world, analogue operating levels are standardized, it's
not like in the consumer world where outputs are all over the place.
--scott


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

[Sean Conolly]

"Dave Plowman (News)" > wrote in message
...
> In article >,
> Sean Conolly > wrote:
>> If you're running a *very* low signal through the unit I could see where
>> quantization noise could start becoming a factor. I recall your
>> receiver is running at -10 consumer levels? As far as I know the DEQ
>> only runs at +4, and although you can adjust the volume it's a digital
>> control upstream of the D/A converter.
>
> I/O levels are switchable between pro and domestic. Sadly only both
> together. It can be useful if the in and out can be switched
> independently.
>

Hmmm - check through the manual - and there it is! There's a switch next to
the RTA mic input labeled 'MAX', which I always thought was for the mic
input level. The manual says it's for switching the input/output levels as
you indicated.

Thanks for that - I had to use the text search on the manual to find it
though.

Sean

[Gary Eickmeier]

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

Scott, how often do you make digital recordings without the max gain
possible without clipping? Maybe in a 24 bit system you can go without the
top half dozen bits lighting up, but you wouldn't purposely try to record at
the 8 or 10 bit level.

Gary

[Mike Rivers[_2_]]

On 8/10/2014 11:42 AM, Gary Eickmeier wrote:
> It has a little button called the MAX switch that "raises the maximum level
> present at the MAIN inputs/outputs from +12 to +22 dBu." I have no idea what
> that does, nor does the book tell me.

I explained it in my post earlier today.

Non-Google translation of "raises the maximum level
present at the MAIN inputs/outputs from +12 to +22 dBu." is

Changes the input sensitivity so that input clipping occurs at +12 dBu
or +22 dBu

You need to learn the language of manuals. Knowing something about
specs, conventions, and operating levels helps, too.


--
For a good time, visit http://mikeriversaudio.wordpress.com

[Gary Eickmeier]

Mike Rivers wrote:
> On 8/9/2014 8:08 PM, Gary Eickmeier wrote:


> So your system sounds better with the DEQ out of the signal path even
> with all of its settings as bypassed as you can get? That's
> disappointing. I wouldn't be surprised that there was a barely
> noticeable change for no other reason than that there's another analog
> component in the signal path, but you make it sound like a horror show
> with it in line.

Some of it could be psychoacoustic. But I am just asking about the AD/ DA
converter part.
>
>> 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.
>
> Well, we used to, when there were only about 12 bits that we could
> count on being reasonably accurate. With today's 24-bit systems, we
> prefer more headroom so that we aren't using most of the bits most of the
> time. The positive result is that what's converted between analog to
> digital and back is more accurate. The downside is that you end up with an
> analog output that may be rather low and you'll amplify the analog
> noise when bringing it up to your normal listening level. This is why
> we have the concept of "gain structure" - to get things working
> together without, as you say, making the best use of the available bits,
> or
> without clipping.

Yes, fairly obvious, and the basis of my question.
>
>> 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.
>
> Doesn't it have an input level meter? How about a switch for nominal
> operating level (-10/+4)? According to the manual, there's a switch to
> change the maximum input level of the main XLR inputs from +12 to +22
> dBu. Since the meter is showing a low level, put the switch in the +12
> position and you'll get 10 dB more gain. Do you have "tape out/in"
> jacks somewhere in your system? Tape output is nearly always fixed
> level, somewhere near the design center for the unit. Of course it
> will vary with the input source, but as long as you aren't clipping the
> input or
> have to turn the volume knob way up to get a usable output level, the
> tape output should be around the nominal operating level of the unit.
> This, for home stereo gear, is usually in the ballpark of -10 dBV.
>
> Still, the equalizer should be a unity gain device (plus or minus
> whatever you do with the equalizer) so what you put in should be what
> you get out. But there's all kinds of processing in there. Make sure
> none of the dynamics stuff, the compressor, limiter, or dynamic EQ
> processing is switched in. You might also try using the auxiliary
> output jacks and use the I/O menu to feed those outputs from
> somewhere earlier in the chain than what feeds the main outputs.
>
> Now I don't have a DEQ2496, and I've never used one, so don't take
> this as defending the product. But lots of people use them effectively,
> lots of people just hate everything that Behringer makes. It sounds like
> you've at least listened critically to it and can be objective about
> whether the overall effectiveness is to make your system sound better
> or worse. But explore the options and settings fully before you give
> up on it.
>> So would Behringer have a way of working around these problems
>
> Yes. The input level switch, for one. RTFM
>
> > am I correct in being nervous about it?
>
> You shouldn't be nervous, you should look for the problem and fix it
> (which may mean taking the equalizer out of your system) if it
> bothers you.

OK, I will try it again and study those gain settings and the various
meters. I know they have tried to think of everything, and this is a very
special product that may deserve greater attention from the owner....

Gary

[Scott Dorsey]

Gary Eickmeier > wrote:
>Kludge writes:
>> 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, how often do you make digital recordings without the max gain
>possible without clipping? Maybe in a 24 bit system you can go without the
>top half dozen bits lighting up, but you wouldn't purposely try to record at
>the 8 or 10 bit level.

The 8 bit level would be 96dB down from maximum in a 24 bit system. That is
pretty damn far down.

You have available to you an outrageous amount of dynamic range in a modern
digital system. Don't be afraid to use it. A little matter of 12 or 24 dB
is not going to be a worry.

There was a need to keep tight control over levels in digital systems back
in the early 1980s when converters weren't very linear, and back then a lot
of people would give out advice to "use all the bits." Those days are long,
long gone, thank God. Back then if you recorded with peaks at -60dBFS on the
PCM1610, the sound was audibly distorted and buzzy. Today with modern sigma
delta converters, I can record at -60dBFS and you probably won't even notice
the increase in noise floor since the ambient noise of the hall is _still_
above the converter noise floor. It is a different era.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[geoff]

On 10/08/2014 11:57 p.m., 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
>


But Scott, pretty much ALL power amplifiers are indeed running at 'full
tilt' all the time. Level control is an attenuator to the input signal, no ?

;-)

geoff

[Sean Conolly]

"Gary Eickmeier" > wrote in message
...
> Mike Rivers wrote:
>> On 8/9/2014 8:08 PM, Gary Eickmeier wrote:
>
>
>> So your system sounds better with the DEQ out of the signal path even
>> with all of its settings as bypassed as you can get? That's
>> disappointing. I wouldn't be surprised that there was a barely
>> noticeable change for no other reason than that there's another analog
>> component in the signal path, but you make it sound like a horror show
>> with it in line.
>
> Some of it could be psychoacoustic. But I am just asking about the AD/ DA
> converter part.
>>
>>> 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.
>>
>> Well, we used to, when there were only about 12 bits that we could
>> count on being reasonably accurate. With today's 24-bit systems, we
>> prefer more headroom so that we aren't using most of the bits most of the
>> time. The positive result is that what's converted between analog to
>> digital and back is more accurate. The downside is that you end up with
>> an
>> analog output that may be rather low and you'll amplify the analog
>> noise when bringing it up to your normal listening level. This is why
>> we have the concept of "gain structure" - to get things working
>> together without, as you say, making the best use of the available bits,
>> or
>> without clipping.
>
> Yes, fairly obvious, and the basis of my question.
>>
>>> 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.
>>
>> Doesn't it have an input level meter? How about a switch for nominal
>> operating level (-10/+4)? According to the manual, there's a switch to
>> change the maximum input level of the main XLR inputs from +12 to +22
>> dBu. Since the meter is showing a low level, put the switch in the +12
>> position and you'll get 10 dB more gain. Do you have "tape out/in"
>> jacks somewhere in your system? Tape output is nearly always fixed
>> level, somewhere near the design center for the unit. Of course it
>> will vary with the input source, but as long as you aren't clipping the
>> input or
>> have to turn the volume knob way up to get a usable output level, the
>> tape output should be around the nominal operating level of the unit.
>> This, for home stereo gear, is usually in the ballpark of -10 dBV.
>>
>> Still, the equalizer should be a unity gain device (plus or minus
>> whatever you do with the equalizer) so what you put in should be what
>> you get out. But there's all kinds of processing in there. Make sure
>> none of the dynamics stuff, the compressor, limiter, or dynamic EQ
>> processing is switched in. You might also try using the auxiliary
>> output jacks and use the I/O menu to feed those outputs from
>> somewhere earlier in the chain than what feeds the main outputs.
>>
>> Now I don't have a DEQ2496, and I've never used one, so don't take
>> this as defending the product. But lots of people use them effectively,
>> lots of people just hate everything that Behringer makes. It sounds like
>> you've at least listened critically to it and can be objective about
>> whether the overall effectiveness is to make your system sound better
>> or worse. But explore the options and settings fully before you give
>> up on it.
>>> So would Behringer have a way of working around these problems
>>
>> Yes. The input level switch, for one. RTFM
>>
>> > am I correct in being nervous about it?
>>
>> You shouldn't be nervous, you should look for the problem and fix it
>> (which may mean taking the equalizer out of your system) if it
>> bothers you.
>
> OK, I will try it again and study those gain settings and the various
> meters. I know they have tried to think of everything, and this is a very
> special product that may deserve greater attention from the owner....

