John Curl's Blowtorch preamplifier part III

Status
Not open for further replies.
Member
Joined 2004
Paid Member
Sorry to intrude, at this time, about the mains-filter topic (so long ago :)).
Attached is a AC graph and schematic of the mains-filter as depicted in RN Marsh his US patent 5260862.

There are two questions that I have about the simulation:
1: Why do I not see 20dB of more suppression at 1kHz (maybe I made some wrong assumptions about the mains impedance (or otherwise) and maybe I made some wrong assumption about the load).
2: What about the 56u capacitor (C8), is that not a bit large? and if 56u is needed would this be a suitable capacitor https://nl.mouser.com/datasheet/2/212/F3303_C44E-1104307.pdf

Frans.

Richard can provide details on the circuit but I believe its inspiration is from an analysis of a Tice clock.

First the source and load impedance's are far too simple. The usual source Z is figured as 50 Ohms but that's a convenience to match instruments. A milliOhm source at 60 Hz becomes more like 100 Ohms near 1 MHz. Loads are usually resistive in series with inductive except for any input filters. A rectified load is simply messy.

A 56 uF cap across the line will draw a lot of reactive current heating up all the connections and the wire. You won't be paying for it. I would not recommend it however.

I would use a 10 milliOhm in series with 20 uH in parallel with 100 Ohms as a model for the source just to have something to work with. The resistor for the load is an OK starting point. Realize that the real world will never be that simple.

You could get far more complex with a sine source (with about 2% distortion) and a rectified load. Then look at the spectrum of the voltage and the current with and without the filter.
 
Member
Joined 2004
Paid Member
I presume excursion has a large part to play in this effect, as you can guess from my signature I like the woofer supported wideband approach, what would you suggest is a reasonable crossover frequency in this case?

I have done some analysis of the spectral energy distribution of music. As I remember, except for hip-hop, its centered around 100 Hz. Hip-hop is lower. You can use Audacity for this as I remember. It would be worth the effort to update.
 
Hi Tryphon,
From experience I will disagree with you. In some respects you are right, but you still need a three way system at the very least, actively crossed over will create the illusion you mention. The problem with a two way system is that the doppler effect tears the sonic image apart.
High, anatech,
Once again, I thing I don't disagree with what you said about doppler. But consider the following:
I use at home a big two way system, with a 2" horn and a high slope (acoustic 48db/oct) filter (passive or, now, active). I don't know an other way to reproduce accurately the range from 700 to 20 000 with a single speaker, and to get a similar angle of diffusion (and emissive surface) at the crossing frequency. of course they are "time" aligned at the crossing frequency.
The best system, i heard in my all life. it will be my system for the rest of my life.
My horn is a kind of Jean-Michel LeCleac'h's, totally devoid of those nasal sounds that people believe the appanage of drivers+horns.
*Very* linear after correction with a very good group delay curve.
No risk of doppler in a horn driver: The membrane deplacement is negligible.
One of the reason why I prefer this two way configuration is to avoid any "rupture" in the most sensible frequency area (IE medium and heights) with two devices that move in the same time with a different phase response around the crossing frequency.
Of course I had tried a lot all possible multi way configurations during my (halas) long life. With a lot of speakers, and worked with a lot of various studio monitors, big or little, including JBL, Tannoy, Urei etc ...
The horns present some more advantages against membrane's speakers. Like high efficiency and huge dynamic, good for micro dynamic on percussions, and a more pronounced and constant directivity that excite less the room's resonnances.
 
Last edited:
I have done some analysis of the spectral energy distribution of music. As I remember, except for hip-hop, its centered around 100 Hz. Hip-hop is lower. You can use Audacity for this as I remember. It would be worth the effort to update.

The other way to look at it is how high you can bring the subs (acknowledging their bandwidth and directionality) before you start localizing the sound. If that can get up towards 120-150 Hz, then you're really taking a huge displacement load off your main speakers, which can be realized with a 2-way, if desired.
 
Richard can provide details on the circuit but I believe its inspiration is from an analysis of a Tice clock.

First the source and load impedance's are far too simple. The usual source Z is figured as 50 Ohms but that's a convenience to match instruments. A milliOhm source at 60 Hz becomes more like 100 Ohms near 1 MHz. Loads are usually resistive in series with inductive except for any input filters. A rectified load is simply messy.

A 56 uF cap across the line will draw a lot of reactive current heating up all the connections and the wire. You won't be paying for it. I would not recommend it however.

I would use a 10 milliOhm in series with 20 uH in parallel with 100 Ohms as a model for the source just to have something to work with. The resistor for the load is an OK starting point. Realize that the real world will never be that simple.

You could get far more complex with a sine source (with about 2% distortion) and a rectified load. Then look at the spectrum of the voltage and the current with and without the filter.

I do understand what you are saying but even 1mH in series with 500mOhm it all parallel to 100 Ohm is not coming near to the posted result (in the patent).

In the light of this I want to know why I can not reproduce the patent results, it shows a sine generator with an series impedance (that is not specified) and it does not specify any load conditions. Doing just an AC simulation seems like a good start before making it more complex, also it seems in line with the data specified in the patent.

