Freq Response Graph Paper [Archive]

Gary Eickmeier

May 8th 14, 01:56 PM

"John Williamson" > wrote in message
...
> On 08/05/2014 04:46, Gary Eickmeier wrote:
>> Thanks all for the info - will try those searches and see what pops up.
>> My
>> earlier search resulted in one of those nightmares of links within links
>> full of sites that want to download all kinds of crap to you beside what
>> you
>> are after.
>>
>> On equalizers, I wonder why they aren't all digital yet, with infinitely
>> fine number of sliders that are set automatically by the RTA. I know that
>> some receivers have these "room correction" modes, right? I would love to
>> see one that measures the output of your system, then allows you to draw
>> the
>> curve that you want, and have it do it perfectly with no lumpies. Ah
>> well,
>> we'll see now it goes.
>>
> There are systems that go one stage further. You play a sweep with a
> microphone at the listening position, and it automatically sets the
> frequency response to give you a flat(ish) response there.
>
> They're not cheap...

Well, what I mean is you don't necessarily want flat, but a "room curve" at
the listening position. McIntosh used to publish such a curve that their
technicians used in the field to set up their speaker systems when
requested. It has a slight hump below 1k and a gradually falling hi freq
beyond that.

Gary Eickmeier

Gary Eickmeier

May 8th 14, 02:01 PM

"Adrian Tuddenham" > wrote in message
valid.invalid...
> Gary Eickmeier > wrote:
>
>
>> Anyone know where I can beg, buy, or steal some FR graph paper? The
>> measurement equipment companies? I was actually hoping to be able to have
>> it
>> plot the readings for me and then print it out, but I will take anything.
>
> Try:
> http://www.poppyrecords.co.uk/other/Graphs/GraphPaper0-80.gif
> http://www.poppyrecords.co.uk/other/Graphs/GraphPaper20-20.gif
>
> They took about 5 minutes to knock up with ClarisWorks4 spreadsheet; and
> it would also have plotted the graphs for me if I had put in some data.

Beautiful - thanks Adrian.

Gary

William Sommerwerck

May 8th 14, 02:21 PM

"Adrian Tuddenham" wrote in message
nvalid.invalid...

> Why do people persist in calling it an "equaliser"? It is an effects
> unit and does not equalise anything.

Its original purpose was to remove known errors in frequency response. But, of
course, as with many inventions, its use has been perverted.

Current consumer systems can automatically flatten the response at the
listening position. These have been around for several years.

Mike Rivers[_2_]

May 8th 14, 02:43 PM

On 5/7/2014 3:25 PM, Gary Eickmeier wrote:
> I can't believe how difficult it is to find some audio frequency response
> graph paper.

You want semi-log graph paper, four cycles if you want the full audio
range (10 Hz to 100 kHz). Here's a "make your own" web site that's
pretty flexible:
http://customgraph.com/piart.php?art=579

What I'm surprised that I can't find, given how common computers with
sound cards are, is a modern computerized version of the clunky General
Radio chain-driven synchronized oscillator and plotter that we had in
our college lab in 1960. Connect the device you want to test between the
audio output and input of a computer, use a program to generate a slow
sine wave sweep, and generate a plot of what comes back into the
computer's audio input. There are a number of FFT programs but it's just
not the same thing.

I've tried Room EQ Wizard and RightMark but haven't had much success
with either one. This seems so simple. Why hasn't anyone done it?

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

Mike Rivers[_2_]

May 8th 14, 02:48 PM

On 5/8/2014 4:36 AM, Adrian Tuddenham wrote:
> They took about 5 minutes to knock up with ClarisWorks4 spreadsheet; and
> it would also have plotted the graphs for me if I had put in some data.

I've spent years fighting with Excel's plotting routines and still
haven't been able to figure out how to make a normal looking frequency
response graph. Is ClarisWorks a Mac-only program? It's an old name but
I haven't heard much about it in at least 10 years.

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

Scott Dorsey

May 8th 14, 03:52 PM

Gary Eickmeier > wrote:
>
>On equalizers, I wonder why they aren't all digital yet, with infinitely
>fine number of sliders that are set automatically by the RTA. I know that
>some receivers have these "room correction" modes, right? I would love to
>see one that measures the output of your system, then allows you to draw the
>curve that you want, and have it do it perfectly with no lumpies. Ah well,
>we'll see now it goes.

These days, a lot of them _are_ digital. But in the studio world, people
use parametrics instead of graphics so they don't have any of those problems
to begin with.
--scott

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

Scott Dorsey

May 8th 14, 03:53 PM

John Williamson > wrote:
>>
>There are systems that go one stage further. You play a sweep with a
>microphone at the listening position, and it automatically sets the
>frequency response to give you a flat(ish) response there.
>
>They're not cheap...

But the end result sure sounds that way....
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Adrian Tuddenham[_2_]

May 8th 14, 03:58 PM

Mike Rivers > wrote:

> On 5/8/2014 4:36 AM, Adrian Tuddenham wrote: > They took about 5 minutes
> to knock up with ClarisWorks4 spreadsheet; and > it would also have
> plotted the graphs for me if I had put in some data.
>
> I've spent years fighting with Excel's plotting routines and still
> haven't been able to figure out how to make a normal looking frequency
> response graph. Is ClarisWorks a Mac-only program? It's an old name but
> I haven't heard much about it in at least 10 years.

It was developed in the early 1990s; the spreadsheet was based on
Lotus123 (Excel also came from the same origins), the drawing section
started as MacDraw and it also integrated a database, a word processor,
painting and communications packages (it could emulate a Teletype and
program a modem). Version 4, which I still use, was from 1995 and was
the last one that worked as an industrial heavyweight before they added
all the eye-candy. In conjunction with "Pub & Sub" (now also
discontinued in OSX) it could provide all the software to run a
medium-sized business.

Although the full version is only meant for Macs (because the comms
module doesn't work on P.C. hardware) there is a version for Windows
(CW5) and documents are interchangeable between the two platforms as
long as Mac users remember to put the ".cwk" file extension on the file
name.

It exports drawing files in PICT format, which any photographic program
can convert to GIF or JPEG.

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

Scott Dorsey

May 8th 14, 04:00 PM

Adrian Tuddenham > wrote:
>
>Why do people persist in calling it an "equaliser"? It is an effects
>unit and does not equalise anything

It is a historical holdover from the days of telephone practice when filter
banks _were_ used to equalize line response.

Now we are stuck with it, just as we are stuck with "condenser microphones"
that do not condense any fluid and "passive preamplifiers" that are actually
attenuators.

Please notify the Oxford University Press.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Scott Dorsey

May 8th 14, 04:03 PM

Gary Eickmeier > wrote:
>"John Williamson" > wrote in message
>> They're not cheap...
>
>Well, what I mean is you don't necessarily want flat, but a "room curve" at
>the listening position. McIntosh used to publish such a curve that their
>technicians used in the field to set up their speaker systems when
>requested. It has a slight hump below 1k and a gradually falling hi freq
>beyond that.

Yes, and the end result was terrible sound... because making everything
so that all third-octave sections have the same level is NOT the same as
making the system "flat," even ignoring the non-minimum-phase effects.

Thank God that the Seventies are over and people have pretty much abandoned
all that stuff.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Sylvain Robitaille

May 8th 14, 05:01 PM

On Thu, 08 May 2014 09:43:50 -0400, Mike Rivers wrote:

> What I'm surprised that I can't find, given how common computers
> with sound cards are, is a modern computerized version of the
> clunky General Radio chain-driven synchronized oscillator and
> plotter that we had in our college lab in 1960. Connect the device
> you want to test between the audio output and input of a computer,
> use a program to generate a slow sine wave sweep, and generate a
> plot of what comes back into the computer's audio input. ... This
> seems so simple. Why hasn't anyone done it?

Possibly something like this?

http://kokkinizita.linuxaudio.org/linuxaudio/jaaa-pict.html

Admittedly I'm not sure whether it'll do a frequency sweep. I don't use
this software very frequently, and I've not tried to make it do that,
but it seems to be pretty good at giving the user a sense of frequency
response of connected equipment (photos in the link above are of noise
floor not frequency response, I know ...)

--
----------------------------------------------------------------------
Sylvain Robitaille

Systems analyst / AITS Concordia University
Faculty of Engineering and Computer Science Montreal, Quebec, Canada
----------------------------------------------------------------------

Peter Larsen[_3_]

May 8th 14, 05:38 PM

Gary Eickmeier wrote:

> I can't believe how difficult it is to find some audio frequency
> response graph paper. I am doing some measuring and would like to
> plot it from time to time, getting a new 31 band equalizer, just want
> to go to the internet and download some FR graph paper. No dice. I
> have been to several sites that offer free this, free that,
> Wikipedia, have run out of key words to try.

Solving what problem? - measure at iso frequencies and plot on standard
"squares" and be happy.

> Gary Eickmeier

Kind regards

Peter Larsen

Peter Larsen[_3_]

May 8th 14, 05:46 PM

Gary Eickmeier wrote:

> "John Williamson" > wrote in message
> ...

>> There are systems that go one stage further. You play a sweep with a
>> microphone at the listening position, and it automatically sets the
>> frequency response to give you a flat(ish) response there.

>> They're not cheap...

Some actually are, considering that they are built into AV amplifiers.

> Well, what I mean is you don't necessarily want flat, but a "room
> curve" at the listening position. McIntosh used to publish such a
> curve that their technicians used in the field to set up their
> speaker systems when requested. It has a slight hump below 1k and a
> gradually falling hi freq beyond that.

20 Hz - 6 dB, 200 Hz 0 dB and 20 kHz -6 dB is a good starting point for a 20
to 40 square meter room, larger room make it 20 kHz -10 dB, beyond
"larger" - whatever that is - the theather curve 40 to 1000 linear and -3 dB
/ octave above gets to be the target curve. With a sanely designed
loudspeaker system those curves will be what the system tries to provide.

Note that it is often claimed that it malpractice to move sliders on a
graphic eq that are next to each other to opposites, ie. boost one and
attenuate the next. That is misunderstood because just that is how to move
the adjustment center sideways as is "easily seen" in 10 minutes when
equalizing with an analyzer running.

Two important points: the treble may not appear to be detached and a
subwoofer, if available, may not be detectable on male vox.

> Gary Eickmeier

Kind regards

Peter Larsen

Peter Larsen[_3_]

May 8th 14, 05:51 PM

Mike Rivers wrote:

> On 5/7/2014 3:25 PM, Gary Eickmeier wrote:
>> I can't believe how difficult it is to find some audio frequency
>> response graph paper.

> You want semi-log graph paper, four cycles if you want the full audio
> range (10 Hz to 100 kHz). Here's a "make your own" web site that's
> pretty flexible:
> http://customgraph.com/piart.php?art=579

> What I'm surprised that I can't find, given how common computers with
> sound cards are, is a modern computerized version of the clunky
> General Radio chain-driven synchronized oscillator and plotter that
> we had in our college lab in 1960. Connect the device you want to
> test between the audio output and input of a computer, use a program
> to generate a slow sine wave sweep, and generate a plot of what comes
> back into the computer's audio input. There are a number of FFT
> programs but it's just not the same thing.

You want Speaker Workshop by Audua.

> I've tried Room EQ Wizard and RightMark but haven't had much success
> with either one. This seems so simple. Why hasn't anyone done it?

His marketing flunked, but it is should be out there somewhere, I may still
have the download archive if an old computer powers up as expected.

Kind regards

Peter Larsen

Don Pearce[_3_]

May 8th 14, 06:03 PM

On Thu, 08 May 2014 09:43:50 -0400, Mike Rivers >
wrote:

>On 5/7/2014 3:25 PM, Gary Eickmeier wrote:
>> I can't believe how difficult it is to find some audio frequency response
>> graph paper.
>
>You want semi-log graph paper, four cycles if you want the full audio
>range (10 Hz to 100 kHz). Here's a "make your own" web site that's
>pretty flexible:
>http://customgraph.com/piart.php?art=579
>
>What I'm surprised that I can't find, given how common computers with
>sound cards are, is a modern computerized version of the clunky General
>Radio chain-driven synchronized oscillator and plotter that we had in
>our college lab in 1960. Connect the device you want to test between the
>audio output and input of a computer, use a program to generate a slow
>sine wave sweep, and generate a plot of what comes back into the
>computer's audio input. There are a number of FFT programs but it's just
>not the same thing.
>
>I've tried Room EQ Wizard and RightMark but haven't had much success
>with either one. This seems so simple. Why hasn't anyone done it?

Not surprising. You can't equalise a room. You can equalise the path
from one speaker to one ear provided you don't move, and that is it.
You know as well as I do that if you move more than a few inches the
modal patterns surrounding you have changed utterly. "Equalizing" at
one position makes things ten times as bad at another. The best you
can ever do is make the source as good as possible.

d

Gary Eickmeier

May 8th 14, 06:15 PM

"Peter Larsen" > wrote in message
k...
> Gary Eickmeier wrote:
>
>> I can't believe how difficult it is to find some audio frequency
>> response graph paper. I am doing some measuring and would like to
>> plot it from time to time, getting a new 31 band equalizer, just want
>> to go to the internet and download some FR graph paper. No dice. I
>> have been to several sites that offer free this, free that,
>> Wikipedia, have run out of key words to try.
>
> Solving what problem? - measure at iso frequencies and plot on standard
> "squares" and be happy.

Hi Peter -

I sort of ended up doing that, before I got all the responses above. In the
Bruel & Kjaer test CD the booklet has a teensy tiny FR graph that goes along
perfectly with the recorded pink noise narrow band signals. So I scanned it
into my photo edit program, enlarged it to sheet of paper size, and then I
can plot the FR directly above the test band freqs listed along the
horizontal axis. It is also great because it has the vertical axis in dB
from 50 to 100 dB, so I can plot the readings directly without having to
interpolate or translate.

Hey Peter - you are the one with the "orange" freq response plot theory for
recordings, right? That thing has fascinated me ever since. This is the fact
that there is a rising response up to 100 Hz, then a gradually falling
response the rest of the way. For the readers who haven't seen it, it is
based on the frequency analysis graph that probably most audio edit programs
have. You can highlight any section of your track, and that curve will
almoat always hold true. If you have an unwelcome resonance or deviation,
this window will tell you what to cut and how broadly.

Why does this work so well????

Gary Eickmeier

Mike Rivers[_2_]

May 8th 14, 06:22 PM

On 5/8/2014 1:03 PM, Don Pearce wrote:
>> I've tried Room EQ Wizard and RightMark but haven't had much success
>> >with either one. This seems so simple. Why hasn't anyone done it?

> Not surprising. You can't equalise a room.

I know that, and I was never trying to equalize a room. I was using it
like I described it, pretending that the device I was testing was the
room. Room EQ Wizard wasn't simple enough to remember how to work, and
RightMark was never happy with the input level. Either it clipped or it
said that it was too low.

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

Mike Rivers[_2_]

May 8th 14, 06:32 PM

On 5/8/2014 12:51 PM, Peter Larsen wrote:
> Speaker Workshop by Audua.

That looks like it's an FFT sort of program. I want a simple sine sweep
that measures the amplitude at one frequency, them moves on to the next
frequency. I can do this with my NTI Minirator and Minilizer, but it
generates only at 1/3 octave steps. That's not good enough resolution to
see what the response of an equalizer really looks like.

I'm not looking to measure rooms or speakers here, I want to measure
hardware with analog inputs and outputs. I can do it by hand of course,
it's just that this seems like the sort of thing that a computer can do
well, if only someone bothered to write the program.

Actually, Chris Juried of Tube Equipment Corporation has just what I'm
looking for, and he's offered to send me a copy a few times now, but
hasn't, so I guess it may not be ready to get out of the shop yet. He
uses it to characterize transformers and tubes.

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

Frank Stearns

May 8th 14, 06:38 PM

(Scott Dorsey) writes:

>John Williamson > wrote:
>>>
>>There are systems that go one stage further. You play a sweep with a
>>microphone at the listening position, and it automatically sets the
>>frequency response to give you a flat(ish) response there.
>>
>>They're not cheap...

>But the end result sure sounds that way....

Chuckle, no kidding. Freq response is only one thing - uneven or inappropriate
reverb times and phase (time) irregularities being the others, often more
significant. All you can do with graphic EQ is typically make it worse, even though
it "measures" flat. Woo hoo. Parametric is "better" but is still only a partial
solution for many underlying problems.

But hey, whatever floats your boat or bumps your already peaky/dippy bass. (For
that, traps are highly recommended.)

Maybe I missed it. Tell me again, Gary, why you're looking for category of solution,
particularly this one? I thought you were happy with your room.

Frank
Mobile Audio
--

William Sommerwerck

May 8th 14, 06:46 PM

"Scott Dorsey" wrote in message ...

> ...the end result was terrible sound... because making everything
> so that all third-octave sections have the same level is NOT the
> same as making the system "flat," even ignoring the non-minimum-
> phase effects.

I did a fair number of consumer equalizations, all intended to produced "flat"
response at the listening position. If the room wasn't overly reverberant, the
result was invariably an improvement -- lower coloration, better imaging, etc.

> Thank God that the Seventies are over and people have pretty
> much abandoned all that stuff.

Current EQ systems don't necessarily aim for flat steady-state response. But I
haven't studied them, and I'm not sure how they work.

William Sommerwerck

May 8th 14, 06:49 PM

"Scott Dorsey" wrote in message ...

> It is a historical holdover from the days of telephone practice when
> filter banks _were_ used to equalize line response.

> Now we are stuck with it, just as we are stuck with "condenser
> microphones" that do not condense any fluid...

You are mistaken, sir... They condense the electrical fluid!

Gary Eickmeier

May 9th 14, 03:51 AM

"Frank Stearns" > wrote in message
...

> Chuckle, no kidding. Freq response is only one thing - uneven or
> inappropriate
> reverb times and phase (time) irregularities being the others, often more
> significant. All you can do with graphic EQ is typically make it worse,
> even though
> it "measures" flat. Woo hoo. Parametric is "better" but is still only a
> partial
> solution for many underlying problems.
>
> But hey, whatever floats your boat or bumps your already peaky/dippy bass.
> (For
> that, traps are highly recommended.)
>
> Maybe I missed it. Tell me again, Gary, why you're looking for category of
> solution,
> particularly this one? I thought you were happy with your room.
>
> Frank
> Mobile Audio

Hi Frank -

Whew - I am getting some odd reactions about the whole subject of EQing a
room or a speaker system. Basically, my system has no tone controls on the
receiver. I have no big complaints, but some of my audio buddies are always
wanting me to measure the FR in my system, so I did. Just a Radio Shack SLM,
digital, and a B&K test CD with 30 bands of narrow pink noise, but it works
and reveals some anomolies that I cannot correct with my system as is. I am
using a Velodyne subwoofer and just setting it by ear. So in measuring at
the listening position, I am getting a hump at 63 Hz that is about 5 dB
higher than I would like, then fairly smooth thru the midrange from 100 Hz
to 5k, then another hump at 8k of 5 or 6 dB too high, then falling off
smoothly to 20k with the zero crossing at 12.5 and 8 dB down at 16k.

When we did the Linkwitz Challenge with my cheap hacked together prototype
emulating my radiation pattern ideas, the test designer Dave Clark was able
to EQ both challenger speakers, mine and a pair of Behringer box speakers,
to sound just like the Linkwitz Orions in freq response. This was necessary
to eliminate that one variable from the one under test, the radiation
pattern and its effect on imaging. It was impressive to me that he was able
to do this with mine and make it sound so good that it won the challenge.

