[William Sommerwerck]

But if you're trying to fix an error in a mic or a room,
the EQ should make things better.

That's what we've been led to believe. I'm not fully convinced.

With respect to room EQ... Severe room resonances often remain audible, even
though the analyzer reports a net flat response. I've heard this. The
problem is almost certainly caused by the inability of, shall we say,
two-dimensional equalization to correct a three-dimensional error.

[Ron Capik[_3_]]

Roy W. Rising wrote:
Ethan Winer wrote:
On Jul 17, 1:43 pm, Roy W. Rising
wrote:
Really steep slopes cause objectionable phase distortions. I've heard
them!
How do you know that what you heard was phase shift? Using the best
descriptive words you can muster, describe what you believe to be the
objectionable part of a roll-off with a slope steeper than 6 dB per
octave.

--Ethan

OK. I do not know that the cause was phase shift. That said, the mode was
not roll-off. The device was a '60's McCurdy EQ with 3- or 6KHz haystack
boost. When used with more than about 6dB of 6KHz boost, sources had what
I can describe only as a "pinched" characteristic. I think it was masked
in the JBL monitors being used, it became evident when EV Sentry 500s
became available. The system was very well engineered, I seriously doubt
anything down stream was being overloaded.

In another circumstance, a UREI 1/3 octave graphic EQ was being used to
correct for known minor discrepancies in a 'standard' mic. A group of
'golden ears' listened to reference material and characterized what they
heard as "something wrong". None could be any clearer. When the EQ was
flattened, all agreed to leave it that way. Whatever was bothering them
was gone.

I can't remember which publication long ago raised this topic and offered
the "6dB/octave" suggestions. The arguments were persuasive.

Probably not the article you're thinking
of but an interesting read:

Michael Gerzon's
"Why do equalisers sound different?"

http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf



Later...
Ron Capik
--

[William Sommerwerck]

Michael Gerzon's
"Why do equalisers sound different?"


http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf

Michael Gerzon was truly a genius, and I would normally defer to his opinion
about anything. But I don't think he's correctly interpreting Harwood's
discovery...

In the late 1950s, H D Harwood at the BBC made a
discovery whose importance is still not fully
recognised. In investigating the performance of
loudspeakers, he discovered that low-level delayed
resonances severely coloured the reproduced sound
even if these resonances were 40dB below the main
speaker response. At first sight there's nothing very
world-shaking about that. But consider what the effect
of such a delayed resonance is on the amplitude and
phase response; 40dB down means a signal whose
amplitude is only 1% of the main signal. This means
that the amplitude response must vary only between
99% and 101% of flat, ie within ±0.1 dB. The effect
on phase response must similarly be within 1/100 rad,
ie within ±0.6o. In other words even in the late 1950s
Harwood showed that variations in phase response of
around only 1o and in amplitude response of ±0.1dB
produced audible colouration.

I'm not going to get into any discussion about this. Draw your own
conclusions.

[Peter Larsen[_3_]]

Roy W. Rising wrote:

OK. I do not know that the cause was phase shift. That said, the
mode was not roll-off. The device was a '60's McCurdy EQ with 3- or
6KHz haystack boost. When used with more than about 6dB of 6KHz
boost, sources had what I can describe only as a "pinched"
characteristic.

At an early morning guess it was ringing.

In another circumstance, a UREI 1/3 octave graphic EQ was being used
to correct for known minor discrepancies in a 'standard' mic. A
group of 'golden ears' listened to reference material and
characterized what they heard as "something wrong". None could be
any clearer. When the EQ was flattened, all agreed to leave it that
way. Whatever was bothering them was gone.

Sharp peaks are always annoyingly audible. So if you eq to fix something and
misses because of not using a parametric it gets easy to produce a sharp
peak next to the dip you're trying to fix.

FFT analysis can be great at telling you where and how large that pesky peak
that needs fixing is, my CK1's had an 8 dB peak at 16 kHz when I bougth them
some 10+ years ago, now it is only a 6 dB peak ... correcting it makes
violins and multiple singing wimmen sound right, otherwise "there is
something wrong".


