Reply
 
Thread Tools Display Modes
  #41   Report Post  
Posted to rec.audio.pro
Adrian Tuddenham[_2_] Adrian Tuddenham[_2_] is offline
external usenet poster
 
Posts: 505
Default Freq Response Graph Paper

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
  #42   Report Post  
Posted to rec.audio.pro
hank alrich hank alrich is offline
external usenet poster
 
Posts: 4,736
Default Freq Response Graph Paper

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
  #43   Report Post  
Posted to rec.audio.pro
Gary Eickmeier Gary Eickmeier is offline
external usenet poster
 
Posts: 1,449
Default Freq Response Graph Paper


"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


  #44   Report Post  
Posted to rec.audio.pro
Ron C[_2_] Ron C[_2_] is offline
external usenet poster
 
Posts: 253
Default Freq Response Graph Paper

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
--
  #45   Report Post  
Posted to rec.audio.pro
Ron C[_2_] Ron C[_2_] is offline
external usenet poster
 
Posts: 253
Default Freq Response Graph Paper

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
--



  #46   Report Post  
Posted to rec.audio.pro
Frank Stearns Frank Stearns is offline
external usenet poster
 
Posts: 1,134
Default Freq Response Graph Paper

"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

--
  #47   Report Post  
Posted to rec.audio.pro
Gary Eickmeier Gary Eickmeier is offline
external usenet poster
 
Posts: 1,449
Default Freq Response Graph Paper


"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


  #48   Report Post  
Posted to rec.audio.pro
Peter Larsen[_3_] Peter Larsen[_3_] is offline
external usenet poster
 
Posts: 2,295
Default Freq Response Graph Paper

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



  #49   Report Post  
Posted to rec.audio.pro
Adrian Tuddenham[_2_] Adrian Tuddenham[_2_] is offline
external usenet poster
 
Posts: 505
Default Freq Response Graph Paper

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
  #50   Report Post  
Posted to rec.audio.pro
Don Pearce[_3_] Don Pearce[_3_] is offline
external usenet poster
 
Posts: 2,417
Default Freq Response Graph Paper

On Sat, 10 May 2014 09:25:43 +0100,
lid (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


  #51   Report Post  
Posted to rec.audio.pro
Adrian Tuddenham[_2_] Adrian Tuddenham[_2_] is offline
external usenet poster
 
Posts: 505
Default Freq Response Graph Paper

Don Pearce wrote:

On Sat, 10 May 2014 09:25:43 +0100,
lid (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
  #52   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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."
  #53   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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."
  #54   Report Post  
Posted to rec.audio.pro
Adrian Tuddenham[_2_] Adrian Tuddenham[_2_] is offline
external usenet poster
 
Posts: 505
Default Freq Response Graph Paper

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
  #55   Report Post  
Posted to rec.audio.pro
Frank Stearns Frank Stearns is offline
external usenet poster
 
Posts: 1,134
Default Freq Response Graph Paper

"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

--


  #56   Report Post  
Posted to rec.audio.pro
William Sommerwerck William Sommerwerck is offline
external usenet poster
 
Posts: 4,718
Default Freq Response Graph Paper

"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.

  #57   Report Post  
Posted to rec.audio.pro
Frank Stearns Frank Stearns is offline
external usenet poster
 
Posts: 1,134
Default Freq Response Graph Paper

"Gary Eickmeier" writes:


"Frank Stearns" wrote in message
news
"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


--
  #58   Report Post  
Posted to rec.audio.pro
Frank Stearns Frank Stearns is offline
external usenet poster
 
Posts: 1,134
Default Freq Response Graph Paper

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

--
  #59   Report Post  
Posted to rec.audio.pro
Sean Conolly Sean Conolly is offline
external usenet poster
 
Posts: 638
Default Freq Response Graph Paper

"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


  #60   Report Post  
Posted to rec.audio.pro
Sean Conolly Sean Conolly is offline
external usenet poster
 
Posts: 638
Default Freq Response Graph Paper

"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






  #61   Report Post  
Posted to rec.audio.pro
Gary Eickmeier Gary Eickmeier is offline
external usenet poster
 
Posts: 1,449
Default Freq Response Graph Paper


"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


  #62   Report Post  
Posted to rec.audio.pro
Trevor Trevor is offline
external usenet poster
 
Posts: 2,820
Default Freq Response Graph Paper


"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.