I am now building, or having built by someone that knows how to build
speakers, a final prototype that my engineer is doing with computer models
and by ear on the voicing and a variable radiation pattrn etc etc for which
I predict it will be very useful to have the ability to shape the response
of the things with a 31 band. I will be using the Velodyne with those new
speakers too, and this Behringer FBQ6200 equalizer has a subwoofer output
and adjustable crossover freq that is almost designed for my situation. I
will be able to shape the sub bass curve as well as the upper range so that
the FR is not a factor in the audibility of my design for rad pat and room
positioning of speakers for soundstaging. The engineer has already listened
in mono to the first one, and he remarks that it has the ability to "shape"
the soundstage anywhere from too far forward and too small, to too far
rearward and too flat a presentation (imaging wise, not FR wise). This is
exactly what I wanted, and should prove educational as well as spectacular
in stereo. To screw this up by not being able to voice it to my room would
be a true tragedy!

Gary

Gary Eickmeier

May 9th 14, 03:52 AM

"Scott Dorsey" > wrote in message
...
> Gary Eickmeier > wrote:
>>
>>On equalizers, I wonder why they aren't all digital yet, with infinitely
>>fine number of sliders that are set automatically by the RTA. I know that
>>some receivers have these "room correction" modes, right? I would love to
>>see one that measures the output of your system, then allows you to draw
>>the
>>curve that you want, and have it do it perfectly with no lumpies. Ah well,
>>we'll see now it goes.
>
> These days, a lot of them _are_ digital. But in the studio world, people
> use parametrics instead of graphics so they don't have any of those
> problems
> to begin with.
> --scott

Is that because a parametric has a broader Q than a 31 band graphic?

Gary

John Williamson

May 9th 14, 08:54 AM

On 09/05/2014 03:52, Gary Eickmeier wrote:
> "Scott Dorsey" > wrote in message
>> These days, a lot of them _are_ digital. But in the studio world, people
>> use parametrics instead of graphics so they don't have any of those
>> problems
>> to begin with.
>> --scott
>
> Is that because a parametric has a broader Q than a 31 band graphic?
>
A parametric can be set up to have almost any Q that you wish at any
centre frequency that you wish. You could, if that's what's wanted, set
it to almost completely take out a band a tenth of an octave wide,
leaving the rest of the spectrum undisturbed. Or pass only that tenth of
an octave band while almost removing the rest of the spectrum. Or you
can set it to give a 0.5 dB or less lift or cut spread over a five
octave bandwidth.

--
Tciao for Now!

John.

Adrian Tuddenham[_2_]

May 9th 14, 08:56 AM

Gary Eickmeier > wrote:

> "Scott Dorsey" > wrote in message
> ...
> > Gary Eickmeier > wrote:
> >>
> >>On equalizers, I wonder why they aren't all digital yet, with infinitely
> >>fine number of sliders that are set automatically by the RTA. I know that
> >>some receivers have these "room correction" modes, right? I would love to
> >>see one that measures the output of your system, then allows you to draw
> >>the
> >>curve that you want, and have it do it perfectly with no lumpies. Ah well,
> >>we'll see now it goes.
> >
> > These days, a lot of them _are_ digital. But in the studio world, people
> > use parametrics instead of graphics so they don't have any of those
> > problems
> > to begin with.
> > --scott
>
> Is that because a parametric has a broader Q than a 31 band graphic?

Many of the frequency/phase distortions that need equalising are caused
by resonant systems and their effect on the response can be represented
by three parameters: Q, centre frequency and amplitude. The parametric
equaliser allows all three to be adjusted so as to accurately counteract
the distortion.

The graphic effects unit has fixed frequencies and fixed Q factors,
which means that only the amplitude can be adjusted, therefore it
incapable of being used as an equaliser (unless the distortions happen
to lie exactly at the correct frequencies with the correct Q factors).

A similar relationship exists between proper equaliser networks and
"tone controls".

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

Scott Dorsey

May 9th 14, 02:27 PM

Peter Larsen > wrote:
>
>Note that it is often claimed that it malpractice to move sliders on a
>graphic eq that are next to each other to opposites, ie. boost one and
>attenuate the next. That is misunderstood because just that is how to move
>the adjustment center sideways as is "easily seen" in 10 minutes when
>equalizing with an analyzer running.

Well, it's almost always a bad thing to do that becauyse invariably if you
see two adjacent bands that are considerably different in level, you're apt
to be seeing narrowband problems that you can't fix with a graphic equalizer
and attempting to fix them that way is apt to cause more harm than good.

If you're trying to equalize something with an analyzer whose filters are
no more narrow than the equalizer you're using, you're going to be doing
more harm than good. The good news is that we now live with inexpensive
FFT systems that can show you what the narrowband response really is, so you
know what you can fix annd what you can't.

>Two important points: the treble may not appear to be detached and a
>subwoofer, if available, may not be detectable on male vox.

Detached?
--scott

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

Scott Dorsey

May 9th 14, 02:29 PM

Gary Eickmeier > wrote:
>Whew - I am getting some odd reactions about the whole subject of EQing a
>room or a speaker system. Basically, my system has no tone controls on the
>receiver. I have no big complaints, but some of my audio buddies are always
>wanting me to measure the FR in my system, so I did. Just a Radio Shack SLM,
>digital, and a B&K test CD with 30 bands of narrow pink noise, but it works
>and reveals some anomolies that I cannot correct with my system as is. I am
>using a Velodyne subwoofer and just setting it by ear. So in measuring at
>the listening position, I am getting a hump at 63 Hz that is about 5 dB
>higher than I would like, then fairly smooth thru the midrange from 100 Hz
>to 5k, then another hump at 8k of 5 or 6 dB too high, then falling off
>smoothly to 20k with the zero crossing at 12.5 and 8 dB down at 16k.

And that 63 Hz hump moves around, depending on where you do the measurement.

It sounds like you have some serious room problems. Equalization does not
fix room problems, because the frequency response aberration is only the
_symptom_ of a time-domain problem. Fix the room.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Scott Dorsey

May 9th 14, 02:32 PM

Gary Eickmeier > wrote:
>"Scott Dorsey" > wrote in message
>> Gary Eickmeier > wrote:
>>>
>>>On equalizers, I wonder why they aren't all digital yet, with infinitely
>>>fine number of sliders that are set automatically by the RTA. I know that
>>>some receivers have these "room correction" modes, right? I would love to
>>>see one that measures the output of your system, then allows you to draw
>>>the
>>>curve that you want, and have it do it perfectly with no lumpies. Ah well,
>>>we'll see now it goes.
>>
>> These days, a lot of them _are_ digital. But in the studio world, people
>> use parametrics instead of graphics so they don't have any of those
>> problems
>> to begin with.
>
>Is that because a parametric has a broader Q than a 31 band graphic?

It's because a parametric has whatever Q you want it to have, so you can
match whatever it is that you're trying to deal with whether it is a narrowband
problem or a wideband one.

I can notch out PA feedback in a recording without touching the notes a
semitone above or below the feedback tone, with the Q very narrow. I can
make a very minor reduction in the top end with the Q very wide. I just
set the knob for where it needs to go.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Peter Larsen[_3_]

May 9th 14, 03:59 PM

Scott Dorsey wrote:

> Peter Larsen > wrote:

>> Two important points: the treble may not appear to be detached and a
>> subwoofer, if available, may not be detectable on male vox.

> Detached?

Yes, apparently elevated after a dip, something like:

[perception]

xxxxxxxxxxxxxxxxxxxxxxxxxx xxxxxxxx
xxxxxx x
xxxxx

> --scott

Kind regards

Peter Larsen

Gary Eickmeier

May 9th 14, 04:43 PM

"Scott Dorsey" > wrote in message
...

> It's because a parametric has whatever Q you want it to have, so you can
> match whatever it is that you're trying to deal with whether it is a
> narrowband
> problem or a wideband one.
>
> I can notch out PA feedback in a recording without touching the notes a
> semitone above or below the feedback tone, with the Q very narrow. I can
> make a very minor reduction in the top end with the Q very wide. I just
> set the knob for where it needs to go.
> --scott

I suppose I should look up a few of the parametric kind and see just what
they can do, and for how much. It does seem quite ideal to be able to adjust
those factors, if you know just what they are with a really good RTA. But
how many center freqs do most of them have? In my case I have three, maybe
four freqs at which I would like to make a counter-adjustment.

On your room comment, one thing that happened is that when I moved the
Velodyne from two or three ft out from the corner along one side wall to
right into the corner, the bass response jumped up so much that I had to
completely reset the level control. I do believe in corner placement, but I
didn't realize it would make that kind of difference. Anyway, my response is
quite smooth and flat other than those two anomolies, one at 63 and one at
8k. I think the 8k one is the main reason I wanted to do something about my
system, my big band recordings sounding a little to bright, horns not as
"creamy smooth" as in real life but everything else wonderful.

Gary Eickmeier

Adrian Tuddenham[_2_]

May 9th 14, 06:02 PM

Gary Eickmeier > wrote:

> "Scott Dorsey" > wrote in message
> ...
>
>
> > It's because a parametric has whatever Q you want it to have, so you can
> > match whatever it is that you're trying to deal with whether it is a
> > narrowband
> > problem or a wideband one.
[...]
> I suppose I should look up a few of the parametric kind and see just what
> they can do,

Unless they have adjustable centre frequencies and Qs, they can never
equalise resonance-related problems correctly. Attempting to equalise a
resonance with a peak or notch that is even slightly off-tune will
produce horrible phasey-sounding effects on wideband sounds.

I have spent countless hours trying to equalise badly-recorded 78s with
parametric equalisers and I know just how difficult it is. That was
with no more than three resonances and no delay comb effects. The
chances of doing that properly with a conventional graphic effects unit
are nil.

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

hank alrich

May 9th 14, 07:37 PM

Scott Dorsey > wrote:

> Gary Eickmeier > wrote:
> >
> >On equalizers, I wonder why they aren't all digital yet, with infinitely
> >fine number of sliders that are set automatically by the RTA. I know that
> >some receivers have these "room correction" modes, right? I would love to
> >see one that measures the output of your system, then allows you to draw the
> >curve that you want, and have it do it perfectly with no lumpies. Ah well,
> >we'll see now it goes.
>
> These days, a lot of them _are_ digital. But in the studio world, people
> use parametrics instead of graphics so they don't have any of those problems
> to begin with.
> --scott

The Behringer DEQ2496 is a good learning tool, and a good tool period in
experienced hands. Two channels of multi-band, parametric, dynamic,
auto, etc., EQ in one box.

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

Gary Eickmeier

May 9th 14, 08:35 PM

"hank alrich" > wrote in message
...

> The Behringer DEQ2496 is a good learning tool, and a good tool period in
> experienced hands. Two channels of multi-band, parametric, dynamic,
> auto, etc., EQ in one box.

Wow thanks Hank - this is serious. Their only critique was the reliability
and support from the company. But it has an RTA! I already have a good
calibration microphone from another unit that might work OK. There is no
dedicated one for this unit anyway.

Got to think long and hard on this one.

Gary

Ron C[_2_]

May 9th 14, 10:42 PM

On 5/9/2014 3:35 PM, Gary Eickmeier wrote:
>
> "hank alrich" > wrote in message
> ...
>
>> The Behringer DEQ2496 is a good learning tool, and a good tool period in
>> experienced hands. Two channels of multi-band, parametric, dynamic,
>> auto, etc., EQ in one box.
>
> Wow thanks Hank - this is serious. Their only critique was the reliability
> and support from the company. But it has an RTA! I already have a good
> calibration microphone from another unit that might work OK. There is no
> dedicated one for this unit anyway.
>
> Got to think long and hard on this one.
>
> Gary
>
>
This is an amazingly flexible unit for the price. I installed
three of them in my old venue and I have two of my own.
However, I did have three out of the box failures (DOA)
and two that crashed after about a year of regular use.
I traced the failures to overheating. [IMHO the things
could use better ventilation or forced air cooling.]

==
Later...
Ron Capik
--

Ron C[_2_]

May 9th 14, 10:42 PM

On 5/9/2014 1:02 PM, Adrian Tuddenham wrote:
> Gary Eickmeier > wrote:
>
>> "Scott Dorsey" > wrote in message
>> ...
>>
>>
>>> It's because a parametric has whatever Q you want it to have, so you can
>>> match whatever it is that you're trying to deal with whether it is a
>>> narrowband
>>> problem or a wideband one.
> [...]
>> I suppose I should look up a few of the parametric kind and see just what
>> they can do,
>
> Unless they have adjustable centre frequencies and Qs, they can never
> equalise resonance-related problems correctly. Attempting to equalise a
> resonance with a peak or notch that is even slightly off-tune will
> produce horrible phasey-sounding effects on wideband sounds.
>
> I have spent countless hours trying to equalise badly-recorded 78s with
> parametric equalisers and I know just how difficult it is. That was
> with no more than three resonances and no delay comb effects. The
> chances of doing that properly with a conventional graphic effects unit
> are nil.
>
>
Adrian,
Have you tried time reversal on those 78s? One can't
un-ring a bell in real time but can get a lot closer in
reverse time. Try reversing the track and applying
the correction. Also note that some DSP EQs don't
have the same Q for boost and cut. I did some experiments
on this with test signals (broad band noise) in sound forge.
I'd boost a frequency then apply the inverse, same Q and
frequency. Forward boost, forward cut showed phase errors
(time smearing) where forward boost, reverse cut was nearly
error free. The higher the Q the greater the time smearing.

Getting back to the topic at hand, the EQ in a room can only
be applied in real time, so the time smearing will be additive
even if the average power is flattened. Put another way: you
can't suck the energy out of a ringing cavity, you can only
reduce the excitation power.

==
Later...
Ron Capik
--

Frank Stearns

May 10th 14, 12:56 AM

"Gary Eickmeier" > writes:

>"Frank Stearns" > wrote in message
...

>> Chuckle, no kidding. Freq response is only one thing - uneven or
>> inappropriate
>> reverb times and phase (time) irregularities being the others, often more
>> significant. All you can do with graphic EQ is typically make it worse,
>> even though
>> it "measures" flat. Woo hoo. Parametric is "better" but is still only a
>> partial
>> solution for many underlying problems.
>>
>> But hey, whatever floats your boat or bumps your already peaky/dippy bass.
>> (For
>> that, traps are highly recommended.)
>>
>> Maybe I missed it. Tell me again, Gary, why you're looking for category of
>> solution,
>> particularly this one? I thought you were happy with your room.
>>
>> Frank
>> Mobile Audio

>Hi Frank -

>Whew - I am getting some odd reactions about the whole subject of EQing a
>room or a speaker system. Basically, my system has no tone controls on the
>receiver. I have no big complaints, but some of my audio buddies are always
>wanting me to measure the FR in my system, so I did. Just a Radio Shack SLM,
>digital, and a B&K test CD with 30 bands of narrow pink noise, but it works
>and reveals some anomolies that I cannot correct with my system as is. I am
>using a Velodyne subwoofer and just setting it by ear. So in measuring at
>the listening position, I am getting a hump at 63 Hz that is about 5 dB
>higher than I would like, then fairly smooth thru the midrange from 100 Hz
>to 5k, then another hump at 8k of 5 or 6 dB too high, then falling off
>smoothly to 20k with the zero crossing at 12.5 and 8 dB down at 16k.

-snips-

Well, you've hit the nail on the head why so many of us are fanatical about room
treatment.

Choice 1: An EQ box to "correct" room issues

Choice 2: Measurement gear, RealTraps (or DIY traps made of framing lumber, panel
material, 703, caulking, and a range of tools), diffusors, absorbers, patience,
skill.

Choice 1 is easy, but seldom (if ever) works because it simply doesn't go after the
most likely root causes of the problem(s), especially from 200 hz on down.

Choice 2, when done right, produces incredible results. It's sound like you've
never heard it, especially with reasonably good monitors.

YMMV.

Frank
Mobile Audio

--

Gary Eickmeier

May 10th 14, 06:58 AM

"Frank Stearns" > wrote in message
acquisition...
> "Gary Eickmeier" > writes:

>>Hi Frank -
>
>>Whew - I am getting some odd reactions about the whole subject of EQing a
>>room or a speaker system. Basically, my system has no tone controls on the
>>receiver. I have no big complaints, but some of my audio buddies are
>>always
>>wanting me to measure the FR in my system, so I did. Just a Radio Shack
>>SLM,
>>digital, and a B&K test CD with 30 bands of narrow pink noise, but it
>>works
>>and reveals some anomolies that I cannot correct with my system as is. I
>>am
>>using a Velodyne subwoofer and just setting it by ear. So in measuring at
>>the listening position, I am getting a hump at 63 Hz that is about 5 dB
>>higher than I would like, then fairly smooth thru the midrange from 100 Hz
>>to 5k, then another hump at 8k of 5 or 6 dB too high, then falling off
>>smoothly to 20k with the zero crossing at 12.5 and 8 dB down at 16k.
>
> -snips-
>
> Well, you've hit the nail on the head why so many of us are fanatical
> about room
> treatment.
>
> Choice 1: An EQ box to "correct" room issues
>
> Choice 2: Measurement gear, RealTraps (or DIY traps made of framing
> lumber, panel
> material, 703, caulking, and a range of tools), diffusors, absorbers,
> patience,
> skill.
>
>
> Choice 1 is easy, but seldom (if ever) works because it simply doesn't go
> after the
> most likely root causes of the problem(s), especially from 200 hz on down.
>
> Choice 2, when done right, produces incredible results. It's sound like
> you've
> never heard it, especially with reasonably good monitors.
>
> YMMV.
>
> Frank
> Mobile Audio

OK, I can believe you about the bass traps, but what would cause an 8k hump?
I realize that I am not making super careful measurements yet, but I still
see no reason for this.

Gary

Peter Larsen[_3_]

May 10th 14, 07:27 AM

Gary Eickmeier wrote:

> OK, I can believe you about the bass traps, but what would cause an
> 8k hump?

Loudspeaker linearity issue comes to mind as - in my opinion, I didn't say
experience - the primary and usual suspect, but do also consider that glass
has the property of letting some frequency ranges through and reflecting
others.

Actually making valid measurements in a room is not at all simple, Speaker
Workshop did it to my liking well when I tried it some years ago, but I
liked smoothing 1/4 octave warble-tone measurements in a calcsheet even
better.

Note: when you want to make the room "behave" smoothing information about
sharp peaks and dips away from the measurement may not be the best choice.

Kind regards

Peter Larsen

Adrian Tuddenham[_2_]

May 10th 14, 09:25 AM

Ron C > wrote:

> On 5/9/2014 1:02 PM, Adrian Tuddenham wrote:
> > Gary Eickmeier > wrote:
> >
> >> "Scott Dorsey" > wrote in message
> >> ...
> >>
> >>
> >>> It's because a parametric has whatever Q you want it to have, so you can
> >>> match whatever it is that you're trying to deal with whether it is a
> >>> narrowband
> >>> problem or a wideband one.
> > [...]
> >> I suppose I should look up a few of the parametric kind and see just what
> >> they can do,
> >
> > Unless they have adjustable centre frequencies and Qs, they can never
> > equalise resonance-related problems correctly. Attempting to equalise a
> > resonance with a peak or notch that is even slightly off-tune will
> > produce horrible phasey-sounding effects on wideband sounds.
> >
> > I have spent countless hours trying to equalise badly-recorded 78s with
> > parametric equalisers and I know just how difficult it is. That was
> > with no more than three resonances and no delay comb effects. The
> > chances of doing that properly with a conventional graphic effects unit
> > are nil.
> >
> >
> Adrian,
> Have you tried time reversal on those 78s? One can't
> un-ring a bell in real time but can get a lot closer in
> reverse time. Try reversing the track and applying
> the correction.

I haven't tried reversal, but the effect of a pure resonance (without
time-delay components) is the same on the attack and the decay of a
sound, so it shouldn't make any difference. There was once a theory
that reversing a wax cylinder during playback would "improve the way the
stylus impinged on the grooves" and overcome distortion due to wear, but
this is a load of rubbish. The only detectable differences came from
slew-rate problems in the original reproducer, which altered (but did
not improve) with reverse playback. A good-quality playback sounded the
same in both directions.