Kind regards

Peter Larsen

[Peter Larsen[_3_]]

William Sommerwerck wrote:


In the late 1950s, H D Harwood at the BBC made a
discovery whose importance is still not fully
recognised. In investigating the performance of
loudspeakers, he discovered that low-level delayed
resonances severely coloured the reproduced sound
even if these resonances were 40dB below the main
speaker response.

He wrote about it in Wireless World, I think, in the 1970's. Also about the
importance of directivity index.

At first sight there's nothing very
world-shaking about that. But consider what the effect
of such a delayed resonance is on the amplitude and
phase response; 40dB down means a signal whose
amplitude is only 1% of the main signal. This means
that the amplitude response must vary only between
99% and 101% of flat, ie within ±0.1 dB. The effect
on phase response must similarly be within 1/100 rad,
ie within ±0.6o. In other words even in the late 1950s
Harwood showed that variations in phase response of
around only 1o and in amplitude response of ±0.1dB
produced audible colouration.

No he didn't, what he explained was that the delayed resonance was
griveously audible BECAUSE the output from the resonance continued after the
exiting signal had stopped. And that is why you can't fix a poor transducer
with EQ, and why metal center domes are so darn obnoxious.

I'm not going to get into any discussion about this. Draw your own
conclusions.

Kind regards

Peter Larsen

[Arny Krueger]

"Roy W. Rising" wrote in
message

Really steep slopes cause objectionable phase
distortions. I've heard them!

This sentence misses being 100% correct for the lack of a little word: can.

It takes a ton of phase shift equally applied to both channels to be
audible. 1,000 degrees in the midrange, more at higher frequencies.

It takes almost no phase shift applied to just one channel that is mixed
with other similar channels to be audible.

There are steep slope digital filters that have no phase distortion to speak
of. They are called "linear phase filters". They are commonly used on
better DAC chips.

They can still have characteristic audible colorations, or not. It all
depends.

[Arny Krueger]

"Ron Capik" wrote in message

Probably not the article you're thinking
of but an interesting read:

Michael Gerzon's
"Why do equalisers sound different?"

http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf

The short answer to the question: "Why do equalisers sound different?" is:

because that is what they are supposed to do.

;-)

When someone says that they tried to equalize something and it sounded
worse, then they either had inadequate tools, or were personally not up to
the task, or were on mission impossible. Sometimes it is impossible to know
which is the problem.

[Arny Krueger]

"Ty Ford" wrote in message
al.NET

When speaking in real-time though one of Bob's boxes with
the phase rotators engaged and headphones on, your own
voice sounds "stuffy", sort of like you have a head cold.

That effect is no big challenge with a straight up parametric eq.

[William Sommerwerck]

Michael Gerzon's
"Why do equalisers sound different?"

http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf

The short answer to the question: "Why do equalisers sound different?" is:
because that is what they are supposed to do.

That isn't at all what he meant. He meant... "Why do equalizers with
nominally 'identical' settings sound different?"

[Arny Krueger]

"William Sommerwerck" wrote in
message
Michael Gerzon's
"Why do equalisers sound different?"

http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf

The short answer to the question: "Why do equalisers
sound different?" is: because that is what they are
supposed to do.

That isn't at all what he meant. He meant... "Why do
equalizers with nominally 'identical' settings sound
different?"

All answers to that question involve linear distortion, nonlinear
distortion, gain staging, and/or noise.

There are no effective standards for the correlation between the controls
and the response (amplitude or phase) curves on equalizers.

Not all equalizers have inaudible distortion at extreme settings, even if
they have inaudible distortion at nominal settings.

Equipment varies in terms of noise and clipping points, both input and
output. The variances are not all well documented.

[Roy W. Rising[_2_]]

"Peter Larsen" wrote:
Roy W. Rising wrote:

OK. I do not know that the cause was phase shift. That said, the
mode was not roll-off. The device was a '60's McCurdy EQ with 3- or
6KHz haystack boost. When used with more than about 6dB of 6KHz
boost, sources had what I can describe only as a "pinched"
characteristic.

