Working the crossover for better impedance

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Oh probably, I only listened till english anyway. :p

So I listened to Magic Carpet.

The vocals part that I understood clearly were when "Well, you don't know what we can find" part comes in, and mostly when the music almost stops, and "Close your eyes girl, look inside girl" part comes. Other than that, it was really hard for me to make of it.

Imaging is good on my speakers, I could all 3 different instruments being played at once, Left, right, and center. So something to appreciate :p
 
With all respect, but the crossover networks shown here are far from optimised i think. I saw a big current loss at the woofer, and the response and impendance also did not look good. So i took my take on it (starting from Douglas Blake last effort). I think this is way more optimised, and the drivers are phase aligned now in the crossover. And i have to say, these ain't easy drivers to get right in the crossover... You can temper the top frequencies by rasing the R4 value if you think that is needed
 

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With all respect, but the crossover networks shown here are far from optimised i think. I saw a big current loss at the woofer, and the response and impendance also did not look good.

Interesting ... a 16 ohm design, not bad.

However; There as no "big current loss" in my last design ... The woofers were in parallel giving a 4 ohm impedance. That means that each woofer was carrying 1/2 of the total current... 2 amps plus 2 amps == 4 amps, as the current graphs below show you.

In your design, with the woofers in series, both woofers carry all the current but the voltage is divided between them.

Your current graph lines up nicely but it is false to claim current loss in my design.

Also, please note that a 16 ohm design, while perfectly valid, will produce a somewhat reduced output... your graph is showing 100db, mine about 106. This is because 16 ohms draws 1/4 the current of a 4 ohm design... thus producing only 1/4 the power. In this case that's not necessarily a bad thing since Denon specs 6 ohms as a minimum.
 

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Interesting ... a 16 ohm design, not bad.

However; There as no "big current loss" in my last design ... The woofers were in parallel giving a 4 ohm impedance. That means that each woofer was carrying 1/2 of the total current... 2 amps plus 2 amps == 4 amps, as the current graphs below show you.

In your design, with the woofers in series, both woofers carry all the current but the voltage is divided between them.

Your current graph lines up nicely but it is false to claim current loss in my design.

Also, please note that a 16 ohm design, while perfectly valid, will produce a somewhat reduced output... your graph is showing 100db, mine about 106. This is because 16 ohms draws 1/4 the current of a 4 ohm design... thus producing only 1/4 the power. In this case that's not necessarily a bad thing since Denon specs 6 ohms as a minimum.

Thanks for the video on compression, useful but a little verbose and poor articulation, a trend today, minimal information, but from someone privileged enough to have access to it.

I am astounded at 20% distortion, and if this is from masters, my criticism of FM quality on another thread, (which Alan Shaw of Harbeth has stated as up to 10%), becomes engulfed in this. And I have been sitting there worrying about my system.

Bill Woodman of ATC stated that using woofers in series was, paraphrasing, actually riddled with problems.

This compression business serves to remove one of the major forms in musical expression, thus diluting attempts to communicate.

Several years ago KEF decided to attempt 4 ohm speakers because of the increased sensitivity, and they were tightly controlled Z wise.
 
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Interesting ... a 16 ohm design, not bad.

However; There as no "big current loss" in my last design ... The woofers were in parallel giving a 4 ohm impedance. That means that each woofer was carrying 1/2 of the total current... 2 amps plus 2 amps == 4 amps, as the current graphs below show you.

In your design, with the woofers in series, both woofers carry all the current but the voltage is divided between them.

Your current graph lines up nicely but it is false to claim current loss in my design.

Also, please note that a 16 ohm design, while perfectly valid, will produce a somewhat reduced output... your graph is showing 100db, mine about 106. This is because 16 ohms draws 1/4 the current of a 4 ohm design... thus producing only 1/4 the power. In this case that's not necessarily a bad thing since Denon specs 6 ohms as a minimum.

If 16 ohms limits the max SPL attainable, this is definitely something that I'd like addressed if possible. Showing off my system needs to go loud, people (my friends :D) perceive louder as better.

I am no expert in any of this, whatever must be done to get the most out of these two driver set would require all your expertise!

All I can do is measure my current setup and report! :D
 
I am astounded at 20% distortion, and if this is from masters, my criticism of FM quality on another thread, (which Alan Shaw of Harbeth has stated as up to 10%), becomes engulfed in this. And I have been sitting there worrying about my system.

Like it or not, we are all slaves to our sources. The best we can hope to achieve is to faithfully reproduce the information on our source recordings. Our task is to choose and set up equipment that does this for us.

Aside from this single assurance, there is literally nothing we can do about it. A bad recording is a bad recording and nothing done to our systems is going to fix that... well, except in the rare cases where we actually have access to the source recordings themselves.