The frequency distortions I have been trying to counteract are those
caused by the discontinuity at the mouth of a conical recording horn and
various other effects such as multiple reflections inside the horn and
resonances in the recording diaphragm. Many recordings were made with
multiple horns, which cannot be corrected, but some single-horn
recordings can be successfully equalised for mouth effects.

Once the mouth effect has been equalised, the other effects can be
heard, but they are usually quite trivial compared with the acoustics of
the (usually dreadful) recording room, so there is no point in
attempting any further equalisation.

>... Also note that some DSP EQs don't
> have the same Q for boost and cut.

I don't get that problem because I use analogue filters; that way I can
swing them through the signal in real-time and use the phasing effects
as audible clues to the correct settings. Big black knobs on calibrated
scale plates are the way forward!

> Getting back to the topic at hand, the EQ in a room can only
> be applied in real time, so the time smearing will be additive
> even if the average power is flattened. Put another way: you
> can't suck the energy out of a ringing cavity, you can only
> reduce the excitation power.

I think most attempts at equalising a three-dimensional room with a one
or two-dimensional system are doomed to failure.

I have only once partially succeeded in dealing with a problem like
this, when I had to recover an historic mono recording of an artist
performing a live concert in a hall with the P.A. system on the verge of
feedback. I ran it through a parametric filter which was accurately
adjusted to get rid of only the one most prominent peak. Once that had
been removed, I did a second pass to get rid of the next most obvious
peak. After several passes like that, the recording sounded half-decent
- but it was never going to sound particularly good.

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

Don Pearce[_3_]

May 10th 14, 09:34 AM

On Sat, 10 May 2014 09:25:43 +0100,
(Adrian Tuddenham) wrote:

>Ron C > wrote:
>
>> On 5/9/2014 1:02 PM, Adrian Tuddenham wrote:
>> > Gary Eickmeier > wrote:
>> >
>> >> "Scott Dorsey" > wrote in message
>> >> ...
>> >>
>> >>
>> >>> It's because a parametric has whatever Q you want it to have, so you can
>> >>> match whatever it is that you're trying to deal with whether it is a
>> >>> narrowband
>> >>> problem or a wideband one.
>> > [...]
>> >> I suppose I should look up a few of the parametric kind and see just what
>> >> they can do,
>> >
>> > Unless they have adjustable centre frequencies and Qs, they can never
>> > equalise resonance-related problems correctly. Attempting to equalise a
>> > resonance with a peak or notch that is even slightly off-tune will
>> > produce horrible phasey-sounding effects on wideband sounds.
>> >
>> > I have spent countless hours trying to equalise badly-recorded 78s with
>> > parametric equalisers and I know just how difficult it is. That was
>> > with no more than three resonances and no delay comb effects. The
>> > chances of doing that properly with a conventional graphic effects unit
>> > are nil.
>> >
>> >
>> Adrian,
>> Have you tried time reversal on those 78s? One can't
>> un-ring a bell in real time but can get a lot closer in
>> reverse time. Try reversing the track and applying
>> the correction.
>
>I haven't tried reversal, but the effect of a pure resonance (without
>time-delay components) is the same on the attack and the decay of a
>sound, so it shouldn't make any difference. There was once a theory
>that reversing a wax cylinder during playback would "improve the way the
>stylus impinged on the grooves" and overcome distortion due to wear, but
>this is a load of rubbish. The only detectable differences came from
>slew-rate problems in the original reproducer, which altered (but did
>not improve) with reverse playback. A good-quality playback sounded the
>same in both directions.
>
>The frequency distortions I have been trying to counteract are those
>caused by the discontinuity at the mouth of a conical recording horn and
>various other effects such as multiple reflections inside the horn and
>resonances in the recording diaphragm. Many recordings were made with
>multiple horns, which cannot be corrected, but some single-horn
>recordings can be successfully equalised for mouth effects.
>
>Once the mouth effect has been equalised, the other effects can be
>heard, but they are usually quite trivial compared with the acoustics of
>the (usually dreadful) recording room, so there is no point in
>attempting any further equalisation.
>
>
>
>>... Also note that some DSP EQs don't
>> have the same Q for boost and cut.
>
>I don't get that problem because I use analogue filters; that way I can
>swing them through the signal in real-time and use the phasing effects
>as audible clues to the correct settings. Big black knobs on calibrated
>scale plates are the way forward!
>
>> Getting back to the topic at hand, the EQ in a room can only
>> be applied in real time, so the time smearing will be additive
>> even if the average power is flattened. Put another way: you
>> can't suck the energy out of a ringing cavity, you can only
>> reduce the excitation power.
>
>I think most attempts at equalising a three-dimensional room with a one
>or two-dimensional system are doomed to failure.
>
>I have only once partially succeeded in dealing with a problem like
>this, when I had to recover an historic mono recording of an artist
>performing a live concert in a hall with the P.A. system on the verge of
>feedback. I ran it through a parametric filter which was accurately
>adjusted to get rid of only the one most prominent peak. Once that had
>been removed, I did a second pass to get rid of the next most obvious
>peak. After several passes like that, the recording sounded half-decent
>- but it was never going to sound particularly good.

In some recordings, if you are very fortunate, you may be able to
isolate a drum beat or other transient sound. You can then use that to
make a fair approximation to the impulse response of the room. That
can then be reverse-convolved to remove the room response from the
remainder of the recording. In practice, I usually find I prefer the
sound of the original, complete with its shortcomings. The
"improvement" artifacts are usually more objectionable.

d

Adrian Tuddenham[_2_]

May 10th 14, 10:00 AM

Don Pearce > wrote:

> On Sat, 10 May 2014 09:25:43 +0100,
> (Adrian Tuddenham) wrote:
>
> >Ron C > wrote:
> >
> >> On 5/9/2014 1:02 PM, Adrian Tuddenham wrote:
> >> > Gary Eickmeier > wrote:
> >> >
> >> >> "Scott Dorsey" > wrote in message
> >> >> ...
> >> >>
> >> >> >> It's because a parametric has whatever Q you want it to have, so
> >> >you can >> match whatever it is that you're trying to deal with
> >> >whether it is a >> narrowband >> problem or a wideband one. [...] > I
> >> >suppose I should look up a few of the parametric kind and see just
> >> >what > they can do,
> >> >
> >> > Unless they have adjustable centre frequencies and Qs, they can never
> >> > equalise resonance-related problems correctly. Attempting to equalise a
> >> > resonance with a peak or notch that is even slightly off-tune will
> >> > produce horrible phasey-sounding effects on wideband sounds.
> >> >
> >> > I have spent countless hours trying to equalise badly-recorded 78s with
> >> > parametric equalisers and I know just how difficult it is. That was
> >> > with no more than three resonances and no delay comb effects. The
> >> > chances of doing that properly with a conventional graphic effects unit
> >> > are nil.
> >> >
> >> >
> >> Adrian,
> >> Have you tried time reversal on those 78s? One can't
> >> un-ring a bell in real time but can get a lot closer in
> >> reverse time. Try reversing the track and applying
> >> the correction.
> >
> >I haven't tried reversal, but the effect of a pure resonance (without
> >time-delay components) is the same on the attack and the decay of a
> >sound, so it shouldn't make any difference. There was once a theory
> >that reversing a wax cylinder during playback would "improve the way the
> >stylus impinged on the grooves" and overcome distortion due to wear, but
> >this is a load of rubbish. The only detectable differences came from
> >slew-rate problems in the original reproducer, which altered (but did
> >not improve) with reverse playback. A good-quality playback sounded the
> >same in both directions.
> >
> >The frequency distortions I have been trying to counteract are those
> >caused by the discontinuity at the mouth of a conical recording horn and
> >various other effects such as multiple reflections inside the horn and
> >resonances in the recording diaphragm. Many recordings were made with
> >multiple horns, which cannot be corrected, but some single-horn
> >recordings can be successfully equalised for mouth effects.
> >
> >Once the mouth effect has been equalised, the other effects can be
> >heard, but they are usually quite trivial compared with the acoustics of
> >the (usually dreadful) recording room, so there is no point in
> >attempting any further equalisation.
> >
> >
> >
> >>... Also note that some DSP EQs don't
> >> have the same Q for boost and cut.
> >
> >I don't get that problem because I use analogue filters; that way I can
> >swing them through the signal in real-time and use the phasing effects
> >as audible clues to the correct settings. Big black knobs on calibrated
> >scale plates are the way forward!
> >
> >> Getting back to the topic at hand, the EQ in a room can only
> >> be applied in real time, so the time smearing will be additive
> >> even if the average power is flattened. Put another way: you
> >> can't suck the energy out of a ringing cavity, you can only
> >> reduce the excitation power.
> >
> >I think most attempts at equalising a three-dimensional room with a one
> >or two-dimensional system are doomed to failure.
> >
> >I have only once partially succeeded in dealing with a problem like
> >this, when I had to recover an historic mono recording of an artist
> >performing a live concert in a hall with the P.A. system on the verge of
> >feedback. I ran it through a parametric filter which was accurately
> >adjusted to get rid of only the one most prominent peak. Once that had
> >been removed, I did a second pass to get rid of the next most obvious
> >peak. After several passes like that, the recording sounded half-decent
> >- but it was never going to sound particularly good.
>
> In some recordings, if you are very fortunate, you may be able to
> isolate a drum beat or other transient sound. You can then use that to
> make a fair approximation to the impulse response of the room. That
> can then be reverse-convolved to remove the room response from the
> remainder of the recording. In practice, I usually find I prefer the
> sound of the original, complete with its shortcomings. The
> "improvement" artifacts are usually more objectionable.

This was a singer with guitar, recorded in Birmingham (U.K.) Town Hall
in the 1960s. It was on a nitrate disc that had been discovered in an
attic. In this particular case, the hall was part of the wanted sound -
but the peaky feedback was not (despite being fairly typical of the
era). As you say, any further 'improvement' would have made it sound
worse.

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

Scott Dorsey

May 10th 14, 02:02 PM

Gary Eickmeier > wrote:
>
>OK, I can believe you about the bass traps, but what would cause an 8k hump?

Hard reflecting surfaces. Move the microphone around and see how much the
response changes as you change position. You may find it changes dramatically
if you move a few inches, you may find it it doesn't change at all.

You may also find it's a speaker problem, but there is a good chance that it
is a room problem.

And it may not be one hump either, it might be lots of little humps on top
of one another.

>I realize that I am not making super careful measurements yet, but I still
>see no reason for this.

This is pretty typical of untreated living rooms, and it's the reason why
we have controlled playback rooms in studios.
--scott

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

Scott Dorsey

May 10th 14, 02:06 PM

Don Pearce > wrote:
>In some recordings, if you are very fortunate, you may be able to
>isolate a drum beat or other transient sound. You can then use that to
>make a fair approximation to the impulse response of the room. That
>can then be reverse-convolved to remove the room response from the
>remainder of the recording. In practice, I usually find I prefer the
>sound of the original, complete with its shortcomings. The
>"improvement" artifacts are usually more objectionable.

Folks have attempted to do this with acoustical 78s for years and years.
Probably the first example was Stockham's attempt to identify the impulse
response of the system used to record some Caruso recordings back in the
day when DSP processing involved week-long jobs on a minicomputer.

I think it is possible for a lot of recordings. Unfortunately there were
quite a few recordings made with multiple horns, for example one horn on
a singer and one on a piano, connected together with rubber tubing and
a manifold. The resonances on the two horns are different and can never
be separated out so it becomes a matter of compromise.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Adrian Tuddenham[_2_]

May 10th 14, 02:38 PM

Scott Dorsey > wrote:

> Don Pearce > wrote:
> >In some recordings, if you are very fortunate, you may be able to
> >isolate a drum beat or other transient sound. You can then use that to
> >make a fair approximation to the impulse response of the room. That
> >can then be reverse-convolved to remove the room response from the
> >remainder of the recording. In practice, I usually find I prefer the
> >sound of the original, complete with its shortcomings. The
> >"improvement" artifacts are usually more objectionable.
>
> Folks have attempted to do this with acoustical 78s for years and years.
> Probably the first example was Stockham's attempt to identify the impulse
> response of the system used to record some Caruso recordings back in the
> day when DSP processing involved week-long jobs on a minicomputer.

I'm doing it with analogue circuits in real time. Obviously, analogue
is the way of the future.

> I think it is possible for a lot of recordings. Unfortunately there were
> quite a few recordings made with multiple horns, for example one horn on
> a singer and one on a piano, connected together with rubber tubing and
> a manifold. The resonances on the two horns are different and can never
> be separated out so it becomes a matter of compromise.

It seems as though most 'ensemble' recordings were multi-horn jobs,
including solos if they had orchestral accompaniment. The exception was
a vocal solo with piano accompaniment, which would often have been taken
on one horn. There were far more 'talking' records in the catalogue
than nowadays, so they are a useful source of single-horn test material
for anyone experimenting with an equalisation system.

It seems that 'mixers' weren't unknown, either. I have found one
recording with two completley different frequency responses, which
change over when the singer stops and the orchestra plays a short
section. It sounds like a gas tap is being used to shut off one of the
horns.

I have found some recordings pre-1925 by U.K. Columbia which do not
respond to the usual horn equalisation. In some cases it is because
they used a flared horn, not a conical one; but in other cases the most
likely explanation is that they are experimental electric recordings
using the Holman system.

--
~ Adrian Tuddenham ~
(Remove the ".invalid"s and add ".co.uk" to reply)
www.poppyrecords.co.uk

Frank Stearns

May 10th 14, 05:16 PM

"Gary Eickmeier" > writes:

>OK, I can believe you about the bass traps, but what would cause an 8k hump?
>I realize that I am not making super careful measurements yet, but I still
>see no reason for this.

Are you using a calibrated test microphone? Much above 500 hz do NOT trust the
Rat Shacks. The mid-expensive mics aren't much better, but they often do give you a
good calibration curve that you can input into your test system.

Tried taking multiple measurements with the mic moved slightly each time? Remember,
the wavelen at 8k is about 1 3/4"

Any hard, reflective surfaces nearby? Might be getting some HF swarm building up, or
an unfortunate strong phase-add at that hz based on some even multiple of the
wavelen and distance between direct and bounce. It's going to vary because of delay
between direct and bounce, but over time (say 2x the time of the bounce path to your
measurement mic) things very well could be adding enough phase-add energy often
enough to cause your bump. Because it's 8K, nothing fancy needed to test this idea.
Temporarly hang some heavy blankets or comforters on the side walls. I realize
you're screaming at this point, but do you want to find the cause of the bump, or
not?

Anything in the room resonating at that hz? Look for small glass, metal, or hardwood
things, knick knacks, whatever, 1-2" in size.

Frank
Mobile Audio

--

William Sommerwerck

May 10th 14, 05:44 PM

"Scott Dorsey" wrote in message ...

> Probably the first example was Stockham's attempt to identify
> the impulse response of the system used to record some Caruso
> recordings back in the day when DSP processing involved week-
> long jobs on a minicomputer.

This is theoretically possible. (I was advocating such things 40 years ago.
It's unfortunate it hasn't been done with classic tape recordings.)

Was was /actually/ done was a spectral analysis of modern recordings, which
were compared with the spectral analyses of Caruso's recordings. This made it
possible to apply more-or-less rational corrective EQ.

At the time these recordings were released, people who'd heard Caruso live
swore that the EQ'd version sounded more like the live Caruso. I'm not so
sure. I was accustomed to hearing Caruso's voice unprocessed, and the EQ'd
version didn't sound as pleasing.

Frank Stearns

May 10th 14, 11:36 PM

"Gary Eickmeier" > writes:

>"Frank Stearns" > wrote in message
acquisition...
>> "Gary Eickmeier" > writes:
>>
>>>OK, I can believe you about the bass traps, but what would cause an 8k
>>>hump?
>>>I realize that I am not making super careful measurements yet, but I still
>>>see no reason for this.
>>
>> Are you using a calibrated test microphone? Much above 500 hz do NOT trust
>> the
>> Rat Shacks. The mid-expensive mics aren't much better, but they often do
>> give you a
>> good calibration curve that you can input into your test system.
>>
>> Tried taking multiple measurements with the mic moved slightly each time?
>> Remember,
>> the wavelen at 8k is about 1 3/4"
>>
>> Any hard, reflective surfaces nearby? Might be getting some HF swarm
>> building up, or
>> an unfortunate strong phase-add at that hz based on some even multiple of
>> the
>> wavelen and distance between direct and bounce. It's going to vary because
>> of delay
>> between direct and bounce, but over time (say 2x the time of the bounce
>> path to your
>> measurement mic) things very well could be adding enough phase-add energy
>> often
>> enough to cause your bump. Because it's 8K, nothing fancy needed to test
>> this idea.
>> Temporarly hang some heavy blankets or comforters on the side walls. I
>> realize
>> you're screaming at this point, but do you want to find the cause of the
>> bump, or
>> not?
>>
>> Anything in the room resonating at that hz? Look for small glass, metal,
>> or hardwood
>> things, knick knacks, whatever, 1-2" in size.
>>
>> Frank
>> Mobile Audio

>Oh believe me I am not rejecting your suggestions. I have some hard
>reflective surfaces around the speakers and a glass coffee table right in
>front of the listening sofa. I would be very interested in experimenting
>with removing some of that, in both the measurements and the listening.
>Getting the equalizer Monday. Will try the room treatments first.

Well, do be careful in assumptions. I put these things more or less in order of
strongest probability. That is, above 500 hz and before anything else, I'd suspect
the test microphone (unless it was a higher-end brand and a calibration curve had
been applied). And, I'd want additional tests at slightly different positions.

Frank
Mobile Audio

--

Frank Stearns

May 11th 14, 01:05 AM

Frank Stearns > writes:

>"Gary Eickmeier" > writes:

snips, and followup

>Well, do be careful in assumptions. I put these things more or less in order of
>strongest probability. That is, above 500 hz and before anything else, I'd suspect
>the test microphone (unless it was a higher-end brand and a calibration curve had
>been applied).

Forgot to mention: Remember, 8KHz is in that range of your typical condensor mic
resonance bump. Even the little electrets used in a lot of test mics have that, but
it's either flattened in the mic electronics or addressed in the calibration curve.

Frank

--

Sean Conolly

May 11th 14, 05:42 AM

"Gary Eickmeier" > wrote in message
...
>
> I predict it will be very useful to have the ability to shape the response
> of the things with a 31 band. I will be using the Velodyne with those new
> speakers too, and this Behringer FBQ6200 equalizer has a subwoofer output
> and adjustable crossover freq that is almost designed for my situation.

I will predict with great confidence that you would get a much better EQ
from a Behringer DEQ2496. It has 31 bands that can eith respond like a
traditional 31 band (with little peaks or dips between faders) or give you a
'smoothed' response simply following the curve you set.

But the best part is the dual five band parametric EQ. Even if your goal is
to not use any more EQ than you need it's still nice for 'what if'
scenarios: e.g. what if you notch out that 8K bump or fill bring up that dip
at 600 Hz- how will it sound at different point in the room?

Yes it has an RTA screen you can measure the response of pink noise, and let
it auto-set the graphic, but I like watching the RTA screen while moving the
mic or speakers around.

For what you're trying to do with your speakers and room, I highly recommend
it. You would find a lot of uses for it.