... but to be fair, I've had one for many years and never knew that the MAX
button wasn't for the RTA mic. It is in the manual, but it ain't hard to
miss either.

Sean

[Sean Conolly]

"Gary Eickmeier" > wrote in message
...
> 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
>
> Scott, how often do you make digital recordings without the max gain
> possible without clipping? Maybe in a 24 bit system you can go without the
> top half dozen bits lighting up, but you wouldn't purposely try to record
> at the 8 or 10 bit level.

Powers of 2... if you only used half of the 96 dB range you're still using
23 bits - think about that.

The analog section won't have the same dynamic range, and that's why you do
want to make sure the signal is up out of the mud.

Sean

[Ron C[_2_]]

On 8/10/2014 5:10 PM, geoff wrote:
> On 10/08/2014 11:57 p.m., 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
>>
>
>
> But Scott, pretty much ALL power amplifiers are indeed running at 'full
> tilt' all the time. Level control is an attenuator to the input signal,
> no ?
>
> ;-)
>
> geoff

IMHO, "capable of" is not the same as "running at."

Just sayin' ...

==
Later...
Ron Capik
--

[Ron C[_2_]]

On 8/10/2014 7:07 PM, Sean Conolly wrote:
> "Gary Eickmeier" > wrote in message
> ...
>> 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
>>
>> Scott, how often do you make digital recordings without the max gain
>> possible without clipping? Maybe in a 24 bit system you can go without the
>> top half dozen bits lighting up, but you wouldn't purposely try to record
>> at the 8 or 10 bit level.
>
> Powers of 2... if you only used half of the 96 dB range you're still using
> 23 bits - think about that.
>
> The analog section won't have the same dynamic range, and that's why you do
> want to make sure the signal is up out of the mud.
>
> Sean
>
>
Seen another way, you've gone from 16+ million possible
levels to 8+ million possible levels - think about that.

==
Later...
Ron Capik
--

[[email protected]]

Its 6 dB per bit

If you are 12 dB below full scale, you have 2 bits in reserve.

Maek

[PStamler]

Gary: Please do a test of the system's gain with and without the DEQ in place, with the switch set to +12, and again with the switch set to +22, and tell us the result. I still think the DEQ qas, in your original experiment, slightly attenuating the signal.

Oh, and you asked Scott how often he makes recordings without trying to use the whole dynamic range. 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.

[Gary Eickmeier]

Mike Rivers wrote:
> On 8/10/2014 11:42 AM, Gary Eickmeier wrote:
>> It has a little button called the MAX switch that "raises the
>> maximum level present at the MAIN inputs/outputs from +12 to +22
>> dBu." I have no idea what that does, nor does the book tell me.
>
> I explained it in my post earlier today.
>
> Non-Google translation of "raises the maximum level
> present at the MAIN inputs/outputs from +12 to +22 dBu." is
>
> Changes the input sensitivity so that input clipping occurs at +12 dBu
> or +22 dBu
>
> You need to learn the language of manuals. Knowing something about
> specs, conventions, and operating levels helps, too.

OK so at the +22 position, I can raise the gain going into it 10 dB more
than at the +12 position - and then there is a compensating 10 dB pad at the
Main Out jacks?

No wait a minute - that didn't make total sense either. Just thinking out
loud here...I want to put more gain into the AD converter. So I just turn up
the gain - but that makes the volume too loud in the room. So they let me
turn it up 10 dB more anyway but take if off the other end. Yah - I guess
that makes sense. So I want to keep it on the +22 position. So why don't
they just make it that way in the first place? Who needs to be limited to
the +12?

Gary

[Gary Eickmeier]

Sean Conolly wrote:
> "Dave Plowman (News)" > wrote in message
> ...
>> In article >,
>> Sean Conolly > wrote:
>>> If you're running a *very* low signal through the unit I could see
>>> where quantization noise could start becoming a factor. I recall
>>> your receiver is running at -10 consumer levels? As far as I know
>>> the DEQ only runs at +4, and although you can adjust the volume
>>> it's a digital control upstream of the D/A converter.
>>
>> I/O levels are switchable between pro and domestic. Sadly only both
>> together. It can be useful if the in and out can be switched
>> independently.
>>
>
> Hmmm - check through the manual - and there it is! There's a switch
> next to the RTA mic input labeled 'MAX', which I always thought was
> for the mic input level. The manual says it's for switching the
> input/output levels as you indicated.
>
> Thanks for that - I had to use the text search on the manual to find
> it though.
>
> Sean

There are a lot of thngs about this manual that are like that - they seem to
be telling us something and then a minute later you are scratching your head
and have to go back in again. By playing with the unit and reading over
again you gradually get most of it. Probably a translation from German.

There is still one part that I don't get - the EQ function lets you draw
whatever curve you think is appropriate for your idea of a room curve. I
tried to memorize that curve, but it just will not work. I have to re-draw
it every single time I want to do another Automatic EQ. Do you know what I
mean Sean?

Gary

[PStamler]

On Sunday, August 10, 2014 7:13:06 PM UTC-6, Gary Eickmeier wrote:

> There is still one part that I don't get - the EQ function lets you draw
> whatever curve you think is appropriate for your idea of a room curve. I
> tried to memorize that curve, but it just will not work. I have to re-draw
> it every single time I want to do another Automatic EQ. Do you know what I

I don't know about the Automatic EQ, but as I recall the DEQ has a Save function, allowing you to save a particular room EQ curve. In fact, my recollection is that you can save several of them.

As far as why they included a function to switch from max=+12 to max=+22, well, back in the 1970s the Teac company was inventing "prosumer" recording equipment and the whole home-studio thing. At the time pro equipment was almost universally built with a nominal operating level of +4dBu; it had varying amounts of headroom, but it would typically clip at somewhere between +18dBu and +24dBu. Well, Teac was using cheap opamps in their new line of bargain-priced multitrack gear, and they couldn't handle professional levels very well, let along professional load impedances (typically 600 ohms).. So Teac, which soon became Tascam, invented and promulgated a new standard, wherein nominal level became -10dBV, which is about -7.8dBu. 20dB of headroom over that yields about +12dBu.

This new "standard" was not recognized by any professional standards committee, but it became the de facto standard for a lot of cheap gear which was really consumer-grade, but was marketed as "prosumer" or "semi-pro". That included a lot of Behringer gear.

So the switch on the DEQ is to make it alternately compatible with prosumer gear (in the +12dBu position) or real professional gear (+22dB position -- by the way, that's a bit puny as a clipping point for pro gear, but I digress). It's really a half-assed idea, because the DEQ will work just fine with prosumer levels, but as you discovered that doesn't make the meters light up, and people get upset over that.

