B&D wrote in message ...
On 11/18/04 7:47 PM, in article , "Bob Marcus"
wrote:

Sorry - meant to say "the resolving power of a CD Player with a CD
played..." - my bad.

OK, so tell us what you mean by "the resolving power of a CD Player."
It sounds suspiciously like pseudo-technical jargon. Resolution is a
technical term, and every CD player has the same resolution, doesn't
it?

Resolving power - the amount of detail and information successfully
recovered by a CD player without distortions such as jitter.

Still sounds pseudo-technical...

Not pseudo-technical - the format has only 16 bits of data, you are correct,
and the DACs are generally rated to be able to extract more than 16 bits if
presented with them. However, the effective # of bits may not be as high as
16 with a real world player. And of those - through the conversion and
analog stages, the detail that is available is not always presented.

And how do you measure this?

Take the iPod fed with a Apple lossless or AIFF file compared to a NAD C541i
- play a CD such as an early Elvis Costello for an example of a bad
recording. You will be able to hear more detail on the NAD than on the
iPod. Same # of bits in the formats, but the amount of effective detil on
the CDP is higher. Take the becnhmark DAC1 - it will have incrementally
more detail than the NAD.

And how do you know this? Please tell me you are basing this on
something more than sighted comparisons.

bob

On 11/25/04 1:01 AM, in article , "Stewart
Pinkerton" wrote:

Just for the sake of definition, the FR of any amplifier is
defined by both it's own design (something I know nothing about) and
the load connected to it. The latter is out of the amplifiers direct
control. If you wanted to get devious, you could probably screw up the
FR of even the most stable and neutral SS amplifier if you know what
you're doing just by giving it a strategically sadistic load.

Not if it's properly designed to be unconcitionally stable, you can't.
Besides, IME this claimed effect simply doesn't exist. My own amps
certainly have the same FR into a low impedance or capacitive load as
they do into a high impedance speaker, and they go flat down to a
couple of Hz into *any* load. This is basically a nonsense claim.

It is a well known fact that the transient response of most amplifiers will
change depending upon the load - the amount of power available will change
as well. In working with RF generators, we tend to test them into a variety
of loads from 50 Ohms (VSWR 1:1 or thereabouts) to a complete mismatch
(10-inf:1) reactive load and points in between (1.5:1, 2:1. 3:1. 5:1 and so
on). As the VSWR gets towards the higher VSWR's, and the phase changes, you
generally have to back off the power due to some angles giving dissipations
way too high for safe operation and the transient response changes to being
faster or slower (under/over/critically damped response). I cannot see how
Audio amps would be any different. The amount of power available does
change depending upon the resistive load impedance (such as 8 Ohm, vs. 4
Ohm, vs. 2 Ohm), and as the impedance gets reactive, it may change further.

On 25 Nov 2004 16:06:00 GMT, B&D wrote:

On 11/25/04 1:01 AM, in article , "Stewart
Pinkerton" wrote:

Just for the sake of definition, the FR of any amplifier is
defined by both it's own design (something I know nothing about) and
the load connected to it. The latter is out of the amplifiers direct
control. If you wanted to get devious, you could probably screw up the
FR of even the most stable and neutral SS amplifier if you know what
you're doing just by giving it a strategically sadistic load.

Not if it's properly designed to be unconcitionally stable, you can't.
Besides, IME this claimed effect simply doesn't exist. My own amps
certainly have the same FR into a low impedance or capacitive load as
they do into a high impedance speaker, and they go flat down to a
couple of Hz into *any* load. This is basically a nonsense claim.

It is a well known fact that the transient response of most amplifiers will
change depending upon the load - the amount of power available will change
as well. In working with RF generators, we tend to test them into a variety
of loads from 50 Ohms (VSWR 1:1 or thereabouts) to a complete mismatch
(10-inf:1) reactive load and points in between (1.5:1, 2:1. 3:1. 5:1 and so
on). As the VSWR gets towards the higher VSWR's, and the phase changes, you
generally have to back off the power due to some angles giving dissipations
way too high for safe operation and the transient response changes to being
faster or slower (under/over/critically damped response). I cannot see how
Audio amps would be any different. The amount of power available does
change depending upon the resistive load impedance (such as 8 Ohm, vs. 4
Ohm, vs. 2 Ohm), and as the impedance gets reactive, it may change further.