At an early morning guess it was ringing.

Could be. The cause???

In another circumstance, a UREI 1/3 octave graphic EQ was being used
to correct for known minor discrepancies in a 'standard' mic. A
group of 'golden ears' listened to reference material and
characterized what they heard as "something wrong". None could be
any clearer. When the EQ was flattened, all agreed to leave it that
way. Whatever was bothering them was gone.

Sharp peaks are always annoyingly audible. So if you eq to fix something
and misses because of not using a parametric it gets easy to produce a
sharp peak next to the dip you're trying to fix.

FFT analysis can be great at telling you where and how large that pesky
peak that needs fixing is, my CK1's had an 8 dB peak at 16 kHz when I
bougth them some 10+ years ago, now it is only a 6 dB peak ... correcting
it makes violins and multiple singing wimmen sound right, otherwise
"there is something wrong".

Kind regards

Peter Larsen

In the second example, my use of "minor discrepancies" was to connote that
the adjustments were less than plus/minus 2dB. Adjacent sliders were less
than 1dB apart. No "sharp peaks" were present.

--
~ Roy
"If you notice the sound, it's wrong!"

[Ethan Winer[_3_]]

On Jul 19, 8:03 pm, Roy W. Rising
wrote:
OK. I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way.

You already got good replies from Peter and Arny. I'll add only that
people seem to blame "phase shift" for various ills they can't
describe. This is a great example. I know a guy who swears he can hear
the phase shift in a 10-foot guitar cable, which of course is
ridiculous. But he absolutely believes that's what he hears! If you
watch my Audio Myths video I linked earlier there's a demo of phase
shift. I urge you to download and play around with the free Sanford
Phaser VST plug-in used in that demo:

http://www.ethanwiner.com/aes/

--Ethan

[Mark]

OK. *I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". *None could be any clearer. *When the EQ was flattened, all agreed to leave it that way.



steep filters create ringing due to Gibbs phenomenon no matter how the
phase or group delay has been equalized

If you take a perfect square wave and simply truncate the harmonics,
without any change in phase or anything to the the remaining
harmonics, you will still have Gibbs ringing

steep min phase filters cause the most familiar post ringing

steep filters with linear phase and flat group delay have equal pre
and post ringing

you can argue that pre ringing is more objectional compared to post
ringing because we are not accustomed to hearing that

that leads to the conclusion that stepe filters with min phase may
sound better compared to steep filter with linear phase...

but the best sounding filters are filters that are not steep.

it is an unfortunate fact of nature that you simply can't create a
steep change in the frequency response without messing up the time
domain response.

this is not a limitation of analog or digital, it is just the way
nature works..

http://en.wikipedia.org/wiki/Gibbs_phenomenon

Mark

[Arny Krueger]

"Mark" wrote in message

OK. I do not know that the cause was phase shift ...
sources had what I can describe only as a "pinched"
characteristic ... A group of 'golden ears' listened to
reference material and characterized what they heard as
"something wrong". None could be any clearer. When the
EQ was flattened, all agreed to leave it that way.

steep filters create ringing due to Gibbs phenomenon no
matter how the phase or group delay has been equalized

If you take a perfect square wave and simply truncate the
harmonics
without any change in phase or anything to the the
remaining harmonics, you will still have Gibbs ringing

This should be a major concern to people who spend a lot of time listening
to ideal square waves and ideal impulses.

When you're dealing with musical sounds, everything is ringing quite a bit
anyhow. That's what musical instruments do - they resonate and ring.

steep min phase filters cause the most familiar post ringing

So do virtually all musical instruments.

steep filters with linear phase and flat group delay have
equal pre and post ringing

The good news is that linear phase filters most popular use is as brickwall
filters in digital, and the sample rate is usually high enough that the
ear's sensitivity to *anything* is very low.

you can argue that pre ringing is more objectional
compared to post
ringing because we are not accustomed to hearing that

Pre-ringing from linear phase filters is probably most audible when people
downsample to sample rates with nyquist frequencies in the normal audible
range.

that leads to the conclusion that stepe filters with min
phase may sound better compared to steep filter with linear phase...