Sean

Sean Conolly

May 11th 14, 06:50 AM

"Frank Stearns" > wrote in message
acquisition...
> Frank Stearns > writes:
>
>>"Gary Eickmeier" > writes:
>
> snips, and followup
>
>>Well, do be careful in assumptions. I put these things more or less in
>>order of
>>strongest probability. That is, above 500 hz and before anything else, I'd
>>suspect
>>the test microphone (unless it was a higher-end brand and a calibration
>>curve had
>>been applied).
>
> Forgot to mention: Remember, 8KHz is in that range of your typical
> condensor mic
> resonance bump. Even the little electrets used in a lot of test mics have
> that, but
> it's either flattened in the mic electronics or addressed in the
> calibration curve.

See, I always thought that it wasn't that hard to get to get an acceptably
flat response from a very small diaphagm omni just from the physics of how
it works. Obviously it can be hurt by diffraction around the edge of the
element or in the preamp electronics.

Acceptable to me is say +- 1 dB for muso RTA purposes - not for critical
measurements of say mics or speakers - which would need an anechoic chamber
anyway...

Sean

Gary Eickmeier

May 11th 14, 07:20 AM

"Sean Conolly" > wrote in message
...
> "Gary Eickmeier" > wrote in message
> ...
>>
>> I predict it will be very useful to have the ability to shape the
>> response of the things with a 31 band. I will be using the Velodyne with
>> those new speakers too, and this Behringer FBQ6200 equalizer has a
>> subwoofer output and adjustable crossover freq that is almost designed
>> for my situation.
>
> I will predict with great confidence that you would get a much better EQ
> from a Behringer DEQ2496. It has 31 bands that can eith respond like a
> traditional 31 band (with little peaks or dips between faders) or give you
> a 'smoothed' response simply following the curve you set.
>
> But the best part is the dual five band parametric EQ. Even if your goal
> is to not use any more EQ than you need it's still nice for 'what if'
> scenarios: e.g. what if you notch out that 8K bump or fill bring up that
> dip at 600 Hz- how will it sound at different point in the room?
>
> Yes it has an RTA screen you can measure the response of pink noise, and
> let it auto-set the graphic, but I like watching the RTA screen while
> moving the mic or speakers around.
>
> For what you're trying to do with your speakers and room, I highly
> recommend it. You would find a lot of uses for it.
>
> Sean

Well, you highlight my quandary. I want to high pass my main speakers with
the subwoofer crossover, either from the receiver or from the Behringer
because that relieves the main speakers from trying to do the lowest freqs
and gives them more power in their range. But how do I do that with the
2496? Maybe Frank is right though and it is more a microphone problem with
the RS SLM.

Frank - I tried covering the coffee table with a quilt and doing the
measurement all over again, but got the same result. Trying now to use my
calibration microphone that came with an Audiocontrol equalizer/analyzer.
But the mike doesn't want to operate on its own without the analyzer that it
came with. I stuck it into my Zoom H6 recorder and nothing, either with or
withour power applied from teh recorder. So OK, maybe I have to stick it
into the analyzer and then take a line signal out of there and into the
recorder. Then I do a whole series all over again and take the recording
into the edit room and analyze with Adobe Audition.

Anyway, I think I am using the SLM correctly - it is set to fast, c
weighted - is that right? But you may be right, because my AT 2050 mikes
have a problem with a peak at about 10k as well.

Gary

Trevor

May 11th 14, 09:58 AM

"Mike Rivers" > wrote in message
...
> What I'm surprised that I can't find, given how common computers with
> sound cards are, is a modern computerized version of the clunky General
> Radio chain-driven synchronized oscillator and plotter that we had in our
> college lab in 1960. Connect the device you want to test between the audio
> output and input of a computer, use a program to generate a slow sine wave
> sweep, and generate a plot of what comes back into the computer's audio
> input. There are a number of FFT programs but it's just not the same
> thing.

Why not? Many programs can generate a sweep output on one channel and do a
FFT meaurement on the other, and can apply tracking filters if necessary.
Programs like SpectraLab were doing it a couple of decades ago.

Trevor.

Trevor

May 11th 14, 10:09 AM

"Gary Eickmeier" > wrote in message
...
> "hank alrich" > wrote in message
> ...
>
>> The Behringer DEQ2496 is a good learning tool, and a good tool period in
>> experienced hands. Two channels of multi-band, parametric, dynamic,
>> auto, etc., EQ in one box.
>
> Wow thanks Hank - this is serious. Their only critique was the reliability
> and support from the company. But it has an RTA! I already have a good
> calibration microphone from another unit that might work OK. There is no
> dedicated one for this unit anyway.

Not so, I have both the DEQ2496 and the Behringer measurement mic. What it
doesn't have is calibration data, not that any generic data you get with
similar cheap measurement mics is of much use anyway. Actual calibration
will always cost more than the mic unfortunately. But if you already have
one you think is a "good one" it should work just fine.

Trevor.

Trevor

May 11th 14, 10:15 AM

"Ron C" > wrote in message
...
> On 5/9/2014 3:35 PM, Gary Eickmeier wrote:
>> "hank alrich" > wrote in message
>> ...
>>[i]
>>> The Behringer DEQ2496 is a good learning tool, and a good tool period in
>>> experienced hands. Two channels of multi-band, parametric, dynamic,
>>> auto, etc., EQ in one box.
>>
>> Wow thanks Hank - this is serious. Their only critique was the
>> reliability
>> and support from the company. But it has an RTA! I already have a good
>> calibration microphone from another unit that might work OK. There is no
>> dedicated one for this unit anyway.
>>
>> Got to think long and hard on this one.
>>
>>
> This is an amazingly flexible unit for the price. I installed
> three of them in my old venue and I have two of my own.
> However, I did have three out of the box failures (DOA)
> and two that crashed after about a year of regular use.
> I traced the failures to overheating.

Yep, a small computer fan helps, or you can build an external analog power
supply if you want to go that route. (It is multi-voltage though)

Trevor.

Scott Dorsey

May 11th 14, 12:42 PM

Sean Conolly > wrote:
Frank writes:
>>
>> Forgot to mention: Remember, 8KHz is in that range of your typical
>> condensor mic
>> resonance bump. Even the little electrets used in a lot of test mics have
>> that, but
>> it's either flattened in the mic electronics or addressed in the
>> calibration curve.
>
>See, I always thought that it wasn't that hard to get to get an acceptably
>flat response from a very small diaphagm omni just from the physics of how
>it works. Obviously it can be hurt by diffraction around the edge of the
>element or in the preamp electronics.

It's not all that hard. As you point out, the two big issues are the
diffraction and the resonance (not of the diaphragm but of the resonator
placed in front of the diaphragm to extend the response).

The thing is... the market right now is awash in "measurement microphones"
that really aren't measurement microphones at all.

For room work, you don't need a real IEC Type I measurement mike, and you
can get away with one of the inexpensive electret mikes. But what makes the
inexpensive electret mikes useful is the calibration curve that comes with
it.

You pay around $200 for a cheap but usable measurement mike, and my great
suspicion is that about $25 of that is for the mike and $175 is for the
calibration curve.

>Acceptable to me is say +- 1 dB for muso RTA purposes - not for critical
>measurements of say mics or speakers - which would need an anechoic chamber
>anyway...

These days we can do speakers without an anechoic chamber using gated
methods. Technology like MLS has made field measurement a thousand times
easier than it was when all we had were swept sines. On the other hand,
these methods bring new sources of errors to deal with.
--scott

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

Scott Dorsey

May 11th 14, 12:44 PM

Gary Eickmeier > wrote:
>
>Well, you highlight my quandary. I want to high pass my main speakers with
>the subwoofer crossover, either from the receiver or from the Behringer
>because that relieves the main speakers from trying to do the lowest freqs
>and gives them more power in their range. But how do I do that with the
>2496? Maybe Frank is right though and it is more a microphone problem with
>the RS SLM.

My inclination would be just to do it with a second-order passive filter
rather than deal with active filtration. But it is probably easier to do
with an active filter for the first cut approximation.

You should have no problem doing that with any parametric that has a shelving
filter option.
--scott

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

Frank Stearns

May 11th 14, 02:46 PM

"Gary Eickmeier" > writes:

snips

>Frank - I tried covering the coffee table with a quilt and doing the
>measurement all over again, but got the same result. Trying now to use my

No, not the table in front of you (though that's a good place to start) but the
sides walls as well.

This topic is huge. You're putting your toe into a large ocean, where there's a lot
of stuff going on.

If you're serious about this, you've got to first get tools you can trust. Those
cost money, though if spend wisely and with knowledge going in you won't break the
bank. Then you have to know how to interpret what you measure as it's not as cut and
dried as we might hope. This takes experience, and careful thought and
understanding. Then it's also good to experiment and re-measure -- get that
first-hand experience.

I'm all for active cross-overs for subs. I wasn't at first, and it took a lot of
work to modify an active xover so that it was completely transparent, but totally
worth the effort in the long run. There is so much more control, as a better active
crossover will give you some extra controls to deal with what happens at the
crossover freq.

Frank
Mobile Audio

--

Peter Larsen[_3_]

May 11th 14, 05:08 PM

Frank Stearns wrote:

> I'm all for active cross-overs for subs. I wasn't at first, and it
> took a lot of work to modify an active xover so that it was
> completely transparent

I'm quite happy with the Ashly I use with my videoputer.

> Frank

Kind regards

Peter Larsen

None

May 12th 14, 02:01 AM

"Gary Eickmeier" > wrote in message
...
> <snip extended diversion into denial>
> I feel in a hurry to have something to demonstrate and tell the
> world. Then they can take it from there and refine all of the things
> you are talking about.

That last sentence is one of the most hilarious things you've written
here. As a classic text-book example of a "crank", you're usually not
quite so entertaining.

> Thanks for your interest and don't stop reading or give up on me. I
> need all of you guys.

As people for you to ignore?

PStamler

May 12th 14, 02:45 AM

On Sunday, May 11, 2014 5:48:54 PM UTC-6, Gary Eickmeier wrote:

> I basically want to check and make the frequency response "correct" as part
> of the theory and I want to know what that very simple aspect of the problem
> is doing when it sounds the best by ear. I want to eliminate that aspect of
> reproduction as one of the main differences among speakers so that I can
> study what I am after, the spatial characteristics.

Speaker designers, engineers and perceptual psychologist have been working on that "very simple aspect" since the 1920s, with only marginal results. Good luck trying to find a quick and easy answer.

Peace,
Paul

Frank Stearns

May 12th 14, 12:31 PM

"Peter Larsen" > writes:

>Frank Stearns wrote:

>> I'm all for active cross-overs for subs. I wasn't at first, and it
>> took a lot of work to modify an active xover so that it was
>> completely transparent

>I'm quite happy with the Ashly I use with my videoputer.

Ashly stuff is interesting. Often good, basic designs, but sometimes suffer from
cheap parts.

I took the XR1001 crossover (is that what you have?) and did these mods to get the
sound:

1. Beefed up PS; hexfred dioides, HF decoupling across newer, somewhat larger filter
caps.

2. Added decoupling at power pins of each IC. Tied to new ground buss.

3. Upgraded ICs

4. Removed redundant coupling caps, upgraded and HF bypassed others.

5. Removed all but one level pot (LF); changed other gain points to unity with 0.1%
resistors.

6. Biggest help was converting the contour, LF gain, and xover frequency select from
crappy cheap pots to Elma rotary switches loaded up with precision resistors.

After all that, the thing has no sonic fingerprint, whatsover, along with very
stable controls. (Those cheap pots had a bad habit of drifting, which caused no ends
of problems.) And, ultimately, with proper crossing, the whole system just sounds so
much better.

Credits to folks I met here years ago who gave me those mod suggestions -- Monte
McGuire, Stephen Sank, Jim Williams. Others here had some on-off suggestions that
were most useful, Scott Dorsey and others who I'm forgetting; apolgies.

Frank
Mobile Audio
--

Scott Dorsey

May 12th 14, 02:55 PM

On Sunday, May 11, 2014 5:48:54 PM UTC-6, Gary Eickmeier wrote:
>
> I basically want to check and make the frequency response "correct" as part
> of the theory and I want to know what that very simple aspect of the problem
> is doing when it sounds the best by ear. I want to eliminate that aspect of
> reproduction as one of the main differences among speakers so that I can
> study what I am after, the spatial characteristics.

Unfortunately, getting the frequency response correct is the hard part.

Don't forget that it ALSO implies getting the dispersion correct, since
frequency response and pattern are intimately tied together.

You are correct that if the response is not flat, you can't accurately
determine the imaging accuracy. The problem is that the response is never
really flat.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[email protected]

May 12th 14, 04:44 PM

>
> > I basically want to check and make the frequency response "correct" as
>
> > part of the theory and I want to know what that very simple aspect of the
>
> > problem is doing when it sounds the best by ear. I want to eliminate that
>
> > aspect of reproduction as one of the main differences among speakers so
>
> > that I can study what I am after, the spatial characteristics.
>
> >

in that case, here are two options for you

1) use a fixed set of speakers, don't change them, just move them around...maybe buy a variable delay box..

2) concentrate on the mid-range... create source material that is only mid-range and use test your various speakers to match in the mid range, it will be almost impossible to match speakers at the low and high end.

If you can figure out how the spatial effects work in the mid range then you can try to expand to the top and bottom ends.

And for those looking for low cost waterfalls.. try this
http://www.qsl.net/dl4yhf/spectra1.html

Also N track studio has a very good user interface with of an RTA and "draw the response you want EQ".

have fun

Mark

Peter Larsen[_3_]

May 12th 14, 05:18 PM

Frank Stearns wrote:

> "Peter Larsen" > writes:

>> Frank Stearns wrote:

>>> I'm all for active cross-overs for subs. I wasn't at first, and it
>>> took a lot of work to modify an active xover so that it was
>>> completely transparent

>> I'm quite happy with the Ashly I use with my videoputer.

> Ashly stuff is interesting. Often good, basic designs, but sometimes
> suffer from cheap parts.

Never opened it, bought it with a rack case and something else from a pa
company that had discontinued their disco-rental. Taking the bundle was the
best deal.

> I took the XR1001 crossover (is that what you have?)

Yes. It is not obnoxious but also not totally transparent, it definitely has
a signature. It sits in front of a NAD 906 6 x 25 watts amp powering a pair
of small KEF "rear loudspeakers", the old version with silk dome and a pair
of ATC 9" Studio in factory recommended boxes.

> and did these
> mods to get the sound:

That's one for the archive, thanks, imo it has way too many front panel
controls, but tweaking the cross-over Q did nice things in terms of making
the cornerplaced face2face subs vanish and having extended downwards range
for the videoputers audio is one of my better ideas.

I'm not gonna rush into modding it, but it is an interesting idea. Thank you
very much Frank.

> Frank
> Mobile Audio

Kind regards

Peter Larsen

Scott Dorsey

May 12th 14, 08:22 PM

Peter Larsen > wrote:
>Scott Dorsey wrote:
>
>> Peter Larsen > wrote:
>
>>> Two important points: the treble may not appear to be detached and a
>>> subwoofer, if available, may not be detectable on male vox.
>
>> Detached?
>
>Yes, apparently elevated after a dip, something like:
>
>[perception]
>
>xxxxxxxxxxxxxxxxxxxxxxxxxx xxxxxxxx
> xxxxxx x
> xxxxx
>

Are you talking about ringing (meaning that was a time domain plot with
overshoot) or are you talking about trying to equalize a dip with a filter
that wasn't the same width as the dip (meaning that was a frequency domain one)?
--scott

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

[email protected]

May 13th 14, 04:54 AM

I'm an fan of equalizing the speaker only and letting the room do it's thing. A single reflection causes a comb effect and trying to cancel the ripples with an EQ just makes the impulse response into a horrible mess.
I was the designer of the dbx 2020 back around 1980 that was the first (I think ...) consumer RTA/equalizer combo that would auto- EQ a room based on pink noise and an octave EQ. It often didn't sound so great when it was done. But in a consumer environment it usually added more bass which as we all know is what 9 out of 10 consumers prefer :)

Bob

William Sommerwerck

May 13th 14, 02:13 PM

wrote in message ...

> I'm a fan of equalizing the speaker only and letting the room do its thing.
> A single reflection causes a comb effect and trying to cancel the ripples
> with an EQ just makes the impulse response into a horrible mess.
> I was the designer of the dbx 2020 back around 1980 that was the first
> (I think ...) consumer RTA/equalizer combo that would auto- EQ a room
> based on pink noise and an octave EQ. It often didn't sound so great
> when it was done. But in a consumer environment it usually added more
> bass which as we all know is what 9 out of 10 consumers prefer :)

I did about a half-dozen installations using the Crown EQ-2. If the room had
reasonably good acoustics to begin with, the equalization /always/ improved
the sound.

By the way, I used an analyzer whose test signal was a mixture of
swept-frequency square waves.

There are new approaches to EQ that (supposedly) take into account time
factors. Guess I’m going to have to learn about them.

Scott Dorsey

May 13th 14, 02:24 PM

William Sommerwerck > wrote:
>wrote in message ...
>
>> I'm a fan of equalizing the speaker only and letting the room do its thing.
>> A single reflection causes a comb effect and trying to cancel the ripples
>> with an EQ just makes the impulse response into a horrible mess.
>> I was the designer of the dbx 2020 back around 1980 that was the first
>> (I think ...) consumer RTA/equalizer combo that would auto- EQ a room
>> based on pink noise and an octave EQ. It often didn't sound so great
>> when it was done. But in a consumer environment it usually added more
>> bass which as we all know is what 9 out of 10 consumers prefer :)
>
>
>I did about a half-dozen installations using the Crown EQ-2. If the room had
>reasonably good acoustics to begin with, the equalization /always/ improved
>the sound.

And see, my experience going into studios was the first thing I always did was
disable those things, and disabiling them always improved the sound.

>By the way, I used an analyzer whose test signal was a mixture of
>swept-frequency square waves.

Was this an MLS thing or something else?

>There are new approaches to EQ that (supposedly) take into account time
>factors. Guess I’m going to have to learn about them.

Well, that's not really equalization....
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Gary Eickmeier

May 15th 14, 10:55 AM

> wrote in message
...

> in that case, here are two options for you
>
> 1) use a fixed set of speakers, don't change them, just move them
> around...maybe buy a variable delay box..
>
> 2) concentrate on the mid-range... create source material that is only
> mid-range and use test your various speakers to match in the mid range,
> it will be almost impossible to match speakers at the low and high end.
>
> If you can figure out how the spatial effects work in the mid range then
> you can try to expand to the top and bottom ends.
>
>
> And for those looking for low cost waterfalls.. try this
> http://www.qsl.net/dl4yhf/spectra1.html
>
> Also N track studio has a very good user interface with of an RTA and
> "draw the response you want EQ".
>
> have fun
>
> Mark

Sean and Mark -

Sorry I am late getting back to you - has been a very long week with little
time to do some of my own stuff. Those programs look interesting, might be
able to try them next week. Right now, I have just received the equalizer
(6200) and finally got it mounted in the rack but not wired in yet. Anyway,
the thing slid right into an available opening in the rack and will be very
easy to install, all the way down to the sub out with variable crossover
point.

The info I had before on the response problems was no correct. I finally got
the calibration mike working by sticking it into the Audiocontrol C101, and
the readings I got with that agree perfectly with those of a reviewer friend
from a couple years ago, so I trust those more. No peak at 8 to 10k But
there are still problems that I will now be able to correct, for these or
any other test speakers.

I still can't understand why the calibration microphone will not work in
anything but the C101. I inserted it into my Zoom H6 recorder and tried it
with Phantom power or without, with other settings of microphone power but
nothing worked. Anyone know what's up with that?

Thanks,
Gary Eickmeier

Sean Conolly

May 15th 14, 04:32 PM

"Gary Eickmeier" > wrote in message
...
>
> "hank alrich" > wrote in message
> ...
>
>> The Behringer DEQ2496 is a good learning tool, and a good tool period in
>> experienced hands. Two channels of multi-band, parametric, dynamic,
>> auto, etc., EQ in one box.
>
> Wow thanks Hank - this is serious. Their only critique was the reliability
> and support from the company. But it has an RTA! I already have a good
> calibration microphone from another unit that might work OK. There is no
> dedicated one for this unit anyway.
>
> Got to think long and hard on this one.