Sure that available watts (as opposed to VA) changes, but the whole
point of an unconditionally stable amp is that, even into extrenme
loads, it remains stable, and hence its FR remains essentially
unchanged.

--

Stewart Pinkerton | Music is Art - Audio is Engineering

"Stewart Pinkerton" wrote in message
...
On 25 Nov 2004 16:06:00 GMT, B&D wrote:

On 11/25/04 1:01 AM, in article , "Stewart
Pinkerton" wrote:

Just for the sake of definition, the FR of any amplifier is
defined by both it's own design (something I know nothing about) and
the load connected to it. The latter is out of the amplifiers direct
control. If you wanted to get devious, you could probably screw up the
FR of even the most stable and neutral SS amplifier if you know what
you're doing just by giving it a strategically sadistic load.

Not if it's properly designed to be unconcitionally stable, you can't.
Besides, IME this claimed effect simply doesn't exist. My own amps
certainly have the same FR into a low impedance or capacitive load as
they do into a high impedance speaker, and they go flat down to a
couple of Hz into *any* load. This is basically a nonsense claim.

It is a well known fact that the transient response of most amplifiers
will
change depending upon the load - the amount of power available will change
as well. In working with RF generators, we tend to test them into a
variety
of loads from 50 Ohms (VSWR 1:1 or thereabouts) to a complete mismatch
(10-inf:1) reactive load and points in between (1.5:1, 2:1. 3:1. 5:1 and
so
on). As the VSWR gets towards the higher VSWR's, and the phase changes,
you
generally have to back off the power due to some angles giving
dissipations
way too high for safe operation and the transient response changes to
being
faster or slower (under/over/critically damped response). I cannot see
how
Audio amps would be any different. The amount of power available does
change depending upon the resistive load impedance (such as 8 Ohm, vs. 4
Ohm, vs. 2 Ohm), and as the impedance gets reactive, it may change
further.

Thanks, B&D.


Sure that available watts (as opposed to VA) changes, but the whole
point of an unconditionally stable amp is that, even into extrenme
loads, it remains stable, and hence its FR remains essentially
unchanged.


Right, now we're finally talking on the same page. You have just loosely
described that as an amp designer you make efforts to minimize a problem
that I have to deal with. If it were a non-issue, I wouldn't be bothering
you with this discussion and you would have had one less strategic design
issue to deal with in your work.

The real question is what can guys that do what you and B&D do, allow guys
that do what I do to make a better reproduction of the AC signal flow in
wire that we're both responsible for accurately reproducing in one form or
another?

Wessel

--

Stewart Pinkerton | Music is Art - Audio is Engineering

Wessel Dirksen wrote:


Sure that available watts (as opposed to VA) changes, but the whole
point of an unconditionally stable amp is that, even into extrenme
loads, it remains stable, and hence its FR remains essentially
unchanged.


Right, now we're finally talking on the same page. You have just loosely
described that as an amp designer you make efforts to minimize a problem
that I have to deal with.

What exactly is the problem, and why do you have to deal with it?

If it were a non-issue, I wouldn't be bothering
you with this discussion and you would have had one less strategic design
issue to deal with in your work.

What is "it"?


The real question is what can guys that do what you and B&D do, allow guys
that do what I do to make a better reproduction of the AC signal flow in
wire that we're both responsible for accurately reproducing in one form or
another?

What exactly do you mean by making a "better reproduction of the AC flow
in wire"?

If I remember correctly, you theorized that reducing series resistance
in inductors used in speaker crossovers can lead to "better" sound. What
you appear to have missed is that the overall design of the speaker
takes the finite resistance of those inductors into account, assuming
the designer is competent. In fact, the designer might have counted on
those inductors having a certain resistance to achieve an overall
optimial response. By tweaking with those inductors, you could very
likely cause additional errors and non-optimal responses. We are not
talking about designing the best inductor here. We are talking about
designing the "optimal" speaker (given the constraints in size, cost,
component specifications, etc.). The best possible inductor may not lead
to the best possible overall response.