If you're curious about what I mean by "nominal level", you really need to read the Yamaha Sound Reinforcement Handbook. You'll save yourself a lot of heartache and needless worry if you do. A lot.

And you still need to find out if the DEQ, set flat, is introducing a slight attenuation into the playback chain. I'm betting it is, and that's the reason why the sound improved when you removed it from the chain.

Peace,
Paul

[Gary Eickmeier]

PStamler wrote:
> On Sunday, August 10, 2014 7:13:06 PM UTC-6, Gary Eickmeier wrote:
>
>> There is still one part that I don't get - the EQ function lets you
>> draw whatever curve you think is appropriate for your idea of a room
>> curve. I tried to memorize that curve, but it just will not work. I
>> have to re-draw it every single time I want to do another Automatic
>> EQ. Do you know what I
>
> I don't know about the Automatic EQ, but as I recall the DEQ has a
> Save function, allowing you to save a particular room EQ curve. In
> fact, my recollection is that you can save several of them.

Yes, 64 of them. But the one I am talking about apparently can't be saved.
If you put it in one of the memories, and go into the Auto EQ function, you
then cannot retrieve a memorized curve for the target, you have to rebuild
your curve all over again. ****er.
>
> As far as why they included a function to switch from max=+12 to
> max=+22, well, back in the 1970s the Teac company was inventing
> "prosumer" recording equipment and the whole home-studio thing. At
> the time pro equipment was almost universally built with a nominal
> operating level of +4dBu; it had varying amounts of headroom, but it
> would typically clip at somewhere between +18dBu and +24dBu. Well,
> Teac was using cheap opamps in their new line of bargain-priced
> multitrack gear, and they couldn't handle professional levels very
> well, let along professional load impedances (typically 600 ohms). So
> Teac, which soon became Tascam, invented and promulgated a new
> standard, wherein nominal level became -10dBV, which is about
> -7.8dBu. 20dB of headroom over that yields about +12dBu.

Interesting. I had - and still have - the Teac 4 channel Simul Sync reel to
reel recorder for my film work. Yes, film. I shot Super 8 Sound sync sound
in stereo and surround sound and had to do mixdowns from several tracks in
sync with the projector.
>
> This new "standard" was not recognized by any professional standards
> committee, but it became the de facto standard for a lot of cheap
> gear which was really consumer-grade, but was marketed as "prosumer"
> or "semi-pro". That included a lot of Behringer gear.
>
> So the switch on the DEQ is to make it alternately compatible with
> prosumer gear (in the +12dBu position) or real professional gear
> (+22dB position -- by the way, that's a bit puny as a clipping point
> for pro gear, but I digress). It's really a half-assed idea, because
> the DEQ will work just fine with prosumer levels, but as you
> discovered that doesn't make the meters light up, and people get
> upset over that.
>
> If you're curious about what I mean by "nominal level", you really
> need to read the Yamaha Sound Reinforcement Handbook. You'll save
> yourself a lot of heartache and needless worry if you do. A lot.

Getting yet another book on audio won't tell me what the damn Behringer
manual is trying to tell me. Can I google up what you're talking about?
>
> And you still need to find out if the DEQ, set flat, is introducing a
> slight attenuation into the playback chain. I'm betting it is, and
> that's the reason why the sound improved when you removed it from the
> chain.
>
> Peace,
> Paul

Could be, but I am not doing some A/B instant comparison when I switch it in
and out. But I guess I could check it if you think.

Gary

[Sean Conolly]

> wrote in message
...
> Its 6 dB per bit
>
> If you are 12 dB below full scale, you have 2 bits in reserve.

Much better - thank you.

Sean

[geoff]

On 11/08/2014 4:54 p.m., Gary Eickmeier wrote:

>
> Getting yet another book on audio won't tell me what the damn Behringer
> manual is trying to tell me. Can I google up what you're talking about?


Can you ? You'd need to check the manual for Google - but it may not be
100% clear !

;-)

geoff

[Phil W[_3_]]

"Mike Rivers":
> On 8/9/2014 8:08 PM, Gary Eickmeier wrote:
>
>> 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.
>
> Again, RTFM and explore the controls. The Meter menu turns the display
> into three pages worth of metering options, including showing the input
> level.

Well, most people in this group should have learned a while ago, that Mr
Know-it-all-best is not able to RTFM and conclude from what´s written there.
No matter, which subject. This particular moron refuses to go along well
with concepts and devices, which work well for most (if not all) other
users.

[Dave Plowman (News)]

In article >,
PStamler > wrote:
> As far as why they included a function to switch from max=+12 to
> max=+22, well, back in the 1970s the Teac company was inventing
> "prosumer" recording equipment and the whole home-studio thing. At the
> time pro equipment was almost universally built with a nominal operating
> level of +4dBu; it had varying amounts of headroom, but it would
> typically clip at somewhere between +18dBu and +24dBu. Well, Teac was
> using cheap opamps in their new line of bargain-priced multitrack gear,
> and they couldn't handle professional levels very well, let along
> professional load impedances (typically 600 ohms). So Teac, which soon
> became Tascam, invented and promulgated a new standard, wherein nominal
> level became -10dBV, which is about -7.8dBu. 20dB of headroom over that
> yields about +12dBu.

Interesting hypothesis. I'd always assumed that domestic line level was
loosely based around what existed at the output of an AM radio receiver
and the input to its AF stages in the valve days. And it doesn't seem odd
to me that a maker like Teac might adopt this (near enough) too for
something which wasn't specifically aimed at the then pro market.

Pro line levels date back to the early days of telephony, and I'd say it
was first adopted by broadcasters who had to interface with this, rather
than for any intrinsic reason.

--
*Reality is a crutch for people who can't handle drugs.

Dave Plowman London SW
To e-mail, change noise into sound.

[None]

"Gary Eickmeier" > wrote in message
...
> Getting yet another book on audio won't tell me what the damn
Behringer
> manual is trying to tell me.

Woooooooshhhhhh!

[None]

"Phil W" > wrote in message
...
> Well, most people in this group should have learned a while ago,
> that Mr Know-it-all-best is not able to RTFM and conclude from
> what´s written there.
> No matter, which subject. This particular moron refuses to go along
> well with concepts and devices, which work well for most (if not
> all) other users.

He whines that he refuses to read the standard Yamaha text because it
won't explain the Behringer manual to him. Multi-level stupidity.

[Mike Rivers[_2_]]

On 8/10/2014 9:07 PM, Gary Eickmeier wrote:
> ...I want to put more gain into the AD converter. So I just turn up
> the gain - but that makes the volume too loud in the room. So they let me
> turn it up 10 dB more anyway but take if off the other end. Yah - I guess
> that makes sense. So I want to keep it on the +22 position. So why don't
> they just make it that way in the first place? Who needs to be limited to
> the +12?

It's really simple, but you need to understand the difference between
gain and sensitivity.

You have a source with a fixed output level. According to the DEQ's
meter, it's too low for good performance. Look at the MAX switch. If
it's in the +22 position, set it to the +10 position and you'll see a
higher level on the DEQ's meters. If it's already set to +10 then that's
all the level you'll get. Live with it or put another stage of
amplification between your source and the DEQ.

The unit maintains (with everything bypassed) unity gain between input
and output, so the the level coming out will be the same as the level
going in. So whether the switch is set for +10 or +22, you won't get any
loss through the unit, but you won't get any gain, either.

--
For a good time, visit http://mikeriversaudio.wordpress.com

[Mike Rivers[_2_]]

On 8/11/2014 12:54 AM, Gary Eickmeier wrote:
PStamler wrote:
>> I don't know about the Automatic EQ, but as I recall the DEQ has a
>> Save function, allowing you to save a particular room EQ curve. In
>> fact, my recollection is that you can save several of them.