With respect to FM radio ... consider the 10% inherent in the modulation format plus the 20% (or more) in the source recordings ... then whatever is contributed by your amps and speakers... You are likely listening to 50% or more distortion without even realising it.
 
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If 16 ohms limits the max SPL attainable, this is definitely something that I'd like addressed if possible. Showing off my system needs to go loud, people (my friends :D) perceive louder as better.

Okay... that's not true, but if that's what you are stuck with...

Our ears perceive sound differently at different levels. They have evolved to facilitate communication with others. Our most sensitive and detailed hearing is thus at "conversational" levels. That is roughly the loudness of normal speech. Below that we lose information due to background noise and the threshold of our hearing. Above that sensory overload sets in and at some point it just becomes a giant loud roar.

If you really want to know what your speakers can do, set the volume so that you can talk with your friends in a slightly elevated voice... then sit back in the sweet spot and listen. A fine example of the details that reveals was in "Magic Carpet Ride"... riveted symbols, a fine detail that is lost to either too quiet or too loud playback. Also, in that song, most people hear 3 instruments... organ, bass and guitar... but there are actually 2 guitars, bass, drums, organ and vocals.

Another factor in this is XMax ... the limits of speaker cone movement. As you approach that with high volumes, the cone is less and less able to respond to small changes. It's basically working as a compressor, pushing only the big details...

All I can do is measure my current setup and report! :D

The first task, is to get busy taking measurements in REW, from your sweet spot, to find the best speaker positions. I'm looking forward to seeing your results as they will give me good information for the next iteration of this process...
 
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If 16 ohms limits the max SPL attainable, this is definitely something that I'd like addressed if possible. Showing off my system needs to go loud, people (my friends :D) perceive louder as better.

I am no expert in any of this, whatever must be done to get the most out of these two driver set would require all your expertise!

All I can do is measure my current setup and report! :D

The speaker is still capable to go as loud as with 4R, It just need more watts to do that. So that is an easy fix. In series it will go to low in impendance (down to 3R) for your amp to drive and and you will blow up your amp when trying if i hear your amp can only handle 6R and above. So better use it like this in series.

Also, your tweeter can't keep up with the woofers in series as it's efficiency is to low for that on the crossover point. And if you push it further, you get into the upper resonance zone of the woofer, wich is not good at all...
 
Like it or not, we are all slaves to our sources. The best we can hope to achieve is to faithfully reproduce the information on our source recordings. Our task is to choose and set up equipment that does this for us.

Aside from this single assurance, there is literally nothing we can do about it. A bad recording is a bad recording and nothing done to our systems is going to fix that... well, except in the rare cases where we actually have access to the source recordings themselves.

With respect to FM radio ... consider the 10% inherent in the modulation format plus the 20% (or more) in the source recordings ... then whatever is contributed by your amps and speakers... You are likely listening to 50% or more distortion without even realising it.

I was aware of the first and second points, and made just that to my manager at the BBC in '72, we cannot undo the embedding.

I am also aware of the third which I think was implicit in my last post, but the degree of distortion - 20% was to me astounding, and a disgrace.

Even in '72 we were aligning Leevers-Rich tape machines and getting distortion figures of less than 1%, -52dB comes to mind.
 
I am also aware of the third which I think was implicit in my last post, but the degree of distortion - 20% was to me astounding, and a disgrace.

A lot of the distortion in FM comes from the 19khz pilot signal used to decode stereo. They had to limit bandwidth to include it and they had to multiplex sidebands to make it work.

Most recently the high distortion figures appear to have resulted from recording engineers misunderstanding the new digital formats and failing to adjust their recording practices accordingly.

A good hint of this is that most CD players now put out close to 3 volts peak signals. If they had adjusted properly we could have had very high dynamic range recordings at -15db averaging about the standard .5 volts out that were better than vinyl in almost every way. But these guys kept shooting for the 0 VU mark instead of simply recalibrating their meters as shown in the video.

I find it deeply ironic that a new format (PCM Digital) that promised both better frequency response and better dynamic range ended up producing some of the worst recordings ever.
 
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The speaker is still capable to go as loud as with 4R, It just need more watts to do that. So that is an easy fix. In series it will go to low in impendance (down to 3R) for your amp to drive and and you will blow up your amp when trying if i hear your amp can only handle 6R and above. So better use it like this in series.

Also, your tweeter can't keep up with the woofers in series as it's efficiency is to low for that on the crossover point. And if you push it further, you get into the upper resonance zone of the woofer, wich is not good at all...