Like I said - you'll find many ways to use it - it's like a swiss army
knife. My only regret is that I don't have two, one for the live rack and
one for the home rack.

But they do suffer from overheating and shutting down on occasions, so try
to leave some space around it in the rack.

Sean

Gary Eickmeier

May 15th 14, 06:47 PM

"Sean Conolly" > wrote in message
...
> "Gary Eickmeier" > wrote in message
> ...
>>
>> "hank alrich" > wrote in message
>> ...
>>
>>> The Behringer DEQ2496 is a good learning tool, and a good tool period in
>>> experienced hands. Two channels of multi-band, parametric, dynamic,
>>> auto, etc., EQ in one box.
>>
>> Wow thanks Hank - this is serious. Their only critique was the
>> reliability and support from the company. But it has an RTA! I already
>> have a good calibration microphone from another unit that might work OK.
>> There is no dedicated one for this unit anyway.
>>
>> Got to think long and hard on this one.
>
> Like I said - you'll find many ways to use it - it's like a swiss army
> knife. My only regret is that I don't have two, one for the live rack and
> one for the home rack.
>
> But they do suffer from overheating and shutting down on occasions, so try
> to leave some space around it in the rack.
>
> Sean

Sean - and everyone -

One more serious question. This device is a digital I/O DAC with processing.
But I want to use it as a signal processor after the receiver and before the
power amp. This in turn means that I need it to be able to handle the
various gains of all program material I put into it.

But both this one and the 6200 have level meters that I am supposed to use
to set a fixed mid level, not too low and not too high, into and out of the
device. Something there is that doesn't make sense. I need to be able to set
the gain wherever I feel like it without worrying about the preamp circuits
being overloaded.

I just printed out the full manual, available online only, and hope that I
find some answer to all this. Maybe it is not intended as a Hi Fi component,
just as a recording console accessory.

What am I missing here?

Gary

PStamler

May 15th 14, 09:32 PM

On Thursday, May 15, 2014 11:47:40 AM UTC-6, Gary Eickmeier wrote:

> But I want to use it as a signal processor after the receiver and before the
> power amp. This in turn means that I need it to be able to handle the
> various gains of all program material I put into it.

Please get your terms straight; if you're feeding in program material at various levels, then "levels" is the right term, and "gain" isn't. A signal doesn't have gain; a circuit does.

>
> But both this one and the 6200 have level meters that I am supposed to use
> to set a fixed mid level, not too low and not too high, into and out of the
> device. Something there is that doesn't make sense. I need to be able to set
> the gain wherever I feel like it without worrying about the preamp circuits
> being overloaded.

There is no active audio device where you can set either the gain or the signal's level "wherever I feel like it". Every piece of equipment, other than passive networks, has a window within which signal levels must fall for correct operation. If the signal level is too low, the device will add unwanted noise. If the signal level is too high, the device will distort.

This is Audio 101 stuff. Go get yourself a copy of the Yamaha Sound Reinforcement Handbook and read up on gain staging.

Peace,
Paul

PS You also wrote:

>
>Maybe it is not intended as a Hi Fi component,

> just as a recording console accessory.

It's mostly intended for sound reinforcement systems, especially permanently-installed ones.

> What am I missing here?

Like I said, go read the Yamaha Handbook.

Sean Conolly

May 16th 14, 12:05 AM

"Gary Eickmeier" > wrote in message
...
>
> "Sean Conolly" > wrote in message
> ...
>> "Gary Eickmeier" > wrote in message
>> ...
>>>
>>> "hank alrich" > wrote in message
>>> ...
>>>
>>>> The Behringer DEQ2496 is a good learning tool, and a good tool period
>>>> in
>>>> experienced hands. Two channels of multi-band, parametric, dynamic,
>>>> auto, etc., EQ in one box.
>>>
>>> Wow thanks Hank - this is serious. Their only critique was the
>>> reliability and support from the company. But it has an RTA! I already
>>> have a good calibration microphone from another unit that might work OK.
>>> There is no dedicated one for this unit anyway.
>>>
>>> Got to think long and hard on this one.
>>
>> Like I said - you'll find many ways to use it - it's like a swiss army
>> knife. My only regret is that I don't have two, one for the live rack and
>> one for the home rack.
>>
>> But they do suffer from overheating and shutting down on occasions, so
>> try to leave some space around it in the rack.
>>
>> Sean
>
> Sean - and everyone -
>
> One more serious question. This device is a digital I/O DAC with
> processing. But I want to use it as a signal processor after the receiver
> and before the power amp. This in turn means that I need it to be able to
> handle the various gains of all program material I put into it.
>
> But both this one and the 6200 have level meters that I am supposed to use
> to set a fixed mid level, not too low and not too high, into and out of
> the device. Something there is that doesn't make sense. I need to be able
> to set the gain wherever I feel like it without worrying about the preamp
> circuits being overloaded.
>
> I just printed out the full manual, available online only, and hope that I
> find some answer to all this. Maybe it is not intended as a Hi Fi
> component, just as a recording console accessory.
>
> What am I missing here?

The input level is line level plus some headroom, and the EQs and limiter
each have a gain adjustment so you can adjust the final level quite a bit,
albeit with a bit of fiddling. There's probably more than I can remember
right now - all my stuff is line level so I just do the basic gain staging.

And it's not just for recording, for live use I run mine as the final stage
before a 2KW amp.

Sean

Gary Eickmeier

May 16th 14, 06:15 AM

"Sean Conolly" > wrote in message
...
> "Gary Eickmeier" > wrote in message
> ...
>>
>> "Sean Conolly" > wrote in message
>> ...
>>> "Gary Eickmeier" > wrote in message
>>> ...
>>>>
>>>> "hank alrich" > wrote in message
>>>> ...
>>>>
>>>>> The Behringer DEQ2496 is a good learning tool, and a good tool period
>>>>> in
>>>>> experienced hands. Two channels of multi-band, parametric, dynamic,
>>>>> auto, etc., EQ in one box.
>>>>
>>>> Wow thanks Hank - this is serious. Their only critique was the
>>>> reliability and support from the company. But it has an RTA! I already
>>>> have a good calibration microphone from another unit that might work
>>>> OK. There is no dedicated one for this unit anyway.
>>>>
>>>> Got to think long and hard on this one.
>>>
>>> Like I said - you'll find many ways to use it - it's like a swiss army
>>> knife. My only regret is that I don't have two, one for the live rack
>>> and one for the home rack.
>>>
>>> But they do suffer from overheating and shutting down on occasions, so
>>> try to leave some space around it in the rack.
>>>
>>> Sean
>>
>> Sean - and everyone -
>>
>> One more serious question. This device is a digital I/O DAC with
>> processing. But I want to use it as a signal processor after the receiver
>> and before the power amp. This in turn means that I need it to be able to
>> handle the various gains of all program material I put into it.
>>
>> But both this one and the 6200 have level meters that I am supposed to
>> use to set a fixed mid level, not too low and not too high, into and out
>> of the device. Something there is that doesn't make sense. I need to be
>> able to set the gain wherever I feel like it without worrying about the
>> preamp circuits being overloaded.
>>
>> I just printed out the full manual, available online only, and hope that
>> I find some answer to all this. Maybe it is not intended as a Hi Fi
>> component, just as a recording console accessory.
>>
>> What am I missing here?
>
> The input level is line level plus some headroom, and the EQs and limiter
> each have a gain adjustment so you can adjust the final level quite a bit,
> albeit with a bit of fiddling. There's probably more than I can remember
> right now - all my stuff is line level so I just do the basic gain
> staging.
>
> And it's not just for recording, for live use I run mine as the final
> stage before a 2KW amp.
>
> Sean

OK, so there must be enough headroom in it to drive a power amp to any level
I desire. So I don't have to put it in a tape loop and set it to a certain
level.

Gary

Gary Eickmeier

May 17th 14, 03:17 AM

PStamler wrote:

> There is no active audio device where you can set either the gain or
> the signal's level "wherever I feel like it". Every piece of
> equipment, other than passive networks, has a window within which
> signal levels must fall for correct operation. If the signal level is
> too low, the device will add unwanted noise. If the signal level is
> too high, the device will distort.
>
> This is Audio 101 stuff. Go get yourself a copy of the Yamaha Sound
> Reinforcement Handbook and read up on gain staging.

So I cannot use this equalizer in line after the receiver? But Behringer
says I can. Here is their reply:

"Hello Gary,

I would recommend leaving the Input level control of the EQ at the 12
o'clock position. With that set, your volume from the receiver will be you
normal volume control. From that point, the EQ will function precisely how
you are requesting.

--------------- Original Message ---------------
From: Gary Eickmeier ]
Sent: 5/16/2014 10:06 AM
To:
Subject: Re: Behringer (MUSIC Group Support Case 00200534) "FBQ6200" [
ref:_00D90YvhK._50090MGZs4:ref ]

Jeff -

No, it doesn't help. If I may put the equalizer in line to the power
amplifier, I need it to pass on any level I choose to listen at. But if the
unit requires a certain single signal level, then it cannot be used after
the receiver, because I will be changing the signal level all the time.

If it can't be used in this way, please let me know. If it has to be in some
signal processor loop, like a tape loop, let me know. The manual is not very
good on the uses of the unit."

Gary Eickmeier

So there must be enough headroom in the 6200 to handle
all of the settings I might need in the future. I will try it tomorrow,
after I take another reading of my current frequency response. Will let you
know just how it works.

Gary

PStamler

May 17th 14, 04:25 AM

On Friday, May 16, 2014 8:17:26 PM UTC-6, Gary Eickmeier wrote:

> So I cannot use this equalizer in line after the receiver? But Behringer
> says I can. Here is their reply:
>
>
>
> "Hello Gary,
>
>
>
> I would recommend leaving the Input level control of the EQ at the 12
> o'clock position. With that set, your volume from the receiver will be you
> normal volume control. From that point, the EQ will function precisely how
> you are requesting.

I didn't say you couldn't use it after the receiver; I said that you needed to set it in such a way that it wouldn't overload, and the tech from Behringer suggested a setting that would accomplish that, probably pretty close to unity gain.

You're using it with consumer equipment, which means, to oversimplify, that the limiting stage will be the power amp. If you're using the preamp and not clipping the power amp now, you probably won't clip the power amp with the EQ in the chain either, unless you use crazy boost settings on the EQ -- which is a bad idea anyway.

Peace,
Paul

Dave Plowman (News)

May 17th 14, 10:24 AM

In article >,
Gary Eickmeier > wrote:
> So I cannot use this equalizer in line after the receiver?

It might introduce more noise when used like that. Better to use it before
the pre-amp volume control so it receives a 'constant' level. If there is
an insert point or even a tape loop.

--
*War does not determine who is right - only who is left.

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

William Sommerwerck

May 17th 14, 01:44 PM

Gary, my understanding is that you've been doing your research for years. How
can you be ignorant of what equipment does, what its limitations are, and how
it's connected?

My late friend Bill Hamlin was an English major and an actor. He had no
technical training whatever, yet he could rip a hi-fi system apart and put it
back together without having to even glance at the instructions. I knew him
for 35 years, and never once had to explain any of this to him.

Gary Eickmeier

May 17th 14, 03:47 PM

"William Sommerwerck" > wrote in message
...
> Gary, my understanding is that you've been doing your research for years.
> How can you be ignorant of what equipment does, what its limitations are,
> and how it's connected?

William,

I suspect you don't see the problem, or potential problem. Of course I will
see what this particular piece does when I hook it up as I plan to, and as
the Behringer people say that it can be, but there is still an interesting
situation that I have not seen before. I have been doing audio systems for -
let's see - 55 years since my first separates system as a teenager. Built
the Heathkit amp, used preamp and other components. But my real education
came with my film work, doing multiple track sync sound on Super 8 mag film
with flatbed, three sync recorders, mixers - well, I don't need to explain
myself to you over an insult.

Here is the point in a nutshell. If it needs to be input at a constant
level, then it makes no sense to put it after a volume control. If it needs
to be used in a tape loop of some sort, then I would have no control over
the sub out level. This thing seems to be designed as part of a live sound
system for large rooms, DJs, etc, rather than a hi fi component as such.
They seem to think that the main sound that will be going into it will be
from microphones, hence all of their talk about feedback control at certain
frequencies.

My receiver has but one opportunity for a processor loop, which is sort of a
tape loop for CDRs or other little recorders. But I can't press that button
unless I want to UN-press my input select button, such as CD. When digital
came out, with the HD video and all, I had to go to a receiver rather than
my usual preamp/ power amp, which changed the whole game.

So the design intent and amount of headroom and correct usage and running
the signal up and down the range of acceptable input to the unit were my
main questions, which this group should certainly be familiar with! I
thought maybe this was something that most members used all the time, so
just checking because I am big on design intent and not using something
outside its range.

Got a big video job right now, will install the thing when I get back and
report back.

Gary

None

May 17th 14, 04:00 PM

"Gary Eickmeier" > wrote in message
...
> I have been doing audio systems for - let's see - 55 years since my
> first separates system as a teenager.

Wow. A lot of people learn the essentials after only a few years. Yet
you are, in many ways, a complete novice, without the ability to grasp
even the most elementary audio concepts. It wouldn't take long to
learn the basics, such as reading the Yamaha book that's been
recommended to you so many times. Why do you refuse to educate
yourself? Have you just given up on ever understanding the rudimentary
basics of audio? Have you just decided that you'll never want to take
the training wheels off?

hank alrich

May 17th 14, 07:45 PM

Gary Eickmeier > wrote:

> They seem to think that the main sound that will be going into it will be
> from microphones, hence all of their talk about feedback control at certain
> frequencies.

It can be used with the intended measurement mic for acoustical
analysis. It can be used on an auxiliary output feeding stage monitors,
or on the master outputs, matrix outputs, anywhere one might stick an
equalizer, because that's what it is.

As such it is most often deployed following some sort of gain control,
such as the controls for managing levels for those various functions.

It's only mic input is for the measurement mic, is not a balanced input,
and therefore, the DEQ2496 is not intended to receive typical microphone
levels at its XLR inputs.

It can be used very gently, with broad Q settings, and it can be used
extremely precisely with very tight Q settings, and it can do both of
those job simultaneously. It can be set to react only to signals above
specified levels, and much more.

There is a tremendous amount of information in the manual.

I don't know how in this world one gets from the amount of experience
you claim to have, to writing the sentence I have quoted above.
William's point is cogent: to what have you been paying attention while
supposedly gathering all that audio "experience"? Certainly some of the
simplest fundamentals have escaped you completely, while you claim to be
advancing the science of music reproduction.

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

hank alrich

May 17th 14, 07:45 PM

None > wrote:

> "Gary Eickmeier" > wrote in message
> ...
> > I have been doing audio systems for - let's see - 55 years since my
> > first separates system as a teenager.
>
> Wow. A lot of people learn the essentials after only a few years. Yet
> you are, in many ways, a complete novice, without the ability to grasp
> even the most elementary audio concepts. It wouldn't take long to
> learn the basics, such as reading the Yamaha book that's been
> recommended to you so many times. Why do you refuse to educate
> yourself? Have you just given up on ever understanding the rudimentary
> basics of audio? Have you just decided that you'll never want to take
> the training wheels off?
>

He's too busy designing the successor to the unicycle to bother with
unloading the training wheels.

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

Hank

May 17th 14, 09:34 PM

In article >,
Gary Eickmeier > wrote:
>PStamler wrote:
>
>> There is no active audio device where you can set either the gain or
>> the signal's level "wherever I feel like it". Every piece of
>> equipment, other than passive networks, has a window within which
>> signal levels must fall for correct operation. If the signal level is
>> too low, the device will add unwanted noise. If the signal level is
>> too high, the device will distort.
>>
>> This is Audio 101 stuff. Go get yourself a copy of the Yamaha Sound
>> Reinforcement Handbook and read up on gain staging.
>
>So I cannot use this equalizer in line after the receiver? But Behringer
>says I can. Here is their reply:
>
>"Hello Gary,
>
>I would recommend leaving the Input level control of the EQ at the 12
>o'clock position. With that set, your volume from the receiver will be you
>normal volume control. From that point, the EQ will function precisely how
>you are requesting.
>
Define "receiver" as you are using it. If this is an all-in-one
consumer box with both preamp and amplifier in the same unit, you'll
need to tap in between the preamp output and the output amplifier
input. Tape out->equalizer->tape monitor generally will accomplish
this.

Hank

Gary Eickmeier

May 18th 14, 07:31 AM

"Hank" > wrote in message
...

> Define "receiver" as you are using it. If this is an all-in-one
> consumer box with both preamp and amplifier in the same unit, you'll
> need to tap in between the preamp output and the output amplifier
> input. Tape out->equalizer->tape monitor generally will accomplish
> this.

Hank,

I don't know how you could have as much audio experience as you claim and be
so ignorant of a simple term such as "receiver."....

Just kidding, adding to the hilarity from Summerwerck and None.

Mine is a Pioneer of some years back vintage, but it has most of the
features of a modern receiver. As I said, I was forced to go the receiver
route when video connection and Dolby decoding for film tracks and HD video
pass through became necessary. Mine has amplifiers in it, but I don't use
most of them. I go pre out to some Carver M 1.5 amplifiers and to the main
speakers, and sub out to the Velodyne with its internal amplifier. However,
the power amps are still available inside the receiver, and I take advantage
of that in feeding my center channel from the center speaker feed out, and
two extra surround speakers from the surround amp connection of the
receiver.

This all works just fine, but the Behringer is slightly strange to me in
their description of how it is to be used. It has the subwoofer output but
that would make no sense in a tape loop connection. But even if I didn't
want the sub output on the Behringer, I am having trouble picturing using
the tape loop as an external processor because if I need to press that
button on the receiver, I can't also press CD or Video or FM Tuner. And no,
there is no mike input on the Behringer - they expect it to get the mike
feed from the mixing board.

Gary

None

May 18th 14, 01:49 PM

"Gary Eickmeier" > wrote in message
...
> I don't know how you could have as much audio experience as you
> claim and be so ignorant of a simple term such as "receiver."....
>
> am having trouble picturing using the tape loop as an external
> processor because if I need to press that button on the receiver, I
> can't also press CD or Video or FM Tuner.

So, despite your gibbering, you don't know how to work a receiver.
It's as if you're begging to be mocked for your ignorance, yet again.
You're having serious problems with rudimentary concepts like how to
connect things together, and elementary gain staging. Maybe your
bizarre concept that a signal has gain is part of the problem. As long
as you pretend that you know what you're doing, you'll be a
laughingstock.

John Williamson

May 18th 14, 02:07 PM

On 18/05/2014 13:49, None wrote:
> "Gary Eickmeier" > wrote in message
> ...
>> I don't know how you could have as much audio experience as you claim
>> and be so ignorant of a simple term such as "receiver."....
>>
>> am having trouble picturing using the tape loop as an external
>> processor because if I need to press that button on the receiver, I
>> can't also press CD or Video or FM Tuner.
>
> So, despite your gibbering, you don't know how to work a receiver. It's
> as if you're begging to be mocked for your ignorance, yet again. You're
> having serious problems with rudimentary concepts like how to connect
> things together, and elementary gain staging. Maybe your bizarre concept
> that a signal has gain is part of the problem. As long as you pretend
> that you know what you're doing, you'll be a laughingstock.
>
>
He's also confusing domestic gear with professional gear. A`professional
setup would have the connections available to do what he wants, but
would need setting up correctly, while the stuff he has doesn't, and
comes almost ready set up from the maker, while conforming to very few
standards.