Here is an analogy. You have a power amp that uses transistors with a
certain bandwidth. Now, you can replace those transistors with ones with
100 times more bandwidth. But in doing so, you may end up with an
amplifier that oscillates! The designer had taken the bandwidth of the
transistors into consideration when he designed the amp, and he was
depending on those transitors having certain bandwidths. If you replace
those transistors with "better spec'ed" ones, you may jeopardize the
original design and end up with something much inferior.

Wessel Dirksen wrote:

The whole DC resistance thing is important to me and that's what's
fueling my input here. So this reply is about DC resistance in general
as it applies to loudspeaker performance, whether from cable or
inductors. So please consider this a departure from the above. As
such, I appreciate the discussion on this.


I'm curious how others feel about a similar fenomenon with lossy
inductors in the signal path in woofers. It is my personal view
that series DC resistance in the signal path should be
strategically minimized as much as possible. The overall Q factor
of the filter doesn't have to suffer from this if you are in
control of the design process. Once again in the interest of
affording the amplifier maximum EMF control of it's load along
with minimized DC loss along the way.


The DC-resistance of cables/xovers is in series with the voice-coil and thus
will affect the balance between the SPL at resonance(where the resistance is
*much* higher and almost independent of DC-resistance) and the rest of the
transmission range. A high resistance will affect only the higher
frequencies of the bass driver, not the bottom.
Since the transient response is governed by the behaviour at resonance, the
DC-resistance will not make much difference at all.
A higher resistance will always attenuate the higher frequencies so the
bottom seems stronger. There are also simple means of implementing a
*negative* output impedance into the amp, see schematic on my website
http://www.pupazzo.page.ms/
This way a higher DC-resistance can be completely compensated for. Of course
the power loss in a lossy inductor will still occurr, a physical necessity,
but its effect (higher Q at resonance) will disappear. A smaller cabinet
will have a similar effect, so we can adjust the Q to the desired value. The
best transient response will be with a Bessel-characteristic Q=0.56, whereas
the most linear frequency response a Q=0.707 Butterworth.
Now if you replace the inductor with your low DC-resistance coils, you will
have less bass. But for higher levels the core may saturate and create
distortion, if you replaced an air coil with a high DC-resistance. For a
loudspeaker designer it may have been a desired factor to use higher
DC-resistance coils.
If the crossover has a compensation of the voice coil resonance by means of
an R-L-C parallel to the speaker, replacing this coil will almost always
have negative results. The impedance seen by the amp can be too low, the
frequency response will be different from the desired one, and the other
components will be stressed. Apart from less bass.
--
ciao Ban
Bordighera, Italy

"Ban" wrote in message
...
Wessel Dirksen wrote:

The whole DC resistance thing is important to me and that's what's
fueling my input here. So this reply is about DC resistance in general
as it applies to loudspeaker performance, whether from cable or
inductors. So please consider this a departure from the above. As
such, I appreciate the discussion on this.


I'm curious how others feel about a similar fenomenon with lossy
inductors in the signal path in woofers. It is my personal view
that series DC resistance in the signal path should be
strategically minimized as much as possible. The overall Q factor
of the filter doesn't have to suffer from this if you are in
control of the design process. Once again in the interest of
affording the amplifier maximum EMF control of it's load along
with minimized DC loss along the way.


The DC-resistance of cables/xovers is in series with the voice-coil and
thus
will affect the balance between the SPL at resonance(where the resistance
is
*much* higher and almost independent of DC-resistance) and the rest of the
transmission range. A high resistance will affect only the higher
frequencies of the bass driver, not the bottom.

Since the transient response is governed by the behaviour at resonance,
the
DC-resistance will not make much difference at all.

Exactly. It is virtually independant except for slight variation of Qe.

A higher resistance will always attenuate the higher frequencies so the
bottom seems stronger.

Please expound.

There are also simple means of implementing a
*negative* output impedance into the amp, see schematic on my website
http://www.pupazzo.page.ms/
This way a higher DC-resistance can be completely compensated for. Of
course
the power loss in a lossy inductor will still occurr, a physical
necessity,
but its effect (higher Q at resonance) will disappear. A smaller cabinet
will have a similar effect, so we can adjust the Q to the desired value.
The
best transient response will be with a Bessel-characteristic Q=0.56,
whereas
the most linear frequency response a Q=0.707 Butterworth.
Now if you replace the inductor with your low DC-resistance coils, you
will
have less bass. But for higher levels the core may saturate and create
distortion, if you replaced an air coil with a high DC-resistance. For a
loudspeaker designer it may have been a desired factor to use higher
DC-resistance coils.
If the crossover has a compensation of the voice coil resonance by means
of
an R-L-C parallel to the speaker, replacing this coil will almost always
have negative results. The impedance seen by the amp can be too low, the
frequency response will be different from the desired one, and the other
components will be stressed. Apart from less bass.
--
ciao Ban
Bordighera, Italy