Or not. A conclusion with the word "may" in it is arguably not a conclusion
but a speculation. Regardless, its not exactly exact.

but the best sounding filters are filters that are not
steep.

Unless you use the steep filter to deep-six some objectionable sound at a
frequency they put out-of band, which is mostly why people use them.

it is an unfortunate fact of nature that you simply can't
create a steep change in the frequency response without messing up
the time domain response.

AFAIK nobody heavily filters music with filters that run inside the audio
band, that they already like the sound of.

this is not a limitation of analog or digital, it is just
the way nature works..

Right. Most of the things that people complain about when digital does it,
they would complain about it if analog was commonly used (or even able!) to
do it.

[William Sommerwerck]

If you take a perfect square wave and simply truncate the
harmonics without any change in phase or anything to the
remaining harmonics, you will still have Gibbs ringing.

That's not Gibbs ringing. It's simply a waveform.

[Mark]

On Jul 20, 6:17*pm, "William Sommerwerck"
wrote:
If you take a perfect square wave and simply truncate the
harmonics without any change in phase or anything to the
remaining *harmonics, you will still have Gibbs ringing.

That's not Gibbs ringing. It's simply a waveform.

There are numerous references about Gibbs phenomenon and ringing.

There ARE other forms of ringing that could be called "simply a
waveform" but the ringing on a step (square wave) caused by a steep
filter (truncation of Fourier components) is called Gibbs.

I don't care to debate this with you..

Mark

[Mike Rivers]

Arny Krueger wrote:

This should be a major concern to people who spend a lot of time listening
to ideal square waves and ideal impulses.

Isn't that what synthesizers (and their equivalent virtual
instrument plug-ins) mostly produce?

AFAIK nobody heavily filters music with filters that run inside the audio
band, that they already like the sound of.

Your definition of "music" is traditional. Today's music is
frequently mangled with heavy filtering that's modulated so
that a rather boring sound becomes dynamic. It starts with a
wah-wah pedal and goes up (in degrees of phase shift) from
there. Those of us who record what we hear are less
concerned with those effects because we don't intentionally
apply them.


--
"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

[Peter Larsen[_3_]]

Roy W. Rising wrote:

"Peter Larsen" wrote:
Roy W. Rising wrote:

OK. I do not know that the cause was phase shift. That said, the
mode was not roll-off. The device was a '60's McCurdy EQ with 3- or
6KHz haystack boost. When used with more than about 6dB of 6KHz
boost, sources had what I can describe only as a "pinched"
characteristic.

At an early morning guess it was ringing.

Could be. The cause???

A filter is always also a resonator, the use of the wording Q says it, and a
1960's McCurdy EQ - whatever that is since it pre-dates my audio
experience - is likely to use a combination of induction and capacitance,
ie. have real induction coils in each band; real components do not always
exhibit theory-compliant performance. Iron cored inductors may have been
what was in it, and those are "un-simple" components that are not good at
behaving like simple theoretical models suggest that they should. Also - at
a guess - they may have been dimensioned so that they were saturated when a
large boost was used with a strong signal level present. That surely would
sound "pinched" referencing my experience with intentionally overloading
Bang and Olufsen tape reocrders used for tape echo by having a treble boost
in the loop.

In another circumstance, a UREI 1/3 octave graphic EQ was being used
to correct for known minor discrepancies in a 'standard' mic.

We need to talk about what you can and can not fix when it comes to
transducer oddities. Frequency response per se is easy to address, such as
the example issue with the CK1's and what appears to be a resonance in front
of the membrane. But frequency response aberrations in transducers do NOT
have the neat shape in the specsheet, it is smoothed so that a minor bump
may hide s narrow sharp peak, as with that CK1 violin sound issue.

A
group of 'golden ears' listened to reference material and
characterized what they heard as "something wrong". None could be
any clearer. When the EQ was flattened, all agreed to leave it that
way. Whatever was bothering them was gone.