> Yes, 64 of them. But the one I am talking about apparently can't be saved.
> If you put it in one of the memories, and go into the Auto EQ function, you
> then cannot retrieve a memorized curve for the target, you have to rebuild
> your curve all over again. ****er.

That makes sense to me. When you're using Auto EQ, the DEQ is making the
decision as to what the curve should be, at least initially. In that
mode, it doesn't care about a curve that you used at one time.

> Getting yet another book on audio won't tell me what the damn Behringer
> manual is trying to tell me. Can I google up what you're talking about?

Paul's explanation of nominal operating levels is excellent. Study that.
It wouldn't hurt if you were able to make some actual measurements on
your system to see if Paul's suggestion that there's some loss through
the DEQ and the lower listening level is throwing you off. In lieu of
buying test equipment, try turning the volume up just a tad when you
have the DEQ in line and see if that fixes all of your problems in one
simple step.

I'm not going to read the manual to you, but look closely to see if
there's a control for output level in addition to the MAX switch. It
would be functionally similar to the make-up gain control that's found
on most compressors that allows you to bring the output level back to
nominal after the compressor reduces the signal level as it does its job.

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



--
For a good time, visit http://mikeriversaudio.wordpress.com

[Mike Rivers[_2_]]

On 8/11/2014 6:09 AM, Dave Plowman (News) wrote:

> I'd always assumed that domestic line level was
> loosely based around what existed at the output of an AM radio receiver
> and the input to its AF stages in the valve days. And it doesn't seem odd
> to me that a maker like Teac might adopt this (near enough) too for
> something which wasn't specifically aimed at the then pro market.

Actually, most consumer audio gear from before the solid state era
worked at a level somewhere between -20 and -15 dBu. TASCAM liked -10
dBV so users wouldn't be confuse it with dBm (power). They also promoted
the "voltage matched" interfacing concept of having a relatively low
output impedance feeding a relatively high input impedance. With this
setup, very little voltage is dropped across the output impedance and
nearly the full (open circuit) output voltage is what the input sees.

In a "power matched" connection with a 600 ohm source feeding a 600 ohm
input, while that provided the maximum power transfer, half the open
circuit voltage was dropped across the output when it was connected to a
load of equal impedance.


--
For a good time, visit http://mikeriversaudio.wordpress.com

[Scott Dorsey]

Gary Eickmeier > wrote:
>
>No wait a minute - that didn't make total sense either. Just thinking out
>loud here...I want to put more gain into the AD converter. So I just turn up
>the gain - but that makes the volume too loud in the room. So they let me
>turn it up 10 dB more anyway but take if off the other end. Yah - I guess
>that makes sense. So I want to keep it on the +22 position. So why don't
>they just make it that way in the first place? Who needs to be limited to
>the +12?

If you never use the digital inputs and outputs, there is no reason to ever
use the +12 position. However, if you use the digital inputs and outputs,
you may want your digital levels to correspond with some known analogue level
and that level may vary from facility to facility since there are two common
standards.
--scott

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

[Scott Dorsey]

Dave Plowman (News) > wrote:
>In article >,
> PStamler > wrote:
>> As far as why they included a function to switch from max=+12 to
>> max=+22, well, back in the 1970s the Teac company was inventing
>> "prosumer" recording equipment and the whole home-studio thing. At the
>> time pro equipment was almost universally built with a nominal operating
>> level of +4dBu; it had varying amounts of headroom, but it would
>> typically clip at somewhere between +18dBu and +24dBu. Well, Teac was
>> using cheap opamps in their new line of bargain-priced multitrack gear,
>> and they couldn't handle professional levels very well, let along
>> professional load impedances (typically 600 ohms). So Teac, which soon
>> became Tascam, invented and promulgated a new standard, wherein nominal
>> level became -10dBV, which is about -7.8dBu. 20dB of headroom over that
>> yields about +12dBu.
>
>Interesting hypothesis. I'd always assumed that domestic line level was
>loosely based around what existed at the output of an AM radio receiver
>and the input to its AF stages in the valve days. And it doesn't seem odd
>to me that a maker like Teac might adopt this (near enough) too for
>something which wasn't specifically aimed at the then pro market.

"Domestic line level" meaning what comes out of and goes into a typical
home stereo receiver is in fact that way, and it bears no real connection
to either the -10dBV or the +4dBm standards. Also of course those systems
were not constant-Z.

>Pro line levels date back to the early days of telephony, and I'd say it
>was first adopted by broadcasters who had to interface with this, rather
>than for any intrinsic reason.

Yes, but THOSE guys use +8dBm, and some of them used 150 ohm systems instead
of 600 ohm. And back then everything was constant-Z.
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[PStamler]

On Sunday, August 10, 2014 10:54:25 PM UTC-6, Gary Eickmeier wrote:

> > I don't know about the Automatic EQ, but as I recall the DEQ has a
> > Save function, allowing you to save a particular room EQ curve. In
> > fact, my recollection is that you can save several of them.
>
> Yes, 64 of them. But the one I am talking about apparently can't be saved..
>
> If you put it in one of the memories, and go into the Auto EQ function, you
> then cannot retrieve a memorized curve for the target, you have to rebuild
> your curve all over again. ****er.

In that case, an alternative way to save your settings is an old-fashioned one: write them down. Yes, it's slow, but it works.

> > If you're curious about what I mean by "nominal level", you really
> > need to read the Yamaha Sound Reinforcement Handbook. You'll save
> > yourself a lot of heartache and needless worry if you do. A lot.
>
> Getting yet another book on audio won't tell me what the damn Behringer
> manual is trying to tell me. Can I google up what you're talking about?

Not really. The reason I, and others, have suggested that you buy and read the Yamaha book is that you know a certain amount about various audio topics -- just enough to get confused by situations like this one. The Yamaha book, if you read it carefully several times (keep it in the bathroom), will put a furm foundation beneath your understanding. You know some stuff, but you need better understanding of nominal levels, headroom, etc., and the Yamaha book will give you that, and a lot more. With that knowledge, bad manuals like the Behringer one (and I've looked at it, and it is bad) won't leave you flummoxed.

> > And you still need to find out if the DEQ, set flat, is introducing a
> > slight attenuation into the playback chain. I'm betting it is, and
> > that's the reason why the sound improved when you removed it from the
> > chain.
>
> Could be, but I am not doing some A/B instant comparison when I switch it in
> and out. But I guess I could check it if you think.

I think.

Peace,
Paul

[Ron C[_2_]]

On 8/11/2014 12:54 AM, Gary Eickmeier wrote:
> PStamler wrote:
>> On Sunday, August 10, 2014 7:13:06 PM UTC-6, Gary Eickmeier wrote:
>>
>>> There is still one part that I don't get - the EQ function lets you
>>> draw whatever curve you think is appropriate for your idea of a room
>>> curve. I tried to memorize that curve, but it just will not work. I
>>> have to re-draw it every single time I want to do another Automatic
>>> EQ. Do you know what I
>>
>> I don't know about the Automatic EQ, but as I recall the DEQ has a
>> Save function, allowing you to save a particular room EQ curve. In
>> fact, my recollection is that you can save several of them.
>
> Yes, 64 of them. But the one I am talking about apparently can't be saved.
> If you put it in one of the memories, and go into the Auto EQ function, you
> then cannot retrieve a memorized curve for the target, you have to rebuild
> your curve all over again. ****er.
> < ....snip... >
>>
>
> Could be, but I am not doing some A/B instant comparison when I switch it in
> and out. But I guess I could check it if you think.
>
> Gary
>
>
>
I'm not sure I understand your problem.

When I got my DEQ2496 I wanted to transfer the settings from
my old graphic EQs to the DEQ2496. I used the auto EQ function
to clone the response curves. I had no problem saving them.
I did it ~6 years ago, so don't remember the details.