Hi Ban,

Thanks for the reply. I took a look at the link. What a cool set of
speakers! I like how you exploited the Manger; the strategy for emission is
of course especially interesting. How high do they go before radiating into
half space; or does it go annular? Is any of the midrange close to true
omni?

Wessel Dirksen wrote:
It is my personal view
that series DC resistance in the signal path should be
strategically minimized as much as possible. The overall Q
factor of the filter doesn't have to suffer from this if you
are in control of the design process. Once again in the
interest of affording the amplifier maximum EMF control of it's
load along with minimized DC loss along the way.


A higher resistance will always attenuate the higher frequencies so
the bottom seems stronger.

Please expound.

The Q-factor rises and gives a boost at the resonance frequency which
happens to be at the lower end of the transmission range. This is because
the resistance attenuates more when the speaker coil has low impedance
view with fixed font
______________
/lossy inductor\
___ ___
o----|___|-UUU-+
DCR 10mH |
|
| __ /|
+---| | |
o------------------|__|-
\|
bass driver
Resonance 40Hz 30ohms
DC resistance 3.6ohms

The diagramm shows a 10mH coil in series to the woofer to roll-off the
higher frequencies. If we have an air-coil, the DC-resistance is high say
1.2ohms. At resonance(40Hz)the signal is attenuated by 0.34dB. At 100Hz a=
nd=20
above the impedance has dropped to 3.6ohms, which will give an attenuati=
on=20
of 2.5dB. The Q-factor has increased from Q=3D0.7 to 0.9


There are also simple means of implementing a
*negative* output impedance into the amp, see schematic on my website
http://www.pupazzo.page.ms/

Hi Ban,

Thanks for the reply. I took a look at the link. What a cool set of
speakers! I like how you exploited the Manger; the strategy for
emission is of course especially interesting. How high do they go
before radiating into half space; or does it go annular? Is any of
the midrange close to true omni?

The vertical radiating angle for treble(16k) is around +/-13=B0, at 1000H=
z and=20
down we have almost a perfect point source. I'm still improving the imagi=
ng=20
by strategically placing absorbent material near the edges. This brought =
the=20
apparent height of a central mono source down to the horizontal plane, an=
d=20
gives now an absolutly natural reproduction.
The big advantage of these speakers is that there is no tonal change betw=
een=20
near- and farfield, thus you can place them very close(3') to the listeni=
ng=20
position in a room with little acoustic treatment or further away(6') in =
a=20
well dampened room.
--=20
ciao Ban
Bordighera, Italy

"Ban" wrote in message ...

The vertical radiating angle for treble(16k) is around +/-13 , at 1000H
z and
down we have almost a perfect point source. I'm still improving the imagi

Duh, this is a reply to my last reply. That would be -pi/12 from pi at
16k which is more, uh, possible. Still very impressive.

"Ban" wrote in message ...
Wessel Dirksen wrote:
It is my personal view
that series DC resistance in the signal path should be
strategically minimized as much as possible. The overall Q
factor of the filter doesn't have to suffer from this if you
are in control of the design process. Once again in the
interest of affording the amplifier maximum EMF control of it's
load along with minimized DC loss along the way.


A higher resistance will always attenuate the higher frequencies so
the bottom seems stronger.

Please expound.

The Q-factor rises and gives a boost at the resonance frequency which
happens to be at the lower end of the transmission range. This is because
the resistance attenuates more when the speaker coil has low impedance
view with fixed font

/lossy inductor\

o----| |-UUU-+
DCR 10mH |
|
| /|
+---| | |
o------------------| |-
\|
bass driver
Resonance 40Hz 30ohms
DC resistance 3.6ohms

The diagramm shows a 10mH coil in series to the woofer to roll-off the
higher frequencies. If we have an air-coil, the DC-resistance is high say
1.2ohms. At resonance(40Hz)the signal is attenuated by 0.34dB. At 100Hz a
nd
above the impedance has dropped to 3.6ohms, which will give an attenuati
on
of 2.5dB. The Q-factor has increased from Q=0.7 to 0.9


Right. This is what I figured you meant. I'll reply below


Thanks for the reply. I took a look at the link. What a cool set of
speakers! I like how you exploited the Manger; the strategy for
emission is of course especially interesting. How high do they go
before radiating into half space; or does it go annular? Is any of
the midrange close to true omni?