Some of the time it is like that when EQ'ing transducers, it could be fun to
speculate along lines such as whether the mind knows a "probable"
combination of colorations from an "improbable" one in case one doesn't get
the response fix exactly right. Some transducers have aberrations that are
musical - Shure 565 is to my sonic preference such a transducer and some are
obnoxious, to my sonic preference the SM58 is just that, I reckon the grid
is ""acoustically worse"".

[une snippage]

In the second example, my use of "minor discrepancies" was to connote
that the adjustments were less than plus/minus 2dB. Adjacent sliders
were less than 1dB apart. No "sharp peaks" were present.

Roy, here you are plain wrong because of not considering the whole system
and because of being mmislead by smoothed measuring data. You were
equalizing a transducer, such always have sharp peaks - good transducers
just don't have quite as large ones as bad transducers - and a narrow sharp
8 dB peak that is lowered by 2 dB and has the area on both sides of it
lowered with it is quite likely to sound worse rather than better, this
because the peak is still there relative to its surround response and a
wider dip has been added, worst case changing somethng that is audibly
obnoxious to something that is obnoxious combined with something that is
audibly boring.

Some of the time the ""thingies"", be it monitor loudspeakers or recording
mics, are better left sounding as the do ex works; ref. also the general
"can I eq my monitors + room discussion" that comes up every now and then,
frequency response fixes to my sonic taste have to match the actual
frequency response aberration.


Kind regards

Peter Larsen

[Ethan Winer[_3_]]

On Jul 20, 4:46 pm, "Arny Krueger" wrote:
When you're dealing with musical sounds, everything is ringing quite a bit anyhow. That's what musical instruments do - they resonate and ring.

I nominate this for Quote of the Week.

--Ethan

[Live Sound Audio]

For live sound, it really depends on where you're at in the signal chain. For mains EQ I rarely boost any eq because of how it sonically makes the the overall sound strange to my ears.

So I'd be a "minus only" guy for mains EQ and use an FFT scope to figure out where to cut - it's most often somewhere around 250 because of room excitement and design.

Yes there are some FFT areas where it would seem you'd want to boost but I don't.

For channel eq there's no limit - it's just what sounds right + or -

[Frank Stearns]

Ethan Winer writes:

On Jul 20, 4:46 pm, "Arny Krueger" wrote:
When you're dealing with musical sounds, everything is ringing quite a bit anyhow. That's what musical instruments do - they resonate and ring.

I nominate this for Quote of the Week.

And as Ethan will appreciate, rooms where we might hear those instruments will also
resonate and ring -- but hopefully in a good way (such as a good,
repertoire-appropriate concert hall), rather than a small bedroom studio with
+/- 20 dB below 200 Hz, with RT30 componenents equally as crazy!

Frank
Mobile Audio
--

[Scott Dorsey]

Ethan Winer wrote:
You already got good replies from Peter and Arny. I'll add only that
people seem to blame "phase shift" for various ills they can't
describe. This is a great example. I know a guy who swears he can hear
the phase shift in a 10-foot guitar cable, which of course is
ridiculous. But he absolutely believes that's what he hears!

I'll bet he can hear the _amplitude_ alterations made by the high frequency
rolloff in a 10-foot guitar cable, though. Doesn't take much shunt capacitance
to totally screw things up when you have a 1M input impedance? Whoever
decided that was a good idea anyway?
--scott


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

[Les Cargill[_3_]]

Ethan Winer wrote:
On Jul 19, 8:03 pm, Roy W.
wrote:
OK. I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way.

You already got good replies from Peter and Arny. I'll add only that
people seem to blame "phase shift" for various ills they can't
describe. This is a great example. I know a guy who swears he can hear
the phase shift in a 10-foot guitar cable, which of course is
ridiculous. But he absolutely believes that's what he hears!

But he may be right about that. If the input is a 1M input, a
wee amount of capacitance can make a (subtle) difference. Might
not be phase shift, though. Guitar players are really
serious about the guitar-amp interconnect. But they
won't (usually) buffer...