I've never had any trouble listening to the output of the tape loop on
any system I've owned while recording any source. Possibly because they
were all bought or built from scratch with that in mind.

--
Tciao for Now!

John.

William Sommerwerck

May 18th 14, 02:36 PM

"Gary Eickmeier" wrote in message ...

> I am having trouble picturing using the tape loop as an
> external processor because if I need to press that button on the receiver, I
> can't also press CD or Video
> or FM Tuner.

Then it's not a "tape loop". There's no loop!

The Tape Monitor switch was intended for three-head machines. * Throwing the
switch let you hear either the source program, or the actual recording coming
back from the deck. That's what made them so convenient for signal-processing
equipment.

* Are you aware there are true three-head DATs, that actually monitor off the
tape?

Scott Dorsey

May 18th 14, 03:02 PM

John Williamson > wrote:
>
>I've never had any trouble listening to the output of the tape loop on
>any system I've owned while recording any source. Possibly because they
>were all bought or built from scratch with that in mind.

You want to put the device in-between the preamplifier and the amplifier.
If you do not have a break-in loop for that (and some receivers do),
then you have a problem.

The output of the tape loop is _before_ the preamp gain control, while
the original poster wants a point to break in after the gain control.

The tape loop is not where you want to be. If the receiver you have does
not allow you to pull a signal out before the power amp stages, you'll have
to do a little soldering and install connectors there.

Plenty of old receivers did that so that people in the process of upgrading
could use them with larger power amplifiers simply as a preamp, but the
current range of home theatre receivers is designed to be as cheap as possible
and may be omitting such handy features.

My suggestion to the original poster is to ditch the receiver and go with
seperates, and move the receiver to another system where you can watch movies.
The system you want for movie soundtracks is totally different than the system
you want for realistic stereo reproduction; the speaker and room arrangements
are not the same and the requirements for the speakers are different. Attempts
to use the same system for both film surround playback and stereo music playback
result only in poor compromises of both.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Gary Eickmeier

May 18th 14, 06:24 PM

"Scott Dorsey" > wrote in message
...
> John Williamson > wrote:
>>
>>I've never had any trouble listening to the output of the tape loop on
>>any system I've owned while recording any source. Possibly because they
>>were all bought or built from scratch with that in mind.
>
> You want to put the device in-between the preamplifier and the amplifier.
> If you do not have a break-in loop for that (and some receivers do),
> then you have a problem.
>
> The output of the tape loop is _before_ the preamp gain control, while
> the original poster wants a point to break in after the gain control.
>
> The tape loop is not where you want to be. If the receiver you have does
> not allow you to pull a signal out before the power amp stages, you'll
> have
> to do a little soldering and install connectors there.
>
> Plenty of old receivers did that so that people in the process of
> upgrading
> could use them with larger power amplifiers simply as a preamp, but the
> current range of home theatre receivers is designed to be as cheap as
> possible
> and may be omitting such handy features.
>
> My suggestion to the original poster is to ditch the receiver and go with
> seperates, and move the receiver to another system where you can watch
> movies.
> The system you want for movie soundtracks is totally different than the
> system
> you want for realistic stereo reproduction; the speaker and room
> arrangements
> are not the same and the requirements for the speakers are different.
> Attempts
> to use the same system for both film surround playback and stereo music
> playback
> result only in poor compromises of both.
> --scott
> --
> "C'est un Nagra. C'est suisse, et tres, tres precis."

Thanks for the first sensible answer Scott. I checked again, and my receiver
just does not have an external processor loop available. There is a recorder
connection with the usual Out and In jacks, but there is no way to select my
source, such as CD, and also the tape loop. I think what they expect us to
do with this design is that the selected signal is always available to the
Tape Out jacks to record from, and then after recording is done you can play
it back later by selecting the Tape source. But you CANNOT use it
simultaneously with the CD button as a Source/Tape monitor as in the old
days. I have my Carver preamp still available for LP sources, and it works a
little differently. There is a rotary source selector for source such as CD,
and also two tape loop switches that you can use to select Source or Tape.
That is what I am used to and hoping that I could do with the new equalizer,
but as is I'm afraid not.

So why did I select a receiver rather than a separate digital signal
peamp/processor? Because those things cost a fortune! At least when I was
shopping for them. The receiver has been very convenient for a choice of
using it as a preamp only, with separate power amps, and also being able to
use its internal amps as a set of free power amps for additional speakers,
but I hadn't run up against this particular problem before and was very
surprised that I couldn't use it that way.

So I will try it today as a line level but after the volume control
processor between the preamp and power amp. If that works out OK, I will be
happy enough, because I will also be able to use its mono Sub Out feature
with crossover.

Scott have you used products such as this Behringer in your studio work or
some sound reinforcement or PA applicaitons? What I can picture as its main
mission would be such as a small live band using it as an adjunct to their
sound board, the mixer going out thru the Behringer and back in again for
equalization and feedback detection before going out to the amplifier(s) and
speakers. They adjust it for best sound in that room and eliminate possible
feedback problems with the sliders as well. Would that be a good description
of the design intent?

Thanks again,
Gary

Gary Eickmeier

May 18th 14, 06:56 PM

"William Sommerwerck" > wrote in message
...
> "Gary Eickmeier" wrote in message
> ...
>
>> I am having trouble picturing using the tape loop as an
>> external processor because if I need to press that button on the
>> receiver, I can't also press CD or Video
>> or FM Tuner.
>
> Then it's not a "tape loop". There's no loop!
>
> The Tape Monitor switch was intended for three-head machines. * Throwing
> the switch let you hear either the source program, or the actual recording
> coming back from the deck. That's what made them so convenient for
> signal-processing equipment.

Yes, I know William. That is the point of this whole discussion. Please see
my response to Scott above.
>
> * Are you aware there are true three-head DATs, that actually monitor off
> the tape?

I bypassed DAT in my little audio career. Was not doing recording at that
time yet. I saw the Sony PCM F1 in action a couple of times, but have never
experienced DAT.

This leads to a discussion of modern digital recorders, which I had not been
concerned about because they do not need a source/tape monitor function - do
they? We record direct to the memory chip, and that's pretty much that. No
need to check the source against the recording, because - well, I suppose
because you have already checked out the ADC and DAC for accuracy and would
not be using them for professional use if they weren't accurate. Nor could
you read off the chip at the same time as you are recording onto it. I
remember John Atkinson demoing the Sony PCM F1's accuracy at a hi fi show in
England in around 1982. He would insert the recorder between the source (LP
I think) and the playback system and switch between the two. But video
recorders in those days I think had separate record and playback heads, so
if there was no malfunction in the recording onto tape, all he was showing
was the quality of the ADC/DAC throughput. There was no difference, by the
way.

Gary

PStamler

May 19th 14, 12:30 AM

On Sunday, May 18, 2014 11:24:53 AM UTC-6, Gary Eickmeier wrote:
> What I can picture as its main
> mission would be such as a small live band using it as an adjunct to their
> sound board, the mixer going out thru the Behringer and back in again for
> equalization and feedback detection before going out to the amplifier(s) and
>> speakers. They adjust it for best sound in that room and eliminate possible
> feedback problems with the sliders as well. Would that be a good description
> of the design intent?

Typically the Behringer would be used in a simpler way. The mixing board's output would feed the Behringer's input, then the Behringer's output would go directly to the power amps rather than back to the mixer.

Peace,
Paul

hank alrich

May 19th 14, 01:34 AM

Gary Eickmeier > wrote:

> "Hank" > wrote in message
> ...
>
> > Define "receiver" as you are using it. If this is an all-in-one
> > consumer box with both preamp and amplifier in the same unit, you'll
> > need to tap in between the preamp output and the output amplifier
> > input. Tape out->equalizer->tape monitor generally will accomplish
> > this.
>
> Hank,
>
> I don't know how you could have as much audio experience as you claim and be
> so ignorant of a simple term such as "receiver."....
>
> Just kidding, adding to the hilarity from Summerwerck and None.

Know what's funny? Know how there is more than just one guy named
"Gary"? It's like that for guys named "Hank", too.

Reading…

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

Gary Eickmeier

May 19th 14, 06:28 AM

"hank alrich" > wrote in message
...

> Know what's funny? Know how there is more than just one guy named
> "Gary"? It's like that for guys named "Hank", too.

I suspected as much, but I don't think I said anything that would confuse
you with Other Hank.

Gary

Mike Rivers[_2_]

May 19th 14, 11:17 AM

On 5/18/2014 10:02 AM, Scott Dorsey wrote:
> Plenty of old receivers did that so that people in the process of upgrading
> could use them with larger power amplifiers simply as a preamp, but the
> current range of home theatre receivers is designed to be as cheap as possible
> and may be omitting such handy features.

Modern receivers don't have tape outputs and inputs because nobody has a
recorder as part of their home listening system any more. Why record
anything when you can just download it? Those who use a computer as a
recorder and a record as the source are a special breed. If the project
doesn't justify a purpose built phono preamp, a 20 year old stereo
receiver from Goodwill or Craig's List will do the job nicely.

--
"Today's production equipment is IT based and cannot be operated without
a passing knowledge of computing, although it seems that it can be
operated without a passing knowledge of audio" - John Watkinson

Drop by http://mikeriversaudio.wordpress.com now and then

Sean Conolly

May 20th 14, 06:39 PM

"PStamler" > wrote in message
...
> On Sunday, May 18, 2014 11:24:53 AM UTC-6, Gary Eickmeier wrote:
>> What I can picture as its main
>> mission would be such as a small live band using it as an adjunct to
>> their
>> sound board, the mixer going out thru the Behringer and back in again for
>> equalization and feedback detection before going out to the amplifier(s)
>> and
>>> speakers. They adjust it for best sound in that room and eliminate
>>> possible
>> feedback problems with the sliders as well. Would that be a good
>> description
>> of the design intent?
>
> Typically the Behringer would be used in a simpler way. The mixing board's
> output would feed the Behringer's input, then the Behringer's output would
> go directly to the power amps rather than back to the mixer.

Yep. K.I.S.S.

Sean

Gary Eickmeier

May 22nd 14, 01:38 AM

"Sean Conolly" > wrote in message
...
> "PStamler" > wrote in message

>> Typically the Behringer would be used in a simpler way. The mixing
>> board's output would feed the Behringer's input, then the Behringer's
>> output would go directly to the power amps rather than back to the mixer.
>
> Yep. K.I.S.S.
>
> Sean

Well, if someone is still reading, it works just fine in line after the
preamp (receiver) and before the power amps. I finally obtained all of the
adapters I would need to connect it up, and I was more than concerned at
first with a hum and buzz that it was contributing to my system. The
solution was a cheater plug to bypass the ground pin. It is now quiet as a
mouse and works as advertised. At the loudest passages the input level meter
cracks the -6 dB level, well below any concern that it would be overloaded
by using it after the receiver.

Of course I am still playing with it, but the sound is great, full, rich,
dynamic and freq response is now controllable, tho I didn't seem to need
much. It is really lovely having that subwoofer output with adjustable
crossover and level, because now I can equalize the bass region as well.

And the answer to the musical question about modern hi fi receivers is that
there is no tape loop that can act as an external processor loop for
equipment such as the 2496 parametric equalizer. The dumbest feature is that
the power switch is on the back of the thing!

Gary Eickmeier

Ron C[_2_]

May 22nd 14, 03:04 AM

On 5/21/2014 8:38 PM, Gary Eickmeier wrote:
> "Sean Conolly" > wrote in message
> ...
>> "PStamler" > wrote in message
>
>>> Typically the Behringer would be used in a simpler way. The mixing
>>> board's output would feed the Behringer's input, then the Behringer's
>>> output would go directly to the power amps rather than back to the mixer.
>>
>> Yep. K.I.S.S.
>>
>> Sean
>
> Well, if someone is still reading, it works just fine in line after the
> preamp (receiver) and before the power amps. I finally obtained all of the
> adapters I would need to connect it up, and I was more than concerned at
> first with a hum and buzz that it was contributing to my system. The
> solution was a cheater plug to bypass the ground pin. It is now quiet as a
> mouse and works as advertised. At the loudest passages the input level meter
> cracks the -6 dB level, well below any concern that it would be overloaded
> by using it after the receiver.
>
> Of course I am still playing with it, but the sound is great, full, rich,
> dynamic and freq response is now controllable, tho I didn't seem to need
> much. It is really lovely having that subwoofer output with adjustable
> crossover and level, because now I can equalize the bass region as well.
>
> And the answer to the musical question about modern hi fi receivers is that
> there is no tape loop that can act as an external processor loop for
> equipment such as the 2496 parametric equalizer. The dumbest feature is that
> the power switch is on the back of the thing!
>
> Gary Eickmeier
>
>
I'm confused. All my DEQ 2496's have the power switch on the front
and don't have a subwoofer output. Are you maybe talking about the
ultragraph pro 31 band graphic EQ?

==
Later...
Ron Capik
--

Gary Eickmeier

May 22nd 14, 06:29 AM

"Ron C" > wrote in message
...

> I'm confused. All my DEQ 2496's have the power switch on the front
> and don't have a subwoofer output. Are you maybe talking about the
> ultragraph pro 31 band graphic EQ?
>
> ==
> Later...
> Ron Capik

Well, I just looked back briefly in the thread and I didn't make it clear
that I got the 6200. I had already ordered it when I started this thread,
not knowing about the 2496, but it turns out that I can't use the 2496 in my
system because my receiver has no processor loop, which I would think a
digital equalizer/analyzer signal processor would need.

Gary

Ron C[_2_]

May 22nd 14, 02:16 PM

On 5/22/2014 1:29 AM, Gary Eickmeier wrote:
> "Ron C" > wrote in message
> ...
>
>> I'm confused. All my DEQ 2496's have the power switch on the front
>> and don't have a subwoofer output. Are you maybe talking about the
>> ultragraph pro 31 band graphic EQ?
>>
>> ==
>> Later...
>> Ron Capik
>
> Well, I just looked back briefly in the thread and I didn't make it clear
> that I got the 6200. I had already ordered it when I started this thread,
> not knowing about the 2496, but it turns out that I can't use the 2496 in my
> system because my receiver has no processor loop, which I would think a
> digital equalizer/analyzer signal processor would need.
>
> Gary
>
>
Gary,
You can use the 2496 the same way as the 6400, though you
won't have a sub crossover/output.

==
Later...
Ron Capik
--

Scott Dorsey

May 22nd 14, 03:31 PM

Gary Eickmeier > wrote:
>Thanks for the first sensible answer Scott. I checked again, and my receiver
>just does not have an external processor loop available. There is a recorder
>connection with the usual Out and In jacks, but there is no way to select my
>source, such as CD, and also the tape loop. I think what they expect us to
>do with this design is that the selected signal is always available to the
>Tape Out jacks to record from, and then after recording is done you can play
>it back later by selecting the Tape source. But you CANNOT use it
>simultaneously with the CD button as a Source/Tape monitor as in the old
>days. I have my Carver preamp still available for LP sources, and it works a
>little differently. There is a rotary source selector for source such as CD,
>and also two tape loop switches that you can use to select Source or Tape.
>That is what I am used to and hoping that I could do with the new equalizer,
>but as is I'm afraid not.

You really, really need to read the Yamaha Sound Reinforcement Handbook and
get some notion of signal flow through PA systems and how these things are
normally configured.

>Scott have you used products such as this Behringer in your studio work or
>some sound reinforcement or PA applicaitons? What I can picture as its main
>mission would be such as a small live band using it as an adjunct to their
>sound board, the mixer going out thru the Behringer and back in again for
>equalization and feedback detection before going out to the amplifier(s) and
>speakers. They adjust it for best sound in that room and eliminate possible
>feedback problems with the sliders as well. Would that be a good description
>of the design intent?

No, normally this device is used as a speaker processor, that is it comes
in-between the main outputs of the PA console and the speaker systems.
Something like this is used in every PA system in the world pretty much,
although in a lot of cases you will find cruder devices like third-octave
equalizers in use.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Scott Dorsey

May 22nd 14, 03:35 PM

Gary Eickmeier > wrote:
>
>Well, if someone is still reading, it works just fine in line after the
>preamp (receiver) and before the power amps. I finally obtained all of the
>adapters I would need to connect it up, and I was more than concerned at
>first with a hum and buzz that it was contributing to my system. The
>solution was a cheater plug to bypass the ground pin. It is now quiet as a
>mouse and works as advertised. At the loudest passages the input level meter
>cracks the -6 dB level, well below any concern that it would be overloaded
>by using it after the receiver.

Suicide plugs are a very, very stupid idea.

Fix the ground loop by properly configuring the signal wiring rather than
using the suicide plug.

Using a suicide plug on a job is the best way to get instantly fired in the
pro audio world.

In the pro audio world we live with differential inputs and differentially
driven outputs so doing hazardous things is not necessary.
--scott

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

Frank Stearns

May 22nd 14, 05:15 PM

(Scott Dorsey) writes:

>Gary Eickmeier > wrote:
>>
>>Well, if someone is still reading, it works just fine in line after the
>>preamp (receiver) and before the power amps. I finally obtained all of the
>>adapters I would need to connect it up, and I was more than concerned at
>>first with a hum and buzz that it was contributing to my system. The
>>solution was a cheater plug to bypass the ground pin. It is now quiet as a
>>mouse and works as advertised. At the loudest passages the input level meter
>>cracks the -6 dB level, well below any concern that it would be overloaded
>>by using it after the receiver.

>Suicide plugs are a very, very stupid idea.

>Fix the ground loop by properly configuring the signal wiring rather than
>using the suicide plug.

>Using a suicide plug on a job is the best way to get instantly fired in the
>pro audio world.

>In the pro audio world we live with differential inputs and differentially
>driven outputs so doing hazardous things is not necessary.

Scott, you've done a service mentioning this to Gary.

These days, with most modern wiring systems properly installed, particularly in the
context of a "home system" (contrasted with going hundreds of feet between a
console and an amp room), he should not have been able to "solve" the problem by
using a cheater. If he did, the power system likely has some sort of fault, and
perhaps a serious one at that.

Gary, at the very least go out and get one of those $5.00 plug-in wiring checkers.
You plug them in, and the lamp pattern will indicate open ground, open neutral,
flipped hot/neutral/ground, et al. If it's not normal, and you're not absolutely
clear on how to fix whatever you've found wrong, get a licensed electrician. (If
you're selling the place in a few years you'll likely have to do this anyway just to
protect yourself from liability post sale.)

If it shows "normal" you still might want to get an electrician, or start looking
for big leakages between ground and neutral, among other things. (Problem with the
quick testers is that they can't really show "in between" issues that could still be
a problem.)

Your hum problem might be the "canary in the coal mine" for your house wiring. (If
you're in Florida and the house is more than 10-20 years old, I'd start looking
carefully at all the wiring, especially anything going underground. Saw some pretty
weird (and dangerous) problems with older AC wiring in the soggy northwest. Water
won't only rot your wood, it'll play havoc with electrical wiring.)

Frank
Mobile Audio

--

hank alrich

May 22nd 14, 06:15 PM

Gary Eickmeier > wrote:

> "Ron C" > wrote in message
> ...
>
> > I'm confused. All my DEQ 2496's have the power switch on the front
> > and don't have a subwoofer output. Are you maybe talking about the
> > ultragraph pro 31 band graphic EQ?
> >
> > ==
> > Later...
> > Ron Capik
>
> Well, I just looked back briefly in the thread and I didn't make it clear
> that I got the 6200. I had already ordered it when I started this thread,
> not knowing about the 2496, but it turns out that I can't use the 2496 in my
> system because my receiver has no processor loop, which I would think a
> digital equalizer/analyzer signal processor would need.
>
> Gary

Put a DEQ 2496 in series with the output of the receiver. For what do
you need a loop?