The vertical radiating angle for treble(16k) is around +/-13 , at 1000H
z and
down we have almost a perfect point source. I'm still improving the imagi
ng
by strategically placing absorbent material near the edges. This brought
the
apparent height of a central mono source down to the horizontal plane, an
d
gives now an absolutly natural reproduction.
The big advantage of these speakers is that there is no tonal change betw
een
near- and farfield, thus you can place them very close(3') to the listeni
ng
position in a room with little acoustic treatment or further away(6') in
a
well dampened room.

Ban,

Starting with your "point source puppies". Impressive. If I understand
your designations correctly, at 16k there's about pi/12 phase/axis
shift relative to source in less than 4 octaves? This is very gradual
indeed. Geez, it's a true donut only way outside of the audio band!
From the above information and examining the graph and pic's, I would
have guessed near omni radiation would have had to be higher than 1k,
so is this 1k figure meant to exclude narrow band "dirt" from local
diffraction? In that case you can justify braging a higher frequency
figure on your point source limit. I'm curious, there must be some
near-field, direct signal interference issues to deal with in the
crossover region. Have you measured HD and FR compared to the direct
emited signals from the Mangers? Is this significant? Considering the
emission is pure in this frequency range, you can possibly even get
reasonable data on this comparing farfield to stimulus, or even
theory! If any interference has a symmetric pattern, you could
digitally correct it prior to amplification, right? I don't pay
attention much to what is happening in the experimental world, but
being a point source guy myself when it comes to pondering utopia,
this system of yours is as close as I've seen to the perfect
mousetrap. I don't think that there's a plane emitter out there that
can top this (unless it's some 100 meters away and then it's no fun
anymore). Respect.

Back to series DC resistance and electromechanical loudspeaker
transient response. I appreciate the dialog on this because I find
some things a little fuzzy. You have identified the areas that I am
curious about so we're talking on the same page. For the sake of
definition, lets call all cumulative series effects resistive (Rx)
because typical loudspeaker filter influences will be non-reactive
below 50hz. Any changes to parameters made by adding Rx will get a
prime (').

The well known classic formulae for Qe assume Rg=0 and a constant
voltage through Rdc (Rg is very arbitrary for tube amps). Despite
this, it is common practice to add Rx to Rdc (Rdc') for calculation of
Qe'. Obviously this is certainly relevant for the affect Rx has on
Qt'. Also by definition, the ratio of Qe to Qm is important in
determining the affect Rx has on shifting Qt to Qt'. The more a driver
is electrically damped the more a given Rx increases Qt', and vice
versa with mechanical damping. This all describes how Rx changes
driver mechanical output. As such, increases in Rx increase Qt with no
shift in fundamental resonance frequency, whereas reduction of chamber
volume decreases system compliance with an increase in resonance
frequency and Qt.

Now comes driver input which is always electrical. Input signal is
also affected by Rx via changes in input voltage to the driver (this
is what you just described above) As I see it, and as you have
elucidated above, Rx also forms a voltage divider with the driver's
complex load at resonance which include effects of Rdc' (note the
prime). So is Rx part of it's own partial load? Seems very weird but
feasible. In any case, the influence of Rx (due to it's complex load)
can cause the electrical input to the driver to no longer be of a
constant voltage. As a result, the specified parameters (theory) used
to describe the resonant system are no longer entirely comprehensive
to describe the new "alignment" of Qt'.

From impromtu observations from projects, it appears that resonance
impedance corrected systems react less than uncorrected systems to Rx
of reasonable size. I have not done dedicated research on this but I
have plans to do so if you or anyone else here can give a sound
explanation and make it unnecessary. What makes this difficult to
assess is that measured Qt and impulse response will always increase
concurrently so it will be impossible to have an absolute experimental
control to compare to the "filter function" of Rx. An option is
possibly to do some measurements with a currect source or to make two
different chambers, one volume optimized to have identical Qt with Rx
as the other with no Rx. But there will always be lots of room for
confounding data "noise" with this.