If you
watch my Audio Myths video I linked earlier there's a demo of phase
shift. I urge you to download and play around with the free Sanford
Phaser VST plug-in used in that demo:

http://www.ethanwiner.com/aes/

--Ethan

--
Les Cargill

[Richard Webb[_3_]]

On Wed 2038-Jul-21 12:35,
Live Sound Audio writes:
For live sound, it really depends on where you're at in the signal
chain. For mains EQ I rarely boost any eq because of how it
sonically makes the the overall sound strange to my ears.


WOuld agree with that. WHen I've had a knowledgeable person to read appropriate displays etc. I've used pink noise and
analysis tools, otherwise, old blind sound dog uses those
two things on the side of his head.

FOr channel strip eq I try to avoid it if at all possible.
If I can't get a channel sounding good with a modest cut
(preferred) or boost then I try for a different capture
technique, i.e. move microphone, or a different mic
altogether.


Regards,
Richard
.... Remote audio in the southland: See www.gatasound.com
--
| Remove .my.foot for email
| via Waldo's Place USA Fidonet-Internet Gateway Site
| Standard disclaimer: The views of this user are strictly his own.

[ChrisCoaster]

On Jul 16, 4:34*pm, (Scott Dorsey) wrote:
Nono wrote:
After being confused for a while after reading and hearing all this
propagating of substracting EQ-ing as a better alternative for
additive EQ-ing, I come to the conclusion that this claim is inacurate
to say the least.

It is a good rule of thumb, but it is not a hard and fast rule in any
way and you don't need to overintellectualize it.

It is a good idea to think about tone shaping in terms of cutting things
out, and many equalizers sound better that way.

It is like the three-to-one rule for microphone placement; it's usually
a good idea but it's no more than that. *However, it can be a good rule
to follow until you learn the exceptions.
--scott
--
"C'est un Nagra. *C'est suisse, et tres, tres precis."
______________________
Not *exactly* on topic - but involves use of an equalizer:

I analyzed several "Equal-Loudness Contour"s(essentially modified F-M
curves), and applied the settings to my Windows Media player EQ.
Assumptions: We do not know the Q of each band's center point on the
10-band EQ that WMP comes with.

Classic Fletcher-Munson: http://www.customanalogue.com/elsino...unson_700W.gif

Updated Equal-Loudness Contour:
http://upload.wikimedia.org/wikipedi...indos1.svg.png
(I find that correlating to the 60 or 80 phon curve yields best
results for moderate to low-level listening.)


That said, I do find that when I approximate the curves in the above
links, I find my self listening to music for longer periods at a time,
at lower volumes, through both headphones and through loudspeakers!
It's quite phenomenal. I subconsciously find myself hearing MORE of
the recording at LOWER avg. volume levels! I encourage anyone to try
this and report back their results.

The Settings I derived:
Windows MP EQ:
31Hz +12dB
62 +6dB
125 +1dB
250 0 adjustment(leave flat)
500 -1dB
1kHz 0 adjustment(flat)
2kHz -1dB
4 -3
8 +4
16 leave flat

Again these are for MODERATE to LOW volume listening ( 50dB and under)
- not for rocking your house party or jamming through your
Sennheisers(!) Listening too loud may not harm your ears - but it
could clip your amp or damage your speaker drivers with all that
bottom.


-ChrisCoaster

[Adrian Tuddenham[_2_]]

Richard Webb wrote:
[...]
FOr channel strip eq I try to avoid it if at all possible.
If I can't get a channel sounding good with a modest cut
(preferred) or boost then I try for a different capture
technique, i.e. move microphone, or a different mic
altogether.

If the sound is wrong, shouldn't you try moving the mic first?

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

[Anahata]

On Wed, 21 Jul 2010 15:08:42 -0400, Scott Dorsey wrote:

Doesn't take much
shunt capacitance to totally screw things up when you have a 1M input
impedance?

Pedantically, it's the high source impedance of the pickup (designed to
require 1M input at the amplifier) that screws things up with capacitance.

Whoever decided that was a good idea anyway?

I've always assumed it was a legacy from tube amplifier days, when 1M
input impedance was normal anyway, and a high-z pickup coil gave you a
nice big signal. For the same reason mics used to come with High-z, at
least as an option.