==
Later...
Ron Capik
--

[Gary Eickmeier]

Mike Rivers wrote:
> On 8/11/2014 12:54 AM, Gary Eickmeier wrote:

> That makes sense to me. When you're using Auto EQ, the DEQ is making
> the decision as to what the curve should be, at least initially. In
> that mode, it doesn't care about a curve that you used at one time.

Nope - see, that's what is so weird about this piece - you CAN do just what
I am telling you. In the Auto EQ mode you can

1. Exclude certain frequency ranges that would be impossible for it to cope
with, such as the extreme bass or treble, due to not wanting to stress out
your speakers in those ranges if they can't easily achieve your curve.

2. You can set up a curve that you want it ti EQ to, such as the room curve
that many have said is more desireable than flat. But you can't put in a
memorized one, becaue to do that you would have to get out of the RTA mode
and go into Memory. It's just designed that way. You set your curve, then
put it into Auto EQ, and it measures the ambient room noise, such as air
conditioners and power lawnmowers, and tells you if it is too noisy to work
with, and you can raise the gain of trhe pink noise signal. Then when
satisfied that it can do the job, it goes into the EQ routine. You can set
it for Fast, Mid, or Slow and just watch the little electronic "sliders"
tighten up on your curve. After a minute or two there is no further progress
and you can stop it and take a look at the sliders by themselves and adjust
even more if you want. So you go into Memorize mode and do that curve and
give it a name. Then what I do is put on a CD of my own favorite Pink Noise
track and do an RTA on that with the calibrated mike and touch up the
sliders a little more if I want.
>
>> Getting yet another book on audio won't tell me what the damn
>> Behringer manual is trying to tell me. Can I google up what you're
>> talking about?
>
> Paul's explanation of nominal operating levels is excellent. Study
> that. It wouldn't hurt if you were able to make some actual
> measurements on your system to see if Paul's suggestion that there's
> some loss through the DEQ and the lower listening level is throwing
> you off. In lieu of buying test equipment, try turning the volume up
> just a tad when you have the DEQ in line and see if that fixes all of
> your problems in one simple step.

No, there is no lower listening level, the level is set by me using the
volume control. He may be thinking of the time honored principle of the
sound with the higher gain sounding better to the average ear . This is not
like that.
>
> I'm not going to read the manual to you, but look closely to see if
> there's a control for output level in addition to the MAX switch. It
> would be functionally similar to the make-up gain control that's found
> on most compressors that allows you to bring the output level back to
> nominal after the compressor reduces the signal level as it does its
> job.

Yes, there is a Level Offset in the Utility menu mainly for the purpose of
getting it back to a unity gain process after an EQ session.

>
> 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.

Nice article in Recording about whether to leave a system on or off when not
using it. I caught your picture in the Lies Damn Lies and Audio Specs panel
video. But the video is all Ethan. I wished I could see all of you,
especially Dave Moran.

Gary

[Gary Eickmeier]

Ron C wrote:

> I'm not sure I understand your problem.
>
> When I got my DEQ2496 I wanted to transfer the settings from
> my old graphic EQs to the DEQ2496. I used the auto EQ function
> to clone the response curves. I had no problem saving them.
> I did it ~6 years ago, so don't remember the details.
>
> ==
> Later...
> Ron Capik

I would guess that what you did was set the sliders to your older unit's
settings and then let the Auto EQ conform to that, and then you went into
Memory mode and grabbed that as your favorite EQ. IF that is the case, I
would seriously question that procedure. In the first place, you might as
well just set your old slider positions into the unit and memorize them and
there is no need to do an Auto EQ. In the second place, I would think you
would want a second opinion about what EQ you want, based on your calibrated
microphone and the Behringer's capabilities. How did the RTA come out after
you were done? Same as with your old unit?

Interesting to ponder. Anyway, thanks.

Gary

[PStamler]

On Monday, August 11, 2014 8:46:20 PM UTC-6, Gary Eickmeier wrote:

> No, there is no lower listening level, the level is set by me using the
> volume control. He may be thinking of the time honored principle of the
> sound with the higher gain sounding better to the average ear . This is not
> like that.

Have you done a measurement that shows the same system gain with the DEQ in circuit and not? If you haven't, then you really don't know what's going on. Do it, for God's sake. It'll take ten minutes, and then you can rule out that possibility.

Run tone, at 1kHz, into the system with the DEQ present, set the volume control on your receiver so that the level from the speaker is reasonable but not too annoying, and measure the voltage across the speaker terminals. Write that number down. Then take the DEQ out of the system and hook things up directly, without it. Don't touch anything on the receiver's control panel.. Run the same tone into the system, and measure the voltage across the speaker terminals. Write that down, and compare the two readings. What do you get?

Until you've done that test, you don't have any idea what the system is doing.

Peace,
Paul

[PStamler]

On Monday, August 11, 2014 9:04:59 PM UTC-6, PStamler wrote:

> Until you've done that test, you don't have any idea what the system is doing.

Oh, and of course do that test with the DEQ set flat.

Peace,
Paul

[Gary Eickmeier]

PStamler wrote:
> Gary: Please do a test of the system's gain with and without the DEQ
> in place, with the switch set to +12, and again with the switch set
> to +22, and tell us the result. I still think the DEQ qas, in your
> original experiment, slightly attenuating the signal.

I could just listen while pressing the button between +12 and +22. But you
are saying to meaure the level with it in and each setting and then with it
out? I'm not sure what it would matter even if tthere were a difference - I
would just turn the volume control to compensate for it. And besides all
that, as I say, there is a Gain Offset control that can turn the unit higher
or lower to compensate for any of that.
>
> Oh, and you asked Scott how often he makes recordings without trying
> to use the whole dynamic range. 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.

Yes, I have made a few of those, for one reason or another, and after some
judicious boost in mastering they seem to come out just fine. But still, if
you have some control over the situation, you should record a good strong
signal

Gary

[PStamler]

On Monday, August 11, 2014 10:50:03 PM UTC-6, Gary Eickmeier wrote:
> PStamler wrote:
>
> > Gary: Please do a test of the system's gain with and without the DEQ
>
> > in place, with the switch set to +12, and again with the switch set
>
> > to +22, and tell us the result. I still think the DEQ qas, in your
>
> > original experiment, slightly attenuating the signal.
>
>
>
> I could just listen while pressing the button between +12 and +22. But you
>
> are saying to measure the level with it in and each setting and then with it
>> out? I'm not sure what it would matter even if there were a difference - I
> would just turn the volume control to compensate for it. And besides all
> that, as I say, there is a Gain Offset control that can turn the unit higher
> or lower to compensate for any of that.

Yes, of course. But I'm trying to sleuth out the answer to your original question. You reported that taking the DEQ out of circuit made the sound clean up in several ways, and what I'm asking you to do is to verify that the DEQ isn't inducing a level change in the chain. A level change -- even a small one -- could produce all the sonic alterations you heard when you took the DEQ out of the system, and what I'm trying to do is to eliminate the possibility that this was due to a level change. Once that possibility is eliminated one can go on to try and figure out what in the DEQ might be degrading the sound -- but first you have to verify that it's not a level issue.

By the way, I said that the level change can cause sonic alterations even if it's very small. I've done experiments in class where I played two audio files alternately and asked students to answer questions like "Which has more bass?" "Which has deeper bass?" "Which has a more three-dimensional soundstage" and the like. Some of the results were no better than chance, but some seemed to indicate that the students were hearing something real. The class was too small to generate truly valid statistics, but they seemed able to hear differences in files that differed in level by as little as 0.3dB (but were otherwise identical). Note that they *didn't* hear one file as being louder than the other; instead they heard differences in tonal quality, soundstage, etc., suggesting to me that very small level changes can produce noticeable changes in sound. That's why I'm trying to eliminate that possibility before proceeding further with your DEQ problem.

Please go do the level check.