Another question is about amplifier performance. Does Rx belong to Rg
or amplifier load or both? I think it would have to be both. Does a
substantial decrease in damping factor play ball with this as well in
the real world with SS amps? Tube amps I know can be prone to this
where it is obvious that the transfer function of the high pass band
is affected.

Wessel

Wessel Dirksen wrote:


Thanks for the reply. I took a look at the link. What a cool set of
speakers! I like how you exploited the Manger; the strategy for
emission is of course especially interesting. How high do they go
before radiating into half space; or does it go annular? Is any of
the midrange close to true omni?

The vertical radiating angle for treble(16k) is around +/-13 , at
1000H z and
down we have almost a perfect point source. I'm still improving the
imagi ng
by strategically placing absorbent material near the edges. This
brought the
apparent height of a central mono source down to the horizontal
plane, an d
gives now an absolutly natural reproduction.
The big advantage of these speakers is that there is no tonal change
betw een
near- and farfield, thus you can place them very close(3') to the
listeni ng
position in a room with little acoustic treatment or further
away(6') in a
well dampened room.

Ban,

Starting with your "point source puppies". Impressive. If I understand
your designations correctly, at 16k there's about pi/12 phase/axis
shift relative to source in less than 4 octaves? This is very gradual
indeed. Geez, it's a true donut only way outside of the audio band!
From the above information and examining the graph and pic's, I would
have guessed near omni radiation would have had to be higher than 1k,
so is this 1k figure meant to exclude narrow band "dirt" from local
diffraction?

A few words of definition:
The directrivity is expressed with the 0° value -6dB down. If we have +/-13°
of vertical radiation this means 42.4% of the surface of a sphere has
coverage. This is because horizontally the surface is much more than on the
poles.

When you look at the pic with the reflector, you can see that the treble
travels aditionally 7cm from the diaphragm to the reflector. There is also a
direct wave diffracted at the edge of the speaker which arrives 0.2ms
earlier and causes lobing of the radiation pattern. I try to fight this with
damping material on a ring around the edge. another point of disturbance is
the horizontal edge of the reflector itself, which I fight with another
ring of natural wool fibres. A third reflection comes from the waves being
reflected by the other speaker 1.2m away, which I suppress with blankets
covering the reflecting parts of the enclosure.

So the directivity is not caused by a single mechanism but several
reflections ranging from 1kHz and up. But all can be explained with the
physics and some raytracing similar to optical effects.
The assiociated comb-filter effects seem to raise the apparent source
height, which is the reason why I researched this a bit.


In that case you can justify braging a higher frequency
figure on your point source limit. I'm curious, there must be some
near-field, direct signal interference issues to deal with in the
crossover region. Have you measured HD and FR compared to the direct
emited signals from the Mangers? Is this significant? Considering the
emission is pure in this frequency range, you can possibly even get
reasonable data on this comparing farfield to stimulus, or even
theory! If any interference has a symmetric pattern, you could
digitally correct it prior to amplification, right? I don't pay
attention much to what is happening in the experimental world, but
being a point source guy myself when it comes to pondering utopia,
this system of yours is as close as I've seen to the perfect
mousetrap. I don't think that there's a plane emitter out there that
can top this (unless it's some 100 meters away and then it's no fun
anymore). Respect.


My crossover frequency is at 140Hz, where the Manger has -6dB rel. SPL. The
filler is a 10" woofer gives around +1dB SPL at 140Hz and from there goes
down with only 6dB/oct. until 2kHz, where I do some other filter to suppress
the cone resonances. The subwoofer downfires to the floor and radiates
around +2dB at 140Hz. The phase differences will give a 0dB reading when
summed up at the listening position. The radiation angle is 180° into the
upper half of the floor, so the point source has moved down 1m from ear
height to the floor. The reason is to suppress the otherwise unavoidable
floor reflection, at least the first mode at 5ms delay(null at 100Hz).
I am in the moment experimenting adding a negative signal to the subwoofer
for the higher modes of the floor reflection, it seems to be promising.
All the parameters like individual delays of each way and eq settings are
realized with the processor unit, can be stored and compared to other
settings.