--
Anahata
==//== 01638 720444
http://www.treewind.co.uk

[Arny Krueger]

"Live Sound Audio"
wrote in message


For live sound, it really depends on where you're at in
the signal chain. For mains EQ I rarely boost any eq
because of how it sonically makes the the overall sound
strange to my ears.

So I'd be a "minus only" guy for mains EQ and use an FFT
scope to figure out where to cut - it's most often
somewhere around 250 because of room excitement and
design.

If you're trying to equalize out dips due to acoustic cancellation by
peaking the electrical response, you're on mission impossible. You can waste
amazing amounts of power in these dips, and have very little to show for it
but hot equipment.

[[email protected]]

On 2010-07-22 lid(AdrianTuddenham) said:
Newsgroups: rec.audio.pro
If I can't get a channel sounding good with a modest cut
(preferred) or boost then I try for a different capture
technique, i.e. move microphone, or a different mic
altogether.
If the sound is wrong, shouldn't you try moving the mic first?
INdeed. I find this to be the case. If it's wrong that
means a bit of a different mic position will often take care
of it. IF it's a slight tweak I might just live iwth it for
awhile though, move the mic later during a break or
something g.






Richard webb,

replace anything before at with elspider
Remote audio in the southland: see www.gatasound.com

[Ethan Winer[_3_]]

On Jul 21, 3:08 pm, (Scott Dorsey) wrote:
Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway?

Les Paul? Leo Fender?

:-)

--Ethan

[Les Cargill[_3_]]

Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote:
Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway?

Les Paul? Leo Fender?

:-)

--Ethan

Les Paul used a lo-Z Les Paul Recording for some
span of time.

--
Les Cargill

[Ron Capik[_3_]]

ChrisCoaster wrote:
On Jul 16, 4:34 pm, (Scott Dorsey) wrote:
Nono wrote:
After being confused for a while after reading and hearing all this
propagating of substracting EQ-ing as a better alternative for
additive EQ-ing, I come to the conclusion that this claim is inacurate
to say the least.
It is a good rule of thumb, but it is not a hard and fast rule in any
way and you don't need to overintellectualize it.

It is a good idea to think about tone shaping in terms of cutting things
out, and many equalizers sound better that way.

It is like the three-to-one rule for microphone placement; it's usually
a good idea but it's no more than that. However, it can be a good rule
to follow until you learn the exceptions.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
______________________
Not *exactly* on topic - but involves use of an equalizer:

I analyzed several "Equal-Loudness Contour"s(essentially modified F-M
curves), and applied the settings to my Windows Media player EQ.
Assumptions: We do not know the Q of each band's center point on the
10-band EQ that WMP comes with.

...snip...

-ChrisCoaster

However, "we" can surly measure the 10-band EQ, can "we" not?

Later...
Ron Capik
--

[Scott Dorsey]

Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote:
Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway?

Les Paul? Leo Fender?

Les Paul decided it was a bad idea and started making guitars with a
600 ohm balanced "recording" pickup for a while.... no more hum, no more
buzz, no more cable issues...
--scott

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

[Mike Rivers]

Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote:
Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway?

Les Paul? Leo Fender?

There was a Les Paul Recording model with an XLR connector
and low impedance pickups, and 'ol Chet also used low
impedance pickups at one time. I suppose it wasn't embraced
by the manufactueres for a few reasons:

- Cost. They'd need more gain and a more expensive connector
- Compatibility. They'd have to make amplifiers with both
low-Z and high-Z inputs for the next 50 years
- Documentation and education. They'd have to teach all the
guitar players, and even harder, all the dealers, what it's
all about.


--
"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

[ChrisCoaster]

On Jul 22, 1:49*pm, Ron Capik wrote:


However, "we" can surly measure the 10-band EQ, can "we" not?

Later...
Ron Capik
--- Hide quoted text -

- Show quoted text -
________________
Expand on that thought please, Ron?

-CC

[Ron Capik[_3_]]

ChrisCoaster wrote:
On Jul 22, 1:49 pm, Ron Capik wrote:

However, "we" can surly measure the 10-band EQ, can "we" not?

