Sloped baffle

You are looking to reduce your time incoherence, is how I would say it. And yes, moving the tweeter closer may have increased your Paradigm's incoherence. But the only way to tell is to have a friend help you swing, quite literally, an arc between where your ear is and the location of where each driver's cone or dome meets its voice coil. Those should lie along the same arc.

Because you must keep the string or tape measure pulled tight, you would find you cannot just hold that string against your ear. I recommend you tape a dowel rod to a camera tripod, to mark your ear's location.

Also, get out your calculator to find out how far you are off axis. However, do not listen for tone balance, but for 'depth', for each instrument and voice to appear more and more whole, right there in front of you. The opposite is the tweeter and woofer becoming audible on their own, audibly separated away from the mid. The mid's tone range must be our reference point for someone's location, because that's the main tone range we hear every day.

It has been proven to very many people's satisfaction that the ear is not as sensitive to variations in frequency response as we would like to believe- not to say a flat response is unimportant. However, this must be true, as we never get to hear 'the best frequency response' from any source in real life, because we are never in 'the perfect spot'.

However, when you do get the Paradigm speakers into the right tape measure position/arc, the sound may be worse, because that is not 'the position' they intended. So again, always trust your ears.

In that case, have your friend tilt your left speaker back and forth while listening to Diana Krall's voice on just that left speaker. But not in mono. Her well-recorded voice is already in mono, because she and her piano were panned to the center, which means she and the piano are equal in left and right channels. You do not want the distractions of left-right information, but only the depth info and to hear a sharper focus on her voice, which one speaker can deliver.

Best,
Roy

Hey Roy.

Surely enough, I have a gazillion questions, but I won't keep asking! Well...not the gazillion anyway :-)

Thanks for the names of the publications worth reading. I am in the process of getting the first AES papers to get started. Being in Argentina doesn't help in sourcing!!

I do want to ask back about two specific comments you made:

1) why do you advise to cross over the subwoofers at 40Hz? Wouldn't the Classic Scanspeak driver listed above, for example, have an easier time if it had to reproduce down to 60 or 80Hz instead of 40?

2) you state "the Acourate approach is not right". But WHY? I'm following you other advise: to understand why? ;-)
Seriously, I realize it is not "completely" right, like with your passive network. But doesn't it get me closer to "right" than a middle of the road, non-time coherent passive XO?

I carefully re-read your paper on the Calypso HD development. I would say I studied it more that just reading. Lots of fantastic info there. I can see myself following your guidelines to build my DIY cabinets (plus what I hope to learn from the books, of course), and to get it mostly right in choosing drivers. But it would be just too arrogant on my part to assume I will be that good with XO design, and if that's the only path then it might become a deal-breaker for me. That would be a pitty!

Hi Lewinskih01,

Thank you for the questions.
1) The Classic woofer would go naturally to ~40Hz, then you would just fade in the subwoofer using its own its built-in crossover. There would be no crossover on the main speakers, which is a good thing. Also, you would then not have as much phase shift above 40Hz, compared to the main speakers needing a sub up at 60 or 80Hz- there you would hear the sub all of the time. This Scanspeak woofer would not have "an easier time" unless you are going to blast your music screaming loud.

2) on the Acourate approach, the first claim on their home page is
"The powerful software enables you
- to measure your audio system."
Yes, as do other measurement programs. None are doing anything wrong, but their measurement techniques do not match what we hear. A user will be misled by the limitations of its measurement techniques, unless he studies in detail the subjects I touch upon in that measurement-letter I wrote to sixmoons. No calculations can be right when they rely upon measurements that are wrong. An analogy is measuring a car's straight-line performance to tell how it corners.

On the other hand, I do know that after each 'good driver' gets a 'good Zobel' from you, a pocket calculator can then design your crossover. You verify its -3dB points on pink noise with spectrum-analyzer software, by measuring each driver up close. It does not matter if your microphone curve is weird, from your mic being so close, because you are only looking for what happens with and without your crossover.

The Acourate home page also claims you can use their software:
"- to display, interpret and process measurement data."
A novice user will not know how data is to be interpreted, compared to what is being heard.

It also claims
"- to establish correction filters for speaker drivers and the listening room"
For any 'correction', the software will be relying upon measurements having large flaws, as I explained in an earlier post. This includes it not being able to measure cabinet-surface reflections around the tweeter, and not being able to measure the floor reflections between you and the speakers in the same way as you perceive them.

Furthermore, no measurement made at your chair will be accurate below 500Hz, because of room reflections from your floor, the sidewalls, the wall between the two speakers, in that order. And since 500Hz is nearly an octave above middle 'C' on the piano, you are not measuring accurately much of the musical range.


I cannot see the need for expensive measurement software that gives inaccurate results, compared to how we hear. You will get far more use out of the analog test-gear I mentioned above, using less-expensive computer software as your spectrum analyzer, such as software sold by PartsExpress.com

Best,
Roy

Roy,

I looked up Zobel circuits on Wiki, but the theory got beyond me. Can you please explain what a Zobel circuit does in the context of speakers.

For example, my speakers have an impedance peak of 20 ohms at the mid/tweeter x-over point. Would a Zobel change the impedance presented to the amp.

At what cost? Less efficiency? Distortion? Does the Zobel introduce something into the circuit that wasn't there before. Nothing good comes at a cost of nothing bad.

Thanks

Bruce

Hi Bruce,

A Zobel circuit for any driver makes its crossover circuit perform more to 'spec'. Zobels can result in a flat impedance curve, making life easier for an amp, but this does not always happen.

The Zobel circuit for any voice coil is just a capacitor and a resistor placed in parallel with the driver, before any crossover is added. It is there to make that driver's impedance curve appear flat to its crossover parts, so that they work as you would want them to, in terms of 'rate of rolloff' and for your actual -3dB crossover frequency.

To determine the values for its cap and resistor, you can use an inaccurate pocket calculator equation, or you can measure the impedance curve of the driver as you try different values. This takes a sinewave signal generator and a good voltmeter. The driver under test is not in its cabinet nor hooked up to its crossover.

There are likely some internet sources for how to hook up the voltmeter and sinewave generator to measure the impedance of the driver + Zobel at each frequency. You can either plot the values on graph paper or in a spreadsheet, or just write them down.

The value at which that impedance levels off is what you then plug into a crossover-parts calculation as 'your driver's impedance'. Despite how carefully you measure, that impedance will be wrong to some degree.

That error happens because your Zobel circuit was used to flatten what you thought was that driver's electrical impedance, but you've been measuring instead its electrical + mechanical impedance(s). Therefore, you must adjust any Zobel to get what you want.

It would not affect your speakers' 20 Ohm peak because that was created by a higher-order crossover. Your speakers may have Zobel circuits built in to their crossovers, perhaps not visible in a schematic, as they may have been 'wrapped in' with the values of other crossover parts, through 'computer modeling' of that crossover.

If you added Zobel circuits to an existing speaker, most all of its crossover parts would then need to be changed. The end result may lower the impedance presented to an amp, but not enough to be of any concern. The speaker can become easier to drive, since the amp could see a more resistive load at all frequencies (= a less 'reactive' load that stores energy).

But then again, using any high-order crossover circuit in that speaker will more than negate this, because these crossovers make their own impedance curve. Smart designers can add more parts to make the final impedance curve look flat to an amplifier and to a magazine reviewer, but that's an illusion, as a complex crossover still lays between the amp and drivers.

The cost per Zobel is 'not much' and there is no loss of efficiency. No penalty at all comes from using Zobel circuits. Distortion is not increased if you use the best parts you can afford. A Zobel is 'all good'.

Best,
Roy