I chose through scientific simulations and experimental measurements the
most homogen and smooth transition in ear-height from near- to farfield, but
every half a year I come up with something better (hopefully).

Back to series DC resistance and electromechanical loudspeaker
transient response. I appreciate the dialog on this because I find
some things a little fuzzy. You have identified the areas that I am
curious about so we're talking on the same page. For the sake of
definition, lets call all cumulative series effects resistive (Rx)
because typical loudspeaker filter influences will be non-reactive
below 50hz. Any changes to parameters made by adding Rx will get a
prime (').

The well known classic formulae for Qe assume Rg=0 and a constant
voltage through Rdc (Rg is very arbitrary for tube amps). Despite
this, it is common practice to add Rx to Rdc (Rdc') for calculation of
Qe'. Obviously this is certainly relevant for the affect Rx has on
Qt'. Also by definition, the ratio of Qe to Qm is important in
determining the affect Rx has on shifting Qt to Qt'. The more a driver
is electrically damped the more a given Rx increases Qt', and vice
versa with mechanical damping. This all describes how Rx changes
driver mechanical output. As such, increases in Rx increase Qt with no
shift in fundamental resonance frequency, whereas reduction of chamber
volume decreases system compliance with an increase in resonance
frequency and Qt.


Right, amplifier output resistance Rg and Rx from cables and Xover add up to
a single series resistor and determine the voltage divider with Re of the
speaker at the speaker clamps. The box volume will determine the pole
frequency of the highpass filter. Damping material might increase the
apparent volume by as much as 20% and at the same time reduce the Q factor
because of reducing Qm .

Now comes driver input which is always electrical. Input signal is
also affected by Rx via changes in input voltage to the driver (this
is what you just described above) As I see it, and as you have
elucidated above, Rx also forms a voltage divider with the driver's
complex load at resonance which include effects of Rdc' (note the
prime). So is Rx part of it's own partial load? Seems very weird but
feasible. In any case, the influence of Rx (due to it's complex load)
can cause the electrical input to the driver to no longer be of a
constant voltage. As a result, the specified parameters (theory) used
to describe the resonant system are no longer entirely comprehensive
to describe the new "alignment" of Qt'.

From impromtu observations from projects, it appears that resonance
impedance corrected systems react less than uncorrected systems to Rx
of reasonable size. I have not done dedicated research on this but I
have plans to do so if you or anyone else here can give a sound
explanation and make it unnecessary. What makes this difficult to
assess is that measured Qt and impulse response will always increase
concurrently so it will be impossible to have an absolute experimental
control to compare to the "filter function" of Rx. An option is
possibly to do some measurements with a currect source or to make two
different chambers, one volume optimized to have identical Qt with Rx
as the other with no Rx. But there will always be lots of room for
confounding data "noise" with this.

Another question is about amplifier performance. Does Rx belong to Rg
or amplifier load or both? I think it would have to be both. Does a
substantial decrease in damping factor play ball with this as well in
the real world with SS amps? Tube amps I know can be prone to this
where it is obvious that the transfer function of the high pass band
is affected.

Wessel

Nice to read your speculations. It really is a completly scientific
research, no magic at all. Everything can be explained with analysing the
refections in direction and arrival time. Unfortunately audio has such a bad
image in engineering circles because of those quacks, that most of the
"serious" scientists avoid it. Such there has not been the same progress
made as in comparable disciplines.

--
ciao Ban
Bordighera, Italy

On 16 Nov 2004 00:51:14 GMT, (Len Moskowitz)
wrote:

There are some reviewers' opinions that I've come to trust, and others
I've learned to ignore.

Ain't it the truth? Two relevant stories.

1. I know a record reviewer whose musical taste I respect and whose
comments about music I value. However, I've learned to invert his
comments about recording/sound quality as his preferences here are the
opposite of mine. Thus, his comments are still of use. ;-)

2. I got my first reviewing job (concerts/recordings) when a new
publication which attempted to cover every event in NYC. They
offered, in addition to the featured reviews, a lot of unsigned
capsule reviews in about 3 column-inches each. I wrote to the editor
that the value of even these reviews was diminished without the
reference value of a familiar by-line since the reader could not
assess the context of the opinion.

Kal