Note as well that I doubt anyone here would have suggested the FBQ6200.
It has nowhere near flexiblity equivalent to the DEQ2496.

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

hank alrich

May 22nd 14, 06:15 PM

Scott Dorsey > wrote:

> You really, really need to read the Yamaha Sound Reinforcement Handbook and
> get some notion of signal flow through PA systems and how these things are
> normally configured.

I wonder how many times this will be told to Gary?

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

Gary Eickmeier

May 22nd 14, 06:15 PM

"Scott Dorsey" > wrote in message
...
> Gary Eickmeier > wrote:
>>Thanks for the first sensible answer Scott. I checked again, and my
>>receiver
>>just does not have an external processor loop available. There is a
>>recorder
>>connection with the usual Out and In jacks, but there is no way to select
>>my
>>source, such as CD, and also the tape loop. I think what they expect us to
>>do with this design is that the selected signal is always available to the
>>Tape Out jacks to record from, and then after recording is done you can
>>play
>>it back later by selecting the Tape source. But you CANNOT use it
>>simultaneously with the CD button as a Source/Tape monitor as in the old
>>days. I have my Carver preamp still available for LP sources, and it works
>>a
>>little differently. There is a rotary source selector for source such as
>>CD,
>>and also two tape loop switches that you can use to select Source or Tape.
>>That is what I am used to and hoping that I could do with the new
>>equalizer,
>>but as is I'm afraid not.
>
> You really, really need to read the Yamaha Sound Reinforcement Handbook
> and
> get some notion of signal flow through PA systems and how these things are
> normally configured.
>
>>Scott have you used products such as this Behringer in your studio work or
>>some sound reinforcement or PA applicaitons? What I can picture as its
>>main
>>mission would be such as a small live band using it as an adjunct to their
>>sound board, the mixer going out thru the Behringer and back in again for
>>equalization and feedback detection before going out to the amplifier(s)
>>and
>>speakers. They adjust it for best sound in that room and eliminate
>>possible
>>feedback problems with the sliders as well. Would that be a good
>>description
>>of the design intent?
>
> No, normally this device is used as a speaker processor, that is it comes
> in-between the main outputs of the PA console and the speaker systems.
> Something like this is used in every PA system in the world pretty much,
> although in a lot of cases you will find cruder devices like third-octave
> equalizers in use.
> --scott

OK, thanks for the advice. I would have to call an electrician in to check
my house wiring, just to make sure. The facts are, my house was built in
1989 by a builder who was slightly incompetent and hired some of the
cheapest subs. I believe that my wiring was done by his handyman and not an
electrician, on the theory that everything would be inspected by the county
anyway, so why waste money on electricians? Just do it according to code and
save a buck.

Well, I ordered the wiring for my control room to have four 20 amp lines.
When I got there dor a first look at how it was going, I saw just two lines
running from the box to my audio plugs. They just assumed no normal person
needs anything more than that for a hi fi system. So I ordered them to run
the other two, and make sure they were all 20 Amps. They did it, but who
knows how competently. Half the light switches in the house were installed
upside down, so that I had to press the bottom to turn it on, and vice
versa. I was living in California and selling the old house during the first
3 months of building this one. Once upon a time I obtained a polarity
checker and checked all of my plugs for which side was which to be able to
orient the power cords from my (older) equipment the right way. Didn't keep
records on it, but is that what you mean? Anyway, I should have it checked
thoroughly to make sure the grounds are connected to earth.

One more fact to throw in the bin, my sound system in htis house has always
been quiet as a mouse. No hum, buzz, nothing. The Behringer being inserted
between the receiver and power amps is the first time I have had any
problems.

Gary Eickmeier

Frank Stearns

May 22nd 14, 07:27 PM

"Gary Eickmeier" > writes:

snips

>OK, thanks for the advice. I would have to call an electrician in to check
>my house wiring, just to make sure. The facts are, my house was built in
>1989 by a builder who was slightly incompetent and hired some of the
>cheapest subs. I believe that my wiring was done by his handyman and not an
>electrician, on the theory that everything would be inspected by the county
>anyway, so why waste money on electricians? Just do it according to code and
>save a buck.

>the other two, and make sure they were all 20 Amps. They did it, but who

Hmm. Why 80 amps? Are those breakers all on the same bus bar?

>3 months of building this one. Once upon a time I obtained a polarity
>checker and checked all of my plugs for which side was which to be able to
>orient the power cords from my (older) equipment the right way. Didn't keep

Well, unless it's really old stuff, the polarized pin should get you right. If not
polarized, then we hope there's a double insulated transformer in the device.

>records on it, but is that what you mean? Anyway, I should have it checked
>thoroughly to make sure the grounds are connected to earth.

That, and leakage currents.

>One more fact to throw in the bin, my sound system in htis house has always
>been quiet as a mouse. No hum, buzz, nothing. The Behringer being inserted
>between the receiver and power amps is the first time I have had any
>problems.

Well, could be that the new Behringer is at fault and putting a cheater on it simply
means that you put a mask on the problem -- but something could still wrong with it
internally. Uli has been in trouble before IIRC, including fake UL stickers.
(Actually, might be the classic "pin 1" problem on the box.)

What's the I/O? XLR? TRS? RCA? If XLR or TRS and you're coming from or going to
unbalanced gear, what are you doing with the low side (Pin 3 on XLRs or ring on TRS
plugs.)

Finally, remember that Berhinger price-points mean that they cut a LOT of corners in
design and parts quality.

Frank
Mobile Audio
--

Sean Conolly

May 23rd 14, 03:02 AM

"Gary Eickmeier" > wrote in message
...
>
> "Sean Conolly" > wrote in message
> ...
>> "PStamler" > wrote in message
>
>>> Typically the Behringer would be used in a simpler way. The mixing
>>> board's output would feed the Behringer's input, then the Behringer's
>>> output would go directly to the power amps rather than back to the
>>> mixer.
>>
>> Yep. K.I.S.S.
>>
>> Sean
>
> ... and I was more than concerned at first with a hum and buzz that it was
> contributing to my system. The solution was a cheater plug to bypass the
> ground pin.

As has been pointed out - this is never a good idea. A simple solution is to
run everything off the same phase of power, which is easy to test by running
everything from one outlet. And I do mean *everything* that is electrically
connected by cables including your signal source.

Sean

Gary Eickmeier

May 23rd 14, 06:38 AM

Frank Stearns wrote:
> "Gary Eickmeier" > writes:

> Hmm. Why 80 amps? Are those breakers all on the same bus bar?

I suppose so.

>
>> 3 months of building this one. Once upon a time I obtained a polarity
>> checker and checked all of my plugs for which side was which to be
>> able to orient the power cords from my (older) equipment the right
>> way. Didn't keep
>
> Well, unless it's really old stuff, the polarized pin should get you
> right. If not polarized, then we hope there's a double insulated
> transformer in the device.

I have lots of equipment that doesn't have the polarized prongs on the plug.

>> One more fact to throw in the bin, my sound system in htis house has
>> always been quiet as a mouse. No hum, buzz, nothing. The Behringer
>> being inserted between the receiver and power amps is the first time
>> I have had any problems.
>
> Well, could be that the new Behringer is at fault and putting a
> cheater on it simply means that you put a mask on the problem -- but
> something could still wrong with it internally. Uli has been in
> trouble before IIRC, including fake UL stickers. (Actually, might be
> the classic "pin 1" problem on the box.)

This Behringer is starting to annoy me. I just tested my response with the
sub turned OFF, and I was getting full readings in the bass region from the
main speakers. This means that the crossover is not working for me,
preventing the lowest freqs from reaching the main speakers. I will now have
to figure out a new way of doing it, probably with the receiver's crossover
I also have the Richter. Or I could purchase some other accessory crossover.

But why would they do this? Apparently the sub output of the Behringer is
just another output for the subwoofer, but does not prevent the full range
from going out to the main speakers.

>
> What's the I/O? XLR? TRS? RCA? If XLR or TRS and you're coming from
> or going to unbalanced gear, what are you doing with the low side
> (Pin 3 on XLRs or ring on TRS plugs.)

One strange incident - the sub out has only the XLR output. The manual says
to bridge pins 1 and 3, but my music center that sold me the adapter said
that wasn't necessary because pin 1 is ground already.

>
> Finally, remember that Berhinger price-points mean that they cut a
> LOT of corners in design and parts quality.
>
> Frank
> Mobile Audio

Yes, maybe so.

Gary

hank alrich

May 23rd 14, 03:40 PM

Sean Conolly > wrote:

> "Gary Eickmeier" > wrote in message
> ...
> >
> > "Sean Conolly" > wrote in message
> > ...
> >> "PStamler" > wrote in message
> >
> >>> Typically the Behringer would be used in a simpler way. The mixing
> >>> board's output would feed the Behringer's input, then the Behringer's
> >>> output would go directly to the power amps rather than back to the
> >>> mixer.
> >>
> >> Yep. K.I.S.S.
> >>
> >> Sean
> >
> > ... and I was more than concerned at first with a hum and buzz that it was
> > contributing to my system. The solution was a cheater plug to bypass the
> > ground pin.
>
> As has been pointed out - this is never a good idea. A simple solution is to
> run everything off the same phase of power, which is easy to test by running
> everything from one outlet. And I do mean *everything* that is electrically
> connected by cables including your signal source.

Yes, the Beri might be the problem, or it might be revealing a problem
with the rest of the system. In short <heh> it's not worth getting
electrocuted over. Get out the VOM and start looking for the "why?"
behind it to make sure things are installed and conifured safely.

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

Gary Eickmeier

May 23rd 14, 07:34 PM

"hank alrich" > wrote in message
...

> Put a DEQ 2496 in series with the output of the receiver. For what do
> you need a loop?

I was just thinking that a digital I/O would need to ramain at a certain
level, but the 6200 is doing fine on level, just not so fine on what it is
supposed to be doing for me.
>
> Note as well that I doubt anyone here would have suggested the FBQ6200.
> It has nowhere near flexiblity equivalent to the DEQ2496.

Might just take you up on that. I'm pretty sure I can return this thing to
Parts Express.

Gary

Gary Eickmeier

May 23rd 14, 07:40 PM

"hank alrich" > wrote in message
...

> Yes, the Beri might be the problem, or it might be revealing a problem
> with the rest of the system. In short <heh> it's not worth getting
> electrocuted over. Get out the VOM and start looking for the "why?"
> behind it to make sure things are installed and conifured safely.

I think I will see if I can get the 2496 before I look for the ground loop
etc, I think it will not have the connection problems that the 6200 has. I
will then go out from the receiver from 100 Hz on up, with the subwoof freqs
going straight from the sub out of the receiver to the Velodyne. Alternative
plan would be to go full range out of receiver thru Behringer to an
electronic crossover (I already have a Richter Scale) and then to the power
amps and sub.

Gary

david gourley[_2_]

May 24th 14, 12:40 AM

"Gary Eickmeier" > said...news:ACMfv.132741
:

>
> "hank alrich" > wrote in message
> ...
>
>> Yes, the Beri might be the problem, or it might be revealing a problem
>> with the rest of the system. In short <heh> it's not worth getting
>> electrocuted over. Get out the VOM and start looking for the "why?"
>> behind it to make sure things are installed and conifured safely.
>
> I think I will see if I can get the 2496 before I look for the ground
loop
> etc, I think it will not have the connection problems that the 6200 has.
I
> will then go out from the receiver from 100 Hz on up, with the subwoof
freqs
> going straight from the sub out of the receiver to the Velodyne.
Alternative
> plan would be to go full range out of receiver thru Behringer to an
> electronic crossover (I already have a Richter Scale) and then to the
power
> amps and sub.
>
> Gary
>
>
>

You will still likely have the ground loop problem, and it should be the
highest priority for you to fix it.

david

hank alrich

May 24th 14, 01:22 AM

Gary Eickmeier > wrote:

> "hank alrich" > wrote in message
> ...
>
> > Yes, the Beri might be the problem, or it might be revealing a problem
> > with the rest of the system. In short <heh> it's not worth getting
> > electrocuted over. Get out the VOM and start looking for the "why?"
> > behind it to make sure things are installed and conifured safely.
>
> I think I will see if I can get the 2496 before I look for the ground loop
> etc, I think it will not have the connection problems that the 6200 has. I
> will then go out from the receiver from 100 Hz on up, with the subwoof freqs
> going straight from the sub out of the receiver to the Velodyne. Alternative
> plan would be to go full range out of receiver thru Behringer to an
> electronic crossover (I already have a Richter Scale) and then to the power
> amps and sub.
>
> Gary

Are the receiver outputs balanced? How about the 6200's inputs?

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

Gary Eickmeier

May 24th 14, 05:15 PM

"hank alrich" > wrote in message
...
> Gary Eickmeier > wrote:
>
>> "hank alrich" > wrote in message
>> ...
>>
>> > Yes, the Beri might be the problem, or it might be revealing a problem
>> > with the rest of the system. In short <heh> it's not worth getting
>> > electrocuted over. Get out the VOM and start looking for the "why?"
>> > behind it to make sure things are installed and conifured safely.
>>
>> I think I will see if I can get the 2496 before I look for the ground
>> loop
>> etc, I think it will not have the connection problems that the 6200 has.
>> I
>> will then go out from the receiver from 100 Hz on up, with the subwoof
>> freqs
>> going straight from the sub out of the receiver to the Velodyne.
>> Alternative
>> plan would be to go full range out of receiver thru Behringer to an
>> electronic crossover (I already have a Richter Scale) and then to the
>> power
>> amps and sub.
>>
>> Gary
>
> Are the receiver outputs balanced? How about the 6200's inputs?

No, and this is a problem with amateur or home equipment. Everything is RCA
in and out. The 6200 and the 2496 both have balanced XLR in and out, but
they provide for unbalanced use by telling me to bridge pins 1 and 3 (ground
and cold, or negative). Could that be the cause of my ground loop problem,
because I did not do that?

Anyway, I have ordered the exchange of the 6200 for the DEQ2496, and have
begun studying the owners manual. Might have them send the new unit in
advance so that I could have them both here for a day or two, then send the
6200 back if the new one works fine in my system.

This 2496 will be a whole new learning curve for me, so please stick with
me. Probably should start a new thread about it when I start to use it. You
guys were right, it has some amazing capabilities that I hope I can use. I
want to master it for when I get my new speakers in a month or two.

Thanks for your help Hank and all,

Gary

Ron C[_2_]

May 24th 14, 07:00 PM

On 5/24/2014 12:15 PM, Gary Eickmeier wrote:
> "hank alrich" > wrote in message
> ...
>> Gary Eickmeier > wrote:
>>
>>> "hank alrich" > wrote in message
>>> ...
>>>
>>>> Yes, the Beri might be the problem, or it might be revealing a problem
>>>> with the rest of the system. In short <heh> it's not worth getting
>>>> electrocuted over. Get out the VOM and start looking for the "why?"
>>>> behind it to make sure things are installed and conifured safely.
>>>
>>> I think I will see if I can get the 2496 before I look for the ground
>>> loop
>>> etc, I think it will not have the connection problems that the 6200 has.
>>> I
>>> will then go out from the receiver from 100 Hz on up, with the subwoof
>>> freqs
>>> going straight from the sub out of the receiver to the Velodyne.
>>> Alternative
>>> plan would be to go full range out of receiver thru Behringer to an
>>> electronic crossover (I already have a Richter Scale) and then to the
>>> power
>>> amps and sub.
>>>
>>> Gary
>>
>> Are the receiver outputs balanced? How about the 6200's inputs?
>
> No, and this is a problem with amateur or home equipment. Everything is RCA
> in and out. The 6200 and the 2496 both have balanced XLR in and out, but
> they provide for unbalanced use by telling me to bridge pins 1 and 3 (ground
> and cold, or negative). Could that be the cause of my ground loop problem,
> because I did not do that?
>
> Anyway, I have ordered the exchange of the 6200 for the DEQ2496, and have
> begun studying the owners manual. Might have them send the new unit in
> advance so that I could have them both here for a day or two, then send the
> 6200 back if the new one works fine in my system.
>
> This 2496 will be a whole new learning curve for me, so please stick with
> me. Probably should start a new thread about it when I start to use it. You
> guys were right, it has some amazing capabilities that I hope I can use. I
> want to master it for when I get my new speakers in a month or two.
>
> Thanks for your help Hank and all,
>
> Gary
>
>
You might want to check out these adapters:

Hosa Technology GXM133 Signal Converter
Hosa GXF-132 RCA Female to XLR Female

You should also read this tech note:
< http://www.rane.com/note110.html >

==
Later...
Ron Capik
--

Scott Dorsey

May 24th 14, 07:22 PM

Gary Eickmeier > wrote:
>
>No, and this is a problem with amateur or home equipment. Everything is RCA
>in and out. The 6200 and the 2496 both have balanced XLR in and out, but
>they provide for unbalanced use by telling me to bridge pins 1 and 3 (ground
>and cold, or negative). Could that be the cause of my ground loop problem,
>because I did not do that?

Yes. Follow the directions.
--scott

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

Gary Eickmeier

May 24th 14, 09:18 PM

"Scott Dorsey" > wrote in message
...
> Gary Eickmeier > wrote:
>>
>>No, and this is a problem with amateur or home equipment. Everything is
>>RCA
>>in and out. The 6200 and the 2496 both have balanced XLR in and out, but
>>they provide for unbalanced use by telling me to bridge pins 1 and 3
>>(ground
>>and cold, or negative). Could that be the cause of my ground loop problem,
>>because I did not do that?
>
> Yes. Follow the directions.
> --scott

OK, printing Ron's link and looking up the Hosa connectors.

Gary

Sean Conolly

May 25th 14, 01:00 AM

"hank alrich" > wrote in message
...
> Sean Conolly > wrote:
>
>> "Gary Eickmeier" > wrote in message
>> ...
>> >
>> > "Sean Conolly" > wrote in message
>> > ...
>> >> "PStamler" > wrote in message
>> >
>> >>> Typically the Behringer would be used in a simpler way. The mixing
>> >>> board's output would feed the Behringer's input, then the Behringer's
>> >>> output would go directly to the power amps rather than back to the
>> >>> mixer.
>> >>
>> >> Yep. K.I.S.S.
>> >>
>> >> Sean
>> >
>> > ... and I was more than concerned at first with a hum and buzz that it
>> > was
>> > contributing to my system. The solution was a cheater plug to bypass
>> > the
>> > ground pin.
>>
>> As has been pointed out - this is never a good idea. A simple solution is
>> to
>> run everything off the same phase of power, which is easy to test by
>> running
>> everything from one outlet. And I do mean *everything* that is
>> electrically
>> connected by cables including your signal source.
>
> Yes, the Beri might be the problem, or it might be revealing a problem
> with the rest of the system. In short <heh> it's not worth getting
> electrocuted over. Get out the VOM and start looking for the "why?"
> behind it to make sure things are installed and conifured safely.

Having seen that Gary had four 20A circuits run to that room, I will bet
money that they were put on diffrerent phases just to spread the load out.
That's what happens when you ask for four times the current than you really
need.

I've had full bands rehearsing in my my music room with toob amps and full
PA on a single 20A circuit.

Gary - get some extension cords, it's a cheap work-around for that problem
before something bad happens.

Sean

Gary Eickmeier

May 25th 14, 05:16 AM

"Sean Conolly" > wrote in message
...

> Having seen that Gary had four 20A circuits run to that room, I will bet
> money that they were put on diffrerent phases just to spread the load out.
> That's what happens when you ask for four times the current than you
> really need.
>
> I've had full bands rehearsing in my my music room with toob amps and full
> PA on a single 20A circuit.
>
> Gary - get some extension cords, it's a cheap work-around for that problem
> before something bad happens.
>
> Sean

Running my whole system on extension cords is better than having multiple
outlets?