Peace,
Paul

> >
>
> > Oh, and you asked Scott how often he makes recordings without trying
>
> > to use the whole dynamic range. 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.
>
>
>
> Yes, I have made a few of those, for one reason or another, and after some
>
> judicious boost in mastering they seem to come out just fine. But still, if
>
> you have some control over the situation, you should record a good strong
>
> signal
>
>
>
> Gary

[PStamler]

On Monday, August 11, 2014 10:50:03 PM UTC-6, Gary Eickmeier wrote, speaking of recordings made with scanty peak levels:

> Yes, I have made a few of those, for one reason or another, and after some
> judicious boost in mastering they seem to come out just fine. But still, if
> you have some control over the situation, you should record a good strong
> signal

Sure, in theory, but sometimes things just don't work out that way, and on those days I'm grateful that 24-bit recording lets me get away with recording at low levels when I think I'll really need the headroom to handle music with wide dynamic range, or sometimes when I'm remastering a disc cut at a low level, and I need to allow for big scratches.

Peace,
Paul

[Trevor]

"Gary Eickmeier" > wrote in message
...
> Mike Rivers wrote:
>> On 8/10/2014 11:42 AM, Gary Eickmeier wrote:
>>> It has a little button called the MAX switch that "raises the
>>> maximum level present at the MAIN inputs/outputs from +12 to +22
>>> dBu." I have no idea what that does, nor does the book tell me.
>>
>> I explained it in my post earlier today.
>>
>> Non-Google translation of "raises the maximum level
>> present at the MAIN inputs/outputs from +12 to +22 dBu." is
>>
>> Changes the input sensitivity so that input clipping occurs at +12 dBu
>> or +22 dBu
>>
>> You need to learn the language of manuals. Knowing something about
>> specs, conventions, and operating levels helps, too.
>
> OK so at the +22 position, I can raise the gain going into it 10 dB more
> than at the +12 position - and then there is a compensating 10 dB pad at
> the Main Out jacks?
>
> No wait a minute - that didn't make total sense either. Just thinking out
> loud here...I want to put more gain into the AD converter. So I just turn
> up the gain - but that makes the volume too loud in the room. So they let
> me turn it up 10 dB more anyway but take if off the other end. Yah - I
> guess that makes sense. So I want to keep it on the +22 position. So why
> don't they just make it that way in the first place? Who needs to be
> limited to the +12?

Clearly so it can be used with pro or consumer equipment, by those who
understand what voltage levels/sensitivity/gain staging actually means, and
it's effect on noise Vs overload in each device.

Trevor.

[Trevor]

"Dave Plowman (News)" > wrote in message
...
> In article >,
> Sean Conolly > wrote:
>> If you're running a *very* low signal through the unit I could see where
>> quantization noise could start becoming a factor. I recall your
>> receiver is running at -10 consumer levels? As far as I know the DEQ
>> only runs at +4, and although you can adjust the volume it's a digital
>> control upstream of the D/A converter.
>
> I/O levels are switchable between pro and domestic. Sadly only both
> together. It can be useful if the in and out can be switched
> independently.

If you are using consumer devices as both input and output, or pro devices
as input and output, the switching is fine as is. If you are using a motley
collection of both consumer and pro equipment, for which Behringer can
hardly be blamed, a set of pads will fix the problem.

Trevor

[geoff]

On 12/08/2014 3:52 p.m., PStamler wrote:
> On Monday, August 11, 2014 9:04:59 PM UTC-6, PStamler wrote:
>
>> Until you've done that test, you don't have any idea what the system is doing.
>
> Oh, and of course do that test with the DEQ set flat.


.... and all the other functions bypassed !

geoff

[Peter Larsen[_3_]]

On 10-08-2014 01:08, 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

Optimally I think you should feed it digital. Your general comments
remind me of the gain in audio quality I got when I put minimum 15 dB
attenuation (loudspeaker unit sensitivity dependendt) in series with the
outputs from my - back then - LF356 containing x-over. I got the
impression that some opamps plain sound better when running a hotter signal.

> so I yanked
> it out of there and the sound seems a lot better now, tighter, more
> together, larger soundstage, etc.

I don't want to rush into subcribing to the alledged magnitude of the
effect, but there is no such thing as inaudible dsp-processing. Less is
more.

> 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.

Way to go is to put a 15 dB attenuator on its output if you use it with
household level output voltages.

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

There are some guys in France who have made a replacement analog board
for the DCX. Other than that the general comment on Behringers 1 unit
boxes is that that the powersupply arrangement is designed to fit a
budget and that looking into noise on the powersupply possibly is a
worthwhile venture.

> Gary Eickmeier

Oh, there are comments in the thread to the effect that operating level
for it is adjustable. An interesting passtime could be to abx whether
there is an audible difference between the operating levels with the
gain otherwise compensated.

Kind regards

Peter Larsen

[None]

"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!

[None]

"Gary Eickmeier" > wrote in message
...
> I think after doing this stuff for some 60 years I have enough
> understanding of levels to get along.

You seem to spend a lot of your time in the ng proving otherwise .

[Mike Rivers[_2_]]

On 8/11/2014 10:46 PM, Gary Eickmeier wrote:
>> 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.

Apparently you don't. I give up. If you'd like me to help you figure out
how to best use it, I'm available at $50/hour plus travel cost. You need
some hands-on help. Or maybe you should just accept that it was designed
that way because it was designed for people with less experience and
less curiosity than you. You can have a look at the Sabine Graphi-Q2
which does many of the same things but is designed for professionals
like you. It costs about $1200.

> I caught your picture in the Lies Damn Lies and Audio Specs panel
> video. But the video is all Ethan. I wished I could see all of you,
> especially Dave Moran.

So do I, but it was Ethan's video and he gets to say what's in it. The
rest of us on the panel (including Scott) had a lot more to say than
what he included in what he posted on his web site.


--
For a good time, visit http://mikeriversaudio.wordpress.com

[Mike Rivers[_2_]]

On 8/12/2014 12:50 AM, Gary Eickmeier wrote:
> I could just listen while pressing the button between +12 and +22. But you
> are saying to meaure the level with it in and each setting and then with it
> out?

You still don't understand the function of that switch. It won't change
the output level other than perhaps a small amount due to being
imperfect and not having exactly unity gain between input and output
when none of its functions are doing anything (have you checked to see
that they're all really bypassed or inactive?).

By measuring the output level and the input level you'll know whether it
has gain, loss, or behaves like a straight wire.

> I'm not sure what it would matter even if tthere were a difference - I
> would just turn the volume control to compensate for it.

Well, then why not just turn the volume up just a little bit and see if
then it sounds as good in as out? As Paul suggested, the deficiencies
that you say you're hearing may be simply a result of a small volume
change that you might not notice if you can't A/B with a true hardware
bypass.


--
For a good time, visit http://mikeriversaudio.wordpress.com

[John Williamson]

On 12/08/2014 12:14, Mike Rivers wrote:
> On 8/12/2014 12:50 AM, Gary Eickmeier wrote:
>> I could just listen while pressing the button between +12 and +22. But
>> you
>> are saying to meaure the level with it in and each setting and then
>> with it
>> out?
>
> You still don't understand the function of that switch. It won't change
> the output level other than perhaps a small amount due to being
> imperfect and not having exactly unity gain between input and output
> when none of its functions are doing anything (have you checked to see
> that they're all really bypassed or inactive?).
>
> By measuring the output level and the input level you'll know whether it
> has gain, loss, or behaves like a straight wire.
>
All good stuff, but from his past postings here, Mr. Eickmeier seems to
be extremely averse to using any form of measuring equipment for some
reason.


--
Tciao for Now!

John.

[Dave Plowman (News)]

In article >,
Trevor > wrote:
> > I/O levels are switchable between pro and domestic. Sadly only both
> > together. It can be useful if the in and out can be switched
> > independently.

> If you are using consumer devices as both input and output, or pro
> devices as input and output, the switching is fine as is. If you are
> using a motley collection of both consumer and pro equipment, for which
> Behringer can hardly be blamed, a set of pads will fix the problem.