The "just need more watts" part is part of the problem here. These speakers are in a 5.1 setup and the balance between these fronts and the satellites also needs to be maintained. Plus I doubt our friend is going to fork out for a new $1500 AVR when he's already baulked at a $300 Crown amp.

As an aside... he doesn't actually need more watts for 16 ohms... he needs more voltage which will still result in lower wattage.

There's nothing wrong with your design. It looks okay to me and playing with it here I was unable to find any big improvements. But please keep in mind that our friend is learning from us as he goes, so the more concise information we can give him the better.
 
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Time for some visuals!

Placement of everything described.

Speakers are in the same spot as they are placed throughout which is their actual position aswell. 3 ft from the side walls, about 2 feet out from the back wall (back of speaker), and 6 feet apart. My listening position is about 9-10ft away from the speakers. I know this is not ideal. But it has to do for now.

Speaker measurements are done at 1 meter, mic pointed between the tweeter and the upper midwoofer. 90degree Calibration file for UMIK 1 was used. Everything has Psychoacoustic smoothing.

RIGHT SPEAKER @ 1 METER

NO EQ

mDC7AXR.jpg


This is NO EQ VS EQ with -6db at 16khz done through my AVR, which I commented saying made a huge difference earlier.

Oo0Vbam.jpg


DISTORTION OF RIGHT SPEAKER.

rre9ZMZ.jpg




LEFT SPEAKER @ 1 METER

Red is NO eq, Green is with AVR EQ @ -6db down 16khz.


ZPkv6kc.jpg





DISTORTION OF LEFT SPEAKER


TT2NlJr.jpg


The left speaker seems to have some distortion around the 100hz range which didn't show up in the rights....strange?

Measurements at my LISTENING POSITION , Both speakers playing, 9 FT AWAY, MIC pointed straight UP, non 90 degree calibration file used from my UMIK.

NO EQ vs EQ (-6db at 16khz)

Rw78UAe.jpg



Distortion at LISTENING POSITION


H7bGZ3D.jpg




Woah, I had a blast!!!!:):):):)
 
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But please keep in mind that our friend is learning from us as he goes, so the more concise information we can give him the better.

That last one is true, that's why i started my design, and give the file with it so he can play with it. At the end it's him who need to decide what is best, we can only give him options and opinions and teach him how we work and what we think of when we decide how to do a design. This is no competition at all for me... And even if i have some experience with passive crossover, i still learn a lot in this kind of discussions.
 
and to Kustomize123: that peak on top you can eq in a passive crossover with an inductor and a resistor parallel as like i do in my version. The inductor is the lowest point where the filter start to work, and the resistor is the attenuation of the filter. Try to fix this in xsim with your response file.
 
and to Kustomize123: that peak on top you can eq in a passive crossover with an inductor and a resistor parallel as like i do in my version. The inductor is the lowest point where the filter start to work, and the resistor is the attenuation of the filter. Try to fix this in xsim with your response file.

Fair enough. Thankyou!

But since I am wanting to learn aswell, is there a reason that Zaph himself didn't do this?

Also going off topic.

I found some measurements of the B&W CM9. The first (purple) is the CM9.
It seems like they are not flat in the entire range, and when I heard them, I really liked what they were sounding like. Is it possible I just don't like a flat FR? And maybe I need to work something like the CM's being shown here?

B&W CM9 Speaker System HT Labs Measures | Sound & Vision

210bwspeak.meas.jpg
 
Speakers are in the same spot as they are placed throughout which is their actual position aswell. 3 ft from the side walls, about 2 feet out from the back wall (back of speaker), and 6 feet apart. My listening position is about 9-10ft away from the speakers. I know this is not ideal. But it has to do for now.

Can you do one more test for me?

Set the mic at your listening position, pointed level and directly at the center point between speakers, roughly at ear level. Also, cut your levels back a bit... try to get a reading at about 35 to 40 db.

The umic isn't totally omni-directional. It tends to favour it's nose over it's sides, just a little bit. Pointing up you're getting ceiling echo... lets see if we can eliminate some of that and get a better picture of what you are hearing.

If I get the result I'm expecting, I may have a painfully easy solution for you.
 
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Can you do one more test for me?

Set the mic at your listening position, pointed level and directly at the center point between speakers, roughly at ear level. Also, cut your levels back a bit... try to get a reading at about 35 to 40 db.

The umic isn't totally omni-directional. It tends to favour it's nose over it's sides, just a little bit. Pointing up you're getting ceiling echo... lets see if we can eliminate some of that and get a better picture of what you are hearing.

If I get the result I'm expecting, I may have a painfully easy solution for you.

So you mean my mic needs to be at my listening position pointing between the 2 left and right speakers at ear level? That woild mean right where my center channel is, right?
 
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