  #63   Report Post  
Posted to rec.audio.pro
Trevor Trevor is offline
external usenet poster
 
Posts: 2,820
Default Freq Response Graph Paper


"Gary Eickmeier" wrote in message
news
"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.


  #64   Report Post  
Posted to rec.audio.pro
Trevor Trevor is offline
external usenet poster
 
Posts: 2,820
Default Freq Response Graph Paper


"Ron C" wrote in message
...
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.


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.]


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.


  #65   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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."


  #66   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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."
  #67   Report Post  
Posted to rec.audio.pro
Frank Stearns Frank Stearns is offline
external usenet poster
 
Posts: 1,134
Default Freq Response Graph Paper

"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

--
  #68   Report Post  
Posted to rec.audio.pro
Peter Larsen[_3_] Peter Larsen[_3_] is offline
external usenet poster
 
Posts: 2,295
Default Freq Response Graph Paper

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


  #69   Report Post  
Posted to rec.audio.pro
None None is offline
external usenet poster
 
Posts: 782
Default Freq Response Graph Paper

"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?





  #70   Report Post  
Posted to rec.audio.pro
PStamler PStamler is offline
external usenet poster
 
Posts: 882
Default Freq Response Graph Paper

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


  #71   Report Post  
Posted to rec.audio.pro
Frank Stearns Frank Stearns is offline
external usenet poster
 
Posts: 1,134
Default Freq Response Graph Paper

"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
--
  #72   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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."
  #73   Report Post  
Posted to rec.audio.pro
[email protected] makolber@yahoo.com is offline
external usenet poster
 
Posts: 614
Default Freq Response Graph Paper



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





  #74   Report Post  
Posted to rec.audio.pro
Peter Larsen[_3_] Peter Larsen[_3_] is offline
external usenet poster
 
Posts: 2,295
Default Freq Response Graph Paper

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



  #75   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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."


  #76   Report Post  
Posted to rec.audio.pro
[email protected] radams2000@gmail.com is offline
external usenet poster
 
Posts: 16
Default Freq Response Graph Paper

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
  #77   Report Post  
Posted to rec.audio.pro
William Sommerwerck William Sommerwerck is offline
external usenet poster
 
Posts: 4,718
Default Freq Response Graph Paper

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 Im going to have to learn about them.

  #78   Report Post  
Posted to rec.audio.pro
Scott Dorsey Scott Dorsey is offline
external usenet poster
 
Posts: 16,853
Default Freq Response Graph Paper

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 Im 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."
  #79   Report Post  
Posted to rec.audio.pro
Gary Eickmeier Gary Eickmeier is offline
external usenet poster
 
Posts: 1,449
Default Freq Response Graph Paper


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


  #80   Report Post  
Posted to rec.audio.pro
Sean Conolly Sean Conolly is offline
external usenet poster
 
Posts: 638
Default Freq Response Graph Paper

"Gary Eickmeier" wrote in message
news

"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


Reply
Thread Tools
Display Modes

Posting Rules

Smilies are On
[IMG] code is On
HTML code is Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
Chicken Or The Egg? Freq Response Or Impedance? Headphone Help! ChrisCoaster Pro Audio 58 October 17th 11 12:56 AM
What do these freq response and xover specs mean? James Lehman Tech 5 August 31st 05 12:03 AM
I need headphones which has a low freq. response of 4Hz. Any brands to recommend? Scorpio Audio Opinions 4 December 12th 03 02:40 AM
Output stage freq response question JamesG Vacuum Tubes 30 October 14th 03 05:08 PM
Freq response ADS AL6 Paul Hanley Car Audio 1 July 30th 03 02:19 AM


All times are GMT +1. The time now is 01:28 AM.

Powered by: vBulletin
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.
Copyright ©2004-2024 AudioBanter.com.
The comments are property of their posters.
 

About Us

"It's about Audio and hi-fi"