Other low cost makers provide separately switched in and outs. Phonic, for
one.

--
*I'm not being rude. You're just insignificant

Dave Plowman London SW
To e-mail, change noise into sound.

[Mike Rivers[_2_]]

On 8/12/2014 7:23 AM, John Williamson wrote:
> All good stuff, but from his past postings here, Mr. Eickmeier seems to
> be extremely averse to using any form of measuring equipment for some
> reason.

Probably he doesn't have an appropriate meter. There are dB meters that
you can download and, while there isn't a good way to calibrate them to
actual analog voltage without access to a calibrated meter, they can
certainly be used for making relative measurements.

--
For a good time, visit http://mikeriversaudio.wordpress.com

[Scott Dorsey]

"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.

Then why do you keep asking kindergarten-level questions about levels?

You have spent 60 years in the consumer audio world where levels are all
over the place, where there are no standards and no references and everything
runs low-Z outputs into high-Z inputs. In the professional world, the
interfaces are different.

Go out and get a copy of the Yamaha book, and you will be shocked to find
how little you really know about levels.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[John Williamson]

On 12/08/2014 14:12, Mike Rivers wrote:
> On 8/12/2014 7:23 AM, John Williamson wrote:
>> All good stuff, but from his past postings here, Mr. Eickmeier seems to
>> be extremely averse to using any form of measuring equipment for some
>> reason.
>
> Probably he doesn't have an appropriate meter. There are dB meters that
> you can download and, while there isn't a good way to calibrate them to
> actual analog voltage without access to a calibrated meter, they can
> certainly be used for making relative measurements.
>
For voltage comparisons at a fixed frequency, even a cheap DVM from
Radioshack is useful, if not entirely accurate. All that's needed is the
will to use it and a bit if education as to how.

--
Tciao for Now!

John.

[Mike Rivers[_2_]]

On 8/12/2014 9:38 AM, John Williamson wrote:
> For voltage comparisons at a fixed frequency, even a cheap DVM from
> Radioshack is useful, if not entirely accurate. All that's needed is the
> will to use it and a bit if education as to how.

It depends on how expensive your cheap Radio Shack meter is. My cheap
Craftsman meter has only two AC scales, 600v and 200v. On the 200v
scale, mine reads -10 dBV as 00.0v. -9.3 dBV reads 00.1v, and it takes
-6.5 dBv to get it up to 00.2v. On a good day with the sun shining, +4
dBu reads 00.8v, +10 dBu reads 01.9v, +12 dBu reads 02.5v above which it
starts to get reasonably linear. With only 3 digits with the least
significant digit being 0.1v, however, it can only resolve changes of
around 0.5 dB when working around 0 dBu. The good news is that when
there's enough level to get a meaningful reading (about +18 dBu) it's
pretty flat up to 500 Hz.

If his best signal source is a CD player, he'll probably be making
measurements well below 1v and I wouldn't trust the linearity or the
resolution of my cheap meter down in that range, given that what he
needs to do is compare two levels that aren't likely to be different by
more than
1 dB. My Fluke 77 works fine

Maybe the auto-ranging meter that Nick reported as being blown out by
Radio Shack for $15 will do better.

--
For a good time, visit http://mikeriversaudio.wordpress.com

[hank alrich]

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.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

Mike Rivers > wrote:

> You need to learn the language of manuals. Knowing something about
> specs, conventions, and operating levels helps, too.

Yes, _this_, well before re-instructing the world about speakers.

FUNdamentals.

You got the FUN, and you got the mental.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

Gary Eickmeier > wrote:

> No wait a minute - that didn't make total sense either. Just thinking out
> loud here...I

The unit offers full control of input and output levels. Some settings
are akin to turning regular old fashioned knobs, and some are akin to
setting software preferences.

Looking at printed or pixeled letters is not the same as Reading The
Furnished Materials in a manner that leads to comprehension of the basic
functions of the various controls as they apply to different sections of
the device.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

Trevor > wrote:

> "Gary Eickmeier" > wrote in message
> ...
> > Mike Rivers wrote:
> >> On 8/10/2014 11:42 AM, Gary Eickmeier wrote:
> >>> It has a little button called the MAX switch that "raises the
> >>> maximum level present at the MAIN inputs/outputs from +12 to +22
> >>> dBu." I have no idea what that does, nor does the book tell me.
> >>
> >> I explained it in my post earlier today.
> >>
> >> Non-Google translation of "raises the maximum level
> >> present at the MAIN inputs/outputs from +12 to +22 dBu." is
> >>
> >> Changes the input sensitivity so that input clipping occurs at +12 dBu
> >> or +22 dBu
> >>
> >> You need to learn the language of manuals. Knowing something about
> >> specs, conventions, and operating levels helps, too.
> >
> > OK so at the +22 position, I can raise the gain going into it 10 dB more
> > than at the +12 position - and then there is a compensating 10 dB pad at
> > the Main Out jacks?
> >
> > No wait a minute - that didn't make total sense either. Just thinking out
> > loud here...I want to put more gain into the AD converter. So I just turn
> > up the gain - but that makes the volume too loud in the room. So they let
> > me turn it up 10 dB more anyway but take if off the other end. Yah - I
> > guess that makes sense. So I want to keep it on the +22 position. So why
> > don't they just make it that way in the first place? Who needs to be
> > limited to the +12?
>
> Clearly so it can be used with pro or consumer equipment, by those who
> understand what voltage levels/sensitivity/gain staging actually means, and
> it's effect on noise Vs overload in each device.

Yep. Gary hasn't actually gotten into the software heart of the box,
where preference/reference settings can be manipulated well beyond the
range of that input level switch.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

Gary Eickmeier > wrote:

> There is still one part that I don't get - the EQ function lets you draw
> whatever curve you think is appropriate for your idea of a room curve. I
> tried to memorize that curve, but it just will not work.

The DEQ2496's I have installed (in churches and dance studios) hold the
specific settings I loaded into their memories, and then locked in place
to exclude the casual intruder.

It will work fine once you figure out what you're supposed to do to
store the setting.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

None > wrote:

> "Gary Eickmeier" > wrote in message
> ...
> > Getting yet another book on audio won't tell me what the damn
> Behringer
> > manual is trying to tell me.
>
> Woooooooshhhhhh!
>

"Frisbies in the dark"

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

Gary Eickmeier > wrote:

> 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.

This is where you drive off the road and run into the ****ing trees over
and over and over.

If your statement was anywhere near credible you would not have
initiated this thread.

You are not "getting along" with your present "understanding of levels".
In fact, you apparently cannot even insert an EQ in your suginal chain
without several setbacks right around the basic stuff, like levels,
unbalanced/balanced I/O's, etc.

It's damned difficult to help you, maybe impossible.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[Sean Conolly]

"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

[Gary Eickmeier]

"hank alrich" > wrote in message
...


> Yep. Gary hasn't actually gotten into the software heart of the box,
> where preference/reference settings can be manipulated well beyond the
> range of that input level switch.

Hank have you got one of these? What do you mean by "the software heart of
the box? Which screen?

Gary

[Gary Eickmeier]

"hank alrich" > wrote in message
...
> Gary Eickmeier > wrote:
>
>> No wait a minute - that didn't make total sense either. Just thinking out
>> loud here...I
>
> The unit offers full control of input and output levels. Some settings
> are akin to turning regular old fashioned knobs, and some are akin to
> setting software preferences.
>
> Looking at printed or pixeled letters is not the same as Reading The
> Furnished Materials in a manner that leads to comprehension of the basic
> functions of the various controls as they apply to different sections of
> the device.

What are you talking about? Do you have some manual that I don't? I searched
on the internet for something more than comes with it. I found a very
slightly expanded version, but still the same authors and not expanded
enough. Like the MAX button explanation "it switches the max gain from +12
dBu to +22 dBu."