Later...
Ron Capik
--- Hide quoted text -

- Show quoted text -
________________
Expand on that thought please, Ron?

-CC

I'm guessing there must be dozens of programs
out there that can be used to measure the Q of
an equalizer.

One way would be run some pink noise through
the Media Player then set the EQ to some
value and run the same noise through the
Media Player. Take FFTs of both and subtract
the two FFTs.

In a quick experiment using your
settings I got something like:

31 +11.2
62 +7.5
125 +3.3
250 [set as 0.0 reference]
500 -1.9
1L -2.0
2k -2.0
4k -3.6
8k +2.0
16k ~-1.0


Later...
Ron Capik
--

[Ethan Winer[_3_]]

On Jul 22, 2:13 pm, (Scott Dorsey) wrote:
Les Paul decided it was a bad idea and started making guitars with a
600 ohm balanced "recording" pickup for a while.... no more hum, no more
buzz, no more cable issues...

Thanks Scott (and Les and Mike). Yes, I actually remember that now. :-
)

--Ethan

[rochrist]

Mike Rivers wrote:
Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote:
Doesn't take much shunt capacitance to totally screw things up when
you have a 1M input impedance? Whoever decided that was a good idea
anyway?

Les Paul? Leo Fender?

There was a Les Paul Recording model with an XLR connector and low
impedance pickups, and 'ol Chet also used low impedance pickups at one
time. I suppose it wasn't embraced by the manufactueres for a few reasons:


There were two versions of this guitar, the first had the XLR connector
and an impedance converter as part of the cable. That was the Les Paul
Professional. It was followed after no more than two years by the Les
Paul recording that had a normal output connector and had a switch on
the face plate to switch between high and low impedance.

[ChrisCoaster]

On Jul 22, 6:17*pm, Ron Capik wrote:
ChrisCoasterwrote:
On Jul 22, 1:49 pm, Ron Capik wrote:

However, "we" can surly measure the 10-band EQ, can "we" not?

Later...
Ron Capik
--- Hide quoted text -

- Show quoted text -
________________
Expand on that thought please, Ron?

-CC

I'm guessing there must be dozens of programs
out there that can be used to measure the Q of
an equalizer.

One way would be run some pink noise through
the Media Player then set the EQ to some
value and run the same noise through the
Media Player. Take FFTs of both and subtract
the two FFTs.

In a quick experiment using your
settings I got something like:

31 *+11.2
62 *+7.5
125 *+3.3
250 *[set as 0.0 reference]
500 *-1.9
1L *-2.0
2k *-2.0
4k *-3.6
8k *+2.0
16k *~-1.0

Later...
Ron Capik
--

OK, that gives you settings actually quite close to mine - and to what
the curve suggests. But where are the Q-measurments, oh intelligent
master?

-CC

[Arny Krueger]

"ChrisCoaster" wrote in message

On Jul 22, 6:17 pm, Ron Capik wrote:
ChrisCoasterwrote:
On Jul 22, 1:49 pm, Ron Capik
wrote:

However, "we" can surly measure the 10-band EQ, can
"we" not?

Later...
Ron Capik
--- Hide quoted text -

- Show quoted text -
________________
Expand on that thought please, Ron?

-CC

I'm guessing there must be dozens of programs
out there that can be used to measure the Q of
an equalizer.

One way would be run some pink noise through
the Media Player then set the EQ to some
value and run the same noise through the
Media Player. Take FFTs of both and subtract
the two FFTs.

In a quick experiment using your
settings I got something like:

31 +11.2
62 +7.5
125 +3.3
250 [set as 0.0 reference]
500 -1.9
1L -2.0
2k -2.0
4k -3.6
8k +2.0
16k ~-1.0

Later...
Ron Capik
--

OK, that gives you settings actually quite close to mine
- and to what the curve suggests. But where are the
Q-measurments, oh intelligent master?

The Q of a peaking-type equalizer can approximated by the difference between
the 2 -3 dB points on either side of the peak, divided by the center
frequency.