Anyway, when the 2496 arrives I will straighten this all out.

Gary

Frank Stearns

May 25th 14, 01:49 PM

"Gary Eickmeier" > writes:

>"Sean Conolly" > wrote in message
...

>> Having seen that Gary had four 20A circuits run to that room, I will bet
>> money that they were put on diffrerent phases just to spread the load out.
>> That's what happens when you ask for four times the current than you
>> really need.
>>
>> I've had full bands rehearsing in my my music room with toob amps and full
>> PA on a single 20A circuit.
>>
>> Gary - get some extension cords, it's a cheap work-around for that problem
>> before something bad happens.
>>
>> Sean

>Running my whole system on extension cords is better than having multiple
>outlets?

Under some conditions, yes. (And not your whole system, just cording what's
necessary to pull power of the same side of the breaker box.)

Go to your breaker box. Are the four breakers all on the left column or all on the
right column, or are some on one side and some on the other? If you have them on
both sides, you can eliminate one potential (no pun) problem by selecting one side
or the other, then abandoning plugs on the other side. (I'd orient this based on the
biggest source of dynamic peak load, which would typically be the power amplifiers
-- unless you have true class-A amps, and then is doesn't matter so much because
there is no "peak" load -- those guys are drawing full juice all the time.)

If you wanted to, you could start swapping breakers around to get all the audio
circuits on one side. Breakers pop on an off the buss bars pretty easily, but do NOT
attempt this unless you absolutely know what you're doing, and how to redress
everything inside the box once the moves are completed.

All things being equal, perhaps more important than having huge amounts of excess
current for your set-up is the quality of the circuits. A few of the simplest, best
things you can do for home audio power include:

- circuit for audio power is a "home run" to breaker box -- it's one wire to one
outlet, period. There are NO other outlets or switches, no breaks in the line with
wire nuts in a j-box, whatever. Just breaker to wire to one outlet. Nothing more.
(Or, if you do want more outlets, put them in a dual, triple, or quad box right next
to each other.)

- bump up the size of the wire. So, for example, in a 20A circuit, use 10 gauge
instead of the code-required 12 (Doesn't affect code if you go up in capacity). You
at least lower the R in the R-C-L circuit that is your power feed, and that
helps several things. And, make sure it's good wire! (Been talk lately of counterfit
wire in the CAT-x arena (Cu clad Al). Haven't heard of building wire like this, but
who knows.)

- While the wire size typically dictates this, always make sure the screw terminals
of the outlet are used (avoid the "stabber" poke-in terminals). Periodically kill
the power, pull out the outlets, take a screw driver and make sure the terminals are
still very tight. If over time you run across a screw that seems to keep coming
loose, replace the outlet.

- use higher quality plugs that use good metal with good mechanics on how the
contacts wipe the prongs. Might be overkill, but you could use medical-grade
power plugs if it makes you feel better. I would not, however, get involved with
stupidly over-priced "audiophile" AC outlets. (Might be such a thing, dunno. But
don't bother. Medical grade should be more than adequate.)

- you can also help your audio indirectly by putting noisy loads (such as your
computer equipment, projector, phone chargers, etc) on its own home run, and putting
the breaker(s) for those loads on the buss bar opposide your audio loads. That way,
you're shunting a lot of that switching ps noise all the way back to the transformer
in the street. By that time, it's too feeble to cause many problems.

If equipment power supplies were all perfect (or at least well-built), the above
would not matter all that much. But all too many manufacturers cheap-out when it
comes to the PS. You can live that that if your power is really good, not so much so
otherwise. That's why the techniques above can improve chances for a clean system.

Frank
Mobile Audio

--

hank alrich

May 25th 14, 07:10 PM

Gary Eickmeier > wrote:

> This 2496 will be a whole new learning curve for me, so please stick with
> me. Probably should start a new thread about it when I start to use it. You
> guys were right, it has some amazing capabilities that I hope I can use. I
> want to master it for when I get my new speakers in a month or two.

The DEQ2496 will do many things simultaneously. Proceed cautiously,
because for starters there is one hellaciously powerful parametric EQ in
there, badass enough to get you into big trouble quickly.

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

Gary Eickmeier

May 25th 14, 07:44 PM

"hank alrich" > wrote in message
...

> The DEQ2496 will do many things simultaneously. Proceed cautiously,
> because for starters there is one hellaciously powerful parametric EQ in
> there, badass enough to get you into big trouble quickly.

Yes, will do. I try to always start from the ground up, known to unknown, so
that I might understand what each change has done and can go back. I am
trying to get a grip on the frequency response curve itself right now -
which one do we want, back at the listening position? I don't think it is
going to be "flat" as such, but is there a known ideal room curve for EQing
a system? If there is some agreement on this base question, then I know I
will be able to achieve it with the 2496.

Gary

hank alrich

May 25th 14, 08:04 PM

None > wrote:

> "Frank Stearns" > wrote in message
> ...
> > If you wanted to, you could start swapping breakers around to get
> > all the audio
> > circuits on one side.
>
> This is probably not a good thing to recommend to someone who's been
> responding to noise issued by lifting power grounds.

Good call.

"If you wanted to" - a key phrase, and one would hope to Gawd Herself
that Gary knows better than to do this without the assistance of someone
who knows what's what about it.

If he began this phase (heh) of his latest audio journey by ignoring the
instructions for inserting a balanced I/O device into an unbalanced
signal chain, ****ing around in the breaker box is probably not advised.

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

Gary Eickmeier

May 25th 14, 10:16 PM

"hank alrich" > wrote in message
...
> None > wrote:
>
>> "Frank Stearns" > wrote in message
>> ...
>> > If you wanted to, you could start swapping breakers around to get
>> > all the audio
>> > circuits on one side.
>>
>> This is probably not a good thing to recommend to someone who's been
>> responding to noise issued by lifting power grounds.
>
> Good call.
>
> "If you wanted to" - a key phrase, and one would hope to Gawd Herself
> that Gary knows better than to do this without the assistance of someone
> who knows what's what about it.
>
> If he began this phase (heh) of his latest audio journey by ignoring the
> instructions for inserting a balanced I/O device into an unbalanced
> signal chain, ****ing around in the breaker box is probably not advised.

If you are talking about the bridging of pins 1 and 3 for the subwoofer out
on the 6200, I have eliminated that problem by not using that output. That
was the only XLR out on the 6200. On the 2496, the only XLRs I will have to
use will be the inputs, which I am meeting with my dealer on Tuesday to get
some bridged adapters made up. They were the ones who advised me that the
bridging is not necessary because pin 1 is already a ground, and I believed
them. Taking no chances next time and doing it exactly as described in the
manual.

Anything to avoid the wrath of None.

Gary

John Corbett

May 26th 14, 03:18 AM

On 05/25/2014 07:49 AM, Frank Stearns wrote:
> "Gary Eickmeier" > writes:
>
>
>> "Sean Conolly" > wrote in message
>> ...
>
>>> Having seen that Gary had four 20A circuits run to that room, I will bet
>>> money that they were put on diffrerent phases just to spread the load out.
>>> That's what happens when you ask for four times the current than you
>>> really need.
>>>
>>> I've had full bands rehearsing in my my music room with toob amps and full
>>> PA on a single 20A circuit.
>>>
>>> Gary - get some extension cords, it's a cheap work-around for that problem
>>> before something bad happens.
>>>
>>> Sean
>
>> Running my whole system on extension cords is better than having multiple
>> outlets?
>
> Under some conditions, yes. (And not your whole system, just cording what's
> necessary to pull power of the same side of the breaker box.)
>
> Go to your breaker box. Are the four breakers all on the left column or all on the
> right column, or are some on one side and some on the other? If you have them on
> both sides, you can eliminate one potential (no pun) problem by selecting one side
> or the other, then abandoning plugs on the other side. (I'd orient this based on the
> biggest source of dynamic peak load, which would typically be the power amplifiers
> -- unless you have true class-A amps, and then is doesn't matter so much because
> there is no "peak" load -- those guys are drawing full juice all the time.)
>
> If you wanted to, you could start swapping breakers around to get all the audio
> circuits on one side. Breakers pop on an off the buss bars pretty easily, but do NOT
> attempt this unless you absolutely know what you're doing, and how to redress
> everything inside the box once the moves are completed.
>

Someone is confused about the layout of circuit breakers and their
connections in a panel.

For a standard residential service (3-wire 120/240-volt single-phase)
there are two hot buses in the service panel, and often the circuit
breakers are arranged in two columns.

But putting a group of circuits on one side of the panel does not
guarantee that the circuits are fed from the same side of the mains,
because the columns do not simply correspond to the buses.

In fact, the bus and breaker slot connections are interleaved so
adjacent breakers in each column are fed from different buses.
That's how a single-pole breaker protects a 120V circuit but a two-pole
breaker protects a 240 V circuit. If panels were constructed with all
the breakers in a column being connected to the same bus then a two-pole
breaker would have zero volts between its two "hot" outputs.

--John Corbett

John Hardy

May 26th 14, 05:31 AM

On 5/25/2014 9:18 PM, John Corbett wrote:
> On 05/25/2014 07:49 AM, Frank Stearns wrote:
>> "Gary Eickmeier" > writes:
>>
>>
>>> "Sean Conolly" > wrote in message
>>> ...
>>
>>>> Having seen that Gary had four 20A circuits run to that room, I will
>>>> bet
>>>> money that they were put on diffrerent phases just to spread the
>>>> load out.
>>>> That's what happens when you ask for four times the current than you
>>>> really need.
>>>>
>>>> I've had full bands rehearsing in my my music room with toob amps
>>>> and full
>>>> PA on a single 20A circuit.
>>>>
>>>> Gary - get some extension cords, it's a cheap work-around for that
>>>> problem
>>>> before something bad happens.
>>>>
>>>> Sean
>>
>>> Running my whole system on extension cords is better than having
>>> multiple
>>> outlets?
>>
>> Under some conditions, yes. (And not your whole system, just cording
>> what's
>> necessary to pull power of the same side of the breaker box.)
>>
>> Go to your breaker box. Are the four breakers all on the left column
>> or all on the
>> right column, or are some on one side and some on the other? If you
>> have them on
>> both sides, you can eliminate one potential (no pun) problem by
>> selecting one side
>> or the other, then abandoning plugs on the other side. (I'd orient
>> this based on the
>> biggest source of dynamic peak load, which would typically be the
>> power amplifiers
>> -- unless you have true class-A amps, and then is doesn't matter so
>> much because
>> there is no "peak" load -- those guys are drawing full juice all the
>> time.)
>>
>> If you wanted to, you could start swapping breakers around to get all
>> the audio
>> circuits on one side. Breakers pop on an off the buss bars pretty
>> easily, but do NOT
>> attempt this unless you absolutely know what you're doing, and how to
>> redress
>> everything inside the box once the moves are completed.
>>
>
> Someone is confused about the layout of circuit breakers and their
> connections in a panel.
>
> For a standard residential service (3-wire 120/240-volt single-phase)
> there are two hot buses in the service panel, and often the circuit
> breakers are arranged in two columns.
>
> But putting a group of circuits on one side of the panel does not
> guarantee that the circuits are fed from the same side of the mains,
> because the columns do not simply correspond to the buses.
>
> In fact, the bus and breaker slot connections are interleaved so
> adjacent breakers in each column are fed from different buses.
> That's how a single-pole breaker protects a 120V circuit but a two-pole
> breaker protects a 240 V circuit. If panels were constructed with all
> the breakers in a column being connected to the same bus then a two-pole
> breaker would have zero volts between its two "hot" outputs.
>
> --John Corbett
>

I am not an electrician, so my experience is limited. But one bit of
trivia that I know about service panels and breakers: The old
"Pushmatic" breakers and the panels in which they are used DO have
straight buses (not interleaved), with all of the 120V breakers in the
"left" column powered by the same bus (left bus) and one side of the
240V service, the 120V breakers in the "right" column powered by the
other bus (right bus) and the other side of the 240V service. A 240V
breaker fits into both sides/buses/columns at the same time to make the
full 240V connection.

I was quite surprised to learn later in my adult life that there were
service panels with interleaved buses, but my only experience had been
the Pushmatic type in the family home.

Advantages and disadvantages to each approach.

John Hardy

Frank Stearns

May 26th 14, 05:54 AM

John Corbett > writes:

>On 05/25/2014 07:49 AM, Frank Stearns wrote:
>> "Gary Eickmeier" > writes:
>>
>>
>>> "Sean Conolly" > wrote in message
>>> ...
>>
>>>> Having seen that Gary had four 20A circuits run to that room, I will bet
>>>> money that they were put on diffrerent phases just to spread the load out.
>>>> That's what happens when you ask for four times the current than you
>>>> really need.
>>>>
>>>> I've had full bands rehearsing in my my music room with toob amps and full
>>>> PA on a single 20A circuit.
>>>>
>>>> Gary - get some extension cords, it's a cheap work-around for that problem
>>>> before something bad happens.
>>>>
>>>> Sean
>>
>>> Running my whole system on extension cords is better than having multiple
>>> outlets?
>>
>> Under some conditions, yes. (And not your whole system, just cording what's
>> necessary to pull power of the same side of the breaker box.)
>>
>> Go to your breaker box. Are the four breakers all on the left column or all on the
>> right column, or are some on one side and some on the other? If you have them on
>> both sides, you can eliminate one potential (no pun) problem by selecting one side
>> or the other, then abandoning plugs on the other side. (I'd orient this based on the
>> biggest source of dynamic peak load, which would typically be the power amplifiers
>> -- unless you have true class-A amps, and then is doesn't matter so much because
>> there is no "peak" load -- those guys are drawing full juice all the time.)
>>
>> If you wanted to, you could start swapping breakers around to get all the audio
>> circuits on one side. Breakers pop on an off the buss bars pretty easily, but do NOT
>> attempt this unless you absolutely know what you're doing, and how to redress
>> everything inside the box once the moves are completed.
>>

>Someone is confused about the layout of circuit breakers and their
>connections in a panel.

>For a standard residential service (3-wire 120/240-volt single-phase)
>there are two hot buses in the service panel, and often the circuit
>breakers are arranged in two columns.

>But putting a group of circuits on one side of the panel does not
>guarantee that the circuits are fed from the same side of the mains,
>because the columns do not simply correspond to the buses.

>In fact, the bus and breaker slot connections are interleaved so
>adjacent breakers in each column are fed from different buses.
>That's how a single-pole breaker protects a 120V circuit but a two-pole
>breaker protects a 240 V circuit. If panels were constructed with all
>the breakers in a column being connected to the same bus then a two-pole
>breaker would have zero volts between its two "hot" outputs.

John, my bad -- You are absolutely right. Thanks for the correction. (Been too long
since I've been in a service panel!)

Frank
--

Gary Eickmeier

May 26th 14, 07:21 AM

"Frank Stearns" > wrote in message
acquisition...
> John Corbett > writes:
>
>>On 05/25/2014 07:49 AM, Frank Stearns wrote:
>>> "Gary Eickmeier" > writes:
>>>
>>>
>>>> "Sean Conolly" > wrote in message
>>>> ...
>>>
>>>>> Having seen that Gary had four 20A circuits run to that room, I will
>>>>> bet
>>>>> money that they were put on diffrerent phases just to spread the load
>>>>> out.
>>>>> That's what happens when you ask for four times the current than you
>>>>> really need.
>>>>>
>>>>> I've had full bands rehearsing in my my music room with toob amps and
>>>>> full
>>>>> PA on a single 20A circuit.
>>>>>
>>>>> Gary - get some extension cords, it's a cheap work-around for that
>>>>> problem
>>>>> before something bad happens.
>>>>>
>>>>> Sean
>>>
>>>> Running my whole system on extension cords is better than having
>>>> multiple
>>>> outlets?
>>>
>>> Under some conditions, yes. (And not your whole system, just cording
>>> what's
>>> necessary to pull power of the same side of the breaker box.)
>>>
>>> Go to your breaker box. Are the four breakers all on the left column or
>>> all on the
>>> right column, or are some on one side and some on the other? If you have
>>> them on
>>> both sides, you can eliminate one potential (no pun) problem by
>>> selecting one side
>>> or the other, then abandoning plugs on the other side. (I'd orient this
>>> based on the
>>> biggest source of dynamic peak load, which would typically be the power
>>> amplifiers
>>> -- unless you have true class-A amps, and then is doesn't matter so much
>>> because
>>> there is no "peak" load -- those guys are drawing full juice all the
>>> time.)
>>>
>>> If you wanted to, you could start swapping breakers around to get all
>>> the audio
>>> circuits on one side. Breakers pop on an off the buss bars pretty
>>> easily, but do NOT
>>> attempt this unless you absolutely know what you're doing, and how to
>>> redress
>>> everything inside the box once the moves are completed.
>>>
>
>>Someone is confused about the layout of circuit breakers and their
>>connections in a panel.
>
>>For a standard residential service (3-wire 120/240-volt single-phase)
>>there are two hot buses in the service panel, and often the circuit
>>breakers are arranged in two columns.
>
>>But putting a group of circuits on one side of the panel does not
>>guarantee that the circuits are fed from the same side of the mains,
>>because the columns do not simply correspond to the buses.
>
>>In fact, the bus and breaker slot connections are interleaved so
>>adjacent breakers in each column are fed from different buses.
>>That's how a single-pole breaker protects a 120V circuit but a two-pole
>>breaker protects a 240 V circuit. If panels were constructed with all
>>the breakers in a column being connected to the same bus then a two-pole
>>breaker would have zero volts between its two "hot" outputs.
>
> John, my bad -- You are absolutely right. Thanks for the correction. (Been
> too long
> since I've been in a service panel!)
>
> Frank

The breakers are all on the right side column at positions 22, 26, 36, and
38. Would it help to tell you it is a type G1 enclosure for the breakers?
The breakers are horizontal levers, which probably means not "Pushmatic."

Gary

Gary Eickmeier

May 26th 14, 07:38 AM

"Gary Eickmeier" > wrote in message
...

And I have found some proper XLR to female RCA adapters that have pins 1 and
3 already connected, or bridged, at Parts Express.

Side note on bass management: It was a mistake to get the 6200 for the sub
out connection anyway, because the only correct place to do bass management
is in the receiver, for two good reasons. Number 1, the receiver will manage
the bass for all 5 channels and assign the bass for the subwoofer
accordingly. Number 2, the .1 channel - if there is content in the .1
channel for movies or anything, only the receiver can do the decoding
properly. If I sent just the front L and R channels to the equalizer, I
would not get that .1 channel info for the sub at all.

So, hopefully, I will continue to go out from the sub out of the receiver to
the Velodyne, and from the front two channels to the Behringer 2496, and
from the receiver to the center and surround channels still unequalized. The
2496 requires the unbalanced XLR connector be used for input and for output
I can use either the XLR or the phone jack connectors, both unbalanced.

Will probably be a week before I receive the 2496 and send the 6200 back. In
the meantime, I have the Rane connectors info to study and the 2496 manual.

Gary

Scott Dorsey

June 22nd 14, 06:48 PM

Gary, get this month's JAES and read _Listening Environment Preferences for
Sound Engineers_.

In this case, they measured impulse responses of various control rooms in
multiple dimensions, then used wavefront reconstruction techniques to play
back recorded materials through those simulated control room models through
a large speaker array in an anechoic chamber.

They presented these to a number of mixing engineers and asked them to
evaluate the rooms.

The modelling and reproduction was sufficiently accurate that all but one
of the engineers was able to identify the studio where he most commonly
worked, and one engineer was able to identify it down to two samples.

In any event, what is interesting here is that it permits people to evaluate
(simulated) rooms one after the other without their ears having any chance
to adjust to the room. The discussions about which rooms sounded better
and which sounded worse were very interesting.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."