Might be able to get back into it this weekend.

Gary

[Gary Eickmeier]

"hank alrich" > wrote in message
...
> Gary Eickmeier > wrote:
>
>> There is still one part that I don't get - the EQ function lets you draw
>> whatever curve you think is appropriate for your idea of a room curve. I
>> tried to memorize that curve, but it just will not work.
>
> The DEQ2496's I have installed (in churches and dance studios) hold the
> specific settings I loaded into their memories, and then locked in place
> to exclude the casual intruder.
>
> It will work fine once you figure out what you're supposed to do to
> store the setting.

I have figured out what I am supposed to do to store the setting. That is
not pertinent to the problem. Have you done the Auto EQ process?

Gary

[Gary Eickmeier]

"Mike Rivers" > wrote in message
...
> On 8/12/2014 9:38 AM, John Williamson wrote:
>> For voltage comparisons at a fixed frequency, even a cheap DVM from
>> Radioshack is useful, if not entirely accurate. All that's needed is the
>> will to use it and a bit if education as to how.
>
> It depends on how expensive your cheap Radio Shack meter is. My cheap
> Craftsman meter has only two AC scales, 600v and 200v. On the 200v scale,
> mine reads -10 dBV as 00.0v. -9.3 dBV reads 00.1v, and it takes -6.5 dBv
> to get it up to 00.2v. On a good day with the sun shining, +4 dBu reads
> 00.8v, +10 dBu reads 01.9v, +12 dBu reads 02.5v above which it starts to
> get reasonably linear. With only 3 digits with the least significant digit
> being 0.1v, however, it can only resolve changes of around 0.5 dB when
> working around 0 dBu. The good news is that when there's enough level to
> get a meaningful reading (about +18 dBu) it's pretty flat up to 500 Hz.
>
> If his best signal source is a CD player, he'll probably be making
> measurements well below 1v and I wouldn't trust the linearity or the
> resolution of my cheap meter down in that range, given that what he needs
> to do is compare two levels that aren't likely to be different by more
> than
> 1 dB. My Fluke 77 works fine
>
> Maybe the auto-ranging meter that Nick reported as being blown out by
> Radio Shack for $15 will do better.

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 I wanted to know was what
the translated German manual was trying to tell us about that little switch.
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. Also that there are more meters
available on the unit that I should investigate. I will read up on those and
thanks also for that.

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. It
takes a few minutes to make the switch of the unit into or out of the
system, during which I turn the system off. 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.

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. I am comparing the spatial qualities -
imaging - of the new ones to standard practice and you can't do valid
comparisons if you change more than one variable at a time. That is how we
did the Linkwitz challenge comparisons that my prototype speakers won in the
first round. The new ones are the final design with good components and
enclosure design. They feature a new radiation pattern that is doing my
image model theory of how sound should be put into a room very successfully.

I first got the Behringer FBQ 6200, which is a 31 band graphic. I would go
into the music room and run a 31 band pink noise CD and write down the level
of each band from my calibrated microphone and older Audioquest RTA unit,
then go adjust the equalizer and try it again. Then the group advised me on
the 2496 and I began studying what it could do and traded for it. This is
one amazing little piece and I have enjoyed playing with it but apparently
haven't mastered it yet! Going in again this weekend with all of the
suggestions I have got from the group.

Gary

[Gary Eickmeier]

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

>> 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.

Yah, I sure wish I had a digital receiver - if they make such a thing. But
all I have is an older analog Pioneer. I could buy a "high end" AD, but I
don't think there would be that much superiority of it over the Behringer if
they are both modern units.

>
> 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.

Interesting. I have no experience with setting up live sound except to
pester the board guys for a feed for my extra recorder during video taping
or audio recording to get the singer better. So you can isolate feedback to
certain frequencies and kill it that way? Yes, very cool.

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

Gary

[Gary Eickmeier]

"Sean Conolly" > wrote in message
...

> 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

Hey I am never afraid to ask questions, even in this group, so allow me to
be your huckleberry.

I am just glad you are in this discussion - they were starting to beat up on
me again. This 2496 is a real study in features and design. Seems like they
have thought of just about everything, which is why I asked about the gains
business. I have no interest in the dynamic EQ or the stereo width control -
I thought I would love the parametric but so far I haven't had a great need
for it.

Like you I found that I didn't need an EQ so much at home because my
speakers are voiced so well there is nothing needed. I do the subwoofer by
ear because that isn't included in the signal sent to the Behringer.

Gary

[PStamler]

On Tuesday, August 12, 2014 10:14:28 PM UTC-6, Gary Eickmeier wrote:

> 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.

Yes, it would, if there's a level difference. Level differences affect longer-term listening tests as much as they do short-term A/B tests.

You report that the system sounds different with the DEQ in or out, and were wondering why that might be. A small level difference would explain it easily, and before speculating on more complex possible causes you need to eliminate the simple one. The DVM Radio Shack is selling off for $15 would tell you in a couple of minutes.

It
> takes a few minutes to make the switch of the unit into or out of the
> system, during which I turn the system off. 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.

In modern digital equipment, that's fairly unlikely. If the signal level's too small, what you typically get is a crappy signal-to-noise ratio, not changes in stereo imaging.

Go check the (*&%^*& system gain. Without first eliminating that very simple possible cause, you're simply fumbling in the dark.

Peace,
Paul

[Dave Plowman (News)]

In article >,
Gary Eickmeier > wrote:
> All I wanted to know was what the translated German manual was trying
> to tell us about that little switch. 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. Also that there are more meters available on the unit
> that I should investigate. I will read up on those and thanks also for
> that.

The switch alters the gain of the input and the output level at the same
time. Keeping the overall gain the same at nominally unity. Reason being
you'll get slightly improved noise figures with the correct setting for
your application. And make clipping less likely.

There are three meters available by toggling the meter switch. Don't know
why they bothered with the VU one.

--
*Rehab is for quitters.

Dave Plowman London SW
To e-mail, change noise into sound.

[Mike Rivers[_2_]]

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?




--
For a good time, visit http://mikeriversaudio.wordpress.com

[John Williamson]

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.

[hank alrich]

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.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

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.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

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??

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

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.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

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?

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

PStamler > wrote:

> you're simply fumbling in the dark

That would be a step up.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

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!?!

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[hank alrich]

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.

--
shut up and play your guitar * HankAlrich.Com
HankandShaidriMusic.Com
YouTube.Com/WalkinayMusic

[geoff]

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

[Gary Eickmeier]

"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

[None]

"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?

[Peter Larsen[_3_]]

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

[Gary Eickmeier]

"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
>
>

[None]

"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.

[[email protected]]

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!!

[Scott Dorsey]

> 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

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

[geoff]

On 20/08/2014 12:26 a.m., 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?!
>
> Agreed Paul, and I wish this F~~~ING "use all the bits" mentality would disappear completely from mother Earth ONCE AND FOR FRICKIN' ALL!!
>


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
with clipping.

If "max bits" use in the wrong situation in a recording process, it can
make accidental clipping more easily caused with subsequent processing,
if not vigilant. But that's a separate issue.

geoff

[[email protected]]

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.

[Sean Conolly]

"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

[Gary Eickmeier]

> 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

[Gary Eickmeier]

"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

[Sean Conolly]

"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

[PStamler]

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

[Peter Larsen[_3_]]

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

[geoff]

On 20/08/2014 10:58 a.m., wrote:
> 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.
>


Thank you. None of us here knew that.

geoff

[[email protected]]

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?

[None]

> 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.

[Scott Dorsey]

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.

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."

[Scott Dorsey]

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."

[Scott Dorsey]

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."

[hank alrich]

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

[PStamler]

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

[geoff]

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

[None]

"geoff" > wrote in message
...
> 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 ?!!!

Krissie Kretin has been reading up on the numbers from one to ten. All
he needs is a simple answer: Is ten the biggie or the smallie? All
that "reading up" didn't sink in.