There are (still) two questions that I have about the simulation:
1: Why do I not see 20dB of more suppression at 1kHz (maybe I made some wrong assumptions about the mains impedance (or otherwise) and maybe I made some wrong assumption about the load).
2: What about the 56u capacitor (C8), is that not a bit large? and if 56u is needed would this be a suitable capacitor https://nl.mouser.com/datasheet/2/212/F3303_C44E-1104307.pdf
 
I don't think a classic two way system can reproduce a convincing audio image if there is any bass happening at the same time.
The Dynaudio Gemini's (2way MTM) are the most detailed speakers that I own. I have had many folks agree with me. The musician friends are the ones who take greater notice of these details. To each their own.
Chris, hopefully I'll show up for the next GTA audio gathering and bring the Gemini's for you to evaluate. They are technically DIY as I assembled them from a Madisound kit.
 
Or just do a Geddes/JBL M2 and run a 15" up to 1kHz. The econowave boys have been doing that for ages.
The JBL D2430K driver is something special. Your marriage seems counter to nature.
My question about Geddes speakers was his use of cheap and poor drivers.
I even had issues of defects with JBL and TAD ones, so I imagine !
And, while I had never listened to one of his speakers assemblies, I don't like the idea of this foam in the sound path to fight against "Homs".
Looks like chemotherapy.

And, please, don't think I'm any kind of a snobbish: I'm currently an active member of the poor's man hall of fame.
 
Last edited:
Tryphon, of course I mostly agree with you, but your opinion on speakers is somewhat different from mine.
First, I have never found any loudspeaker system virtually 'perfect' or even especially close to reality, except in rare circumstances, where the music/voice seems to be exceptionally natural in reproduction for a moment.
It doesn't seem that any approach in speaker design is really fully accurate, but each type of speaker can be 'engineered' to be the best that it can be.
I include:
Horns, both hybrid with direct radiators, and full 3 way horn speakers.
Direct radiators, 2 way to 5 way
Electrostatic speakers, both small and large.
Each has its strong points, but none is radically superior to the other.
In Dave Wilson's case (that you appear critical of) he just does not want to compromise on the materials that he makes his speaker systems with, so they become VERY expensive. Is he sucessful? Barely, but his speakers would normally be priced outside my budget, for the quality that I could live with, and I would find other approaches more appropriate.
Richard's hybrid horn/direct radiator JBL system seems to be the most 'affordable' for a high quality system that can get loud, and be accurate as well. That is what I would have bought if I was not given an expensive pair of Wilson Speakers to use. For example, if I had an extra $30,000 dollars I would buy something other than loudspeakers for my living room, not that they are bad sounding or anything, but there are others with the extra cash who will get great pleasure from their purchase, in how good they can sound, overall.
Electrostatic speakers in many ways sound better, but they just don't have the 'Balls' and invariably sound 'wimpy' to me, even the great big ones that take up most of a room and need big amps like the JC-1 to drive them.
Direct radiators in general sound a bit 'opaque' which I might attribute to 'FM distortion' that is always present to some degree.
Horns are extremely difficult to tame and to interface with other drivers, due to their phase characteristics and geometrical compromises in their horn shape.
etc, etc
There is no free lunch! '-)
 
It doesn't seem that any approach in speaker design is really fully accurate, but each type of speaker can be 'engineered' to be the best that it can be.

Hence my comment, folks seem to cling very strongly to preferences in speaker technology even though they are so different the concept of accuracy becomes meaningless.

My father passed away 17yr. ago but my mother just last month had to let go of the house. I saved the Magnepans that I bought my father decades ago (he loved Strauss waltzes) and I'm going to set them up here to compare to the MET7's.
 
Hence my comment, folks seem to cling very strongly to preferences in speaker technology even though they are so different the concept of accuracy becomes meaningless.

Perhaps in this context this statement from SL makes sense? -- "All accurate speakers will essentially sound the same when listened to in a setup that is appropriate to their specific design".
 
Tryphon, of course I mostly agree with you, but your opinion on speakers is somewhat different from mine.
So do I ;-)
About Wilson, it is just that the guy exasperate-me, using all the tricks of the snake oil sellers and i believe you will agree that his speakers are just outrageously overpriced for the result ?
That said, I would not refuse if I was offered a pair of one of his products.
 
Member
Joined 2004
Paid Member
In the light of this I want to know why I can not reproduce the patent results, it shows a sine generator with an series impedance (that is not specified) and it does not specify any load conditions. Doing just an AC simulation seems like a good start before making it more complex, also it seems in line with the data specified in the patent.

There are (still) two questions that I have about the simulation:
1: Why do I not see 20dB of more suppression at 1kHz (maybe I made some wrong assumptions about the mains impedance (or otherwise) and maybe I made some wrong assumption about the load).
2: What about the 56u capacitor (C8), is that not a bit large? and if 56u is needed would this be a suitable capacitor https://nl.mouser.com/datasheet/2/212/F3303_C44E-1104307.pdf

Fiddling with the simulation it seems the source was 50 Ohms (not realistic at 1 KHz for AC line applications but works for the simulation). For AC applications you need caps that won't self destruct and take everything near them. There are caps specific to that task. The cap you call out should be OK for that task. The longer leads and construction will have higher ESR but should not really degrade the performance. All the caps should be rated for direct AC services. Essentially they should be "X" caps which have internal fusing to disconnect if they fail. The L's and R's around the caps are the parasitics of the caps.
 
When it comes to 'bang for the buck', I agree with Scott that the Met7's really fill that bill almost perfectly. That is what I normally listen to, when listening to fm or tv, which is most of the time. Now these are relatively inexpensive speakers that have a good 'time' response, and a relatively lousy frequency response, cannot go too loud, and have virtually no real lows or extended highs.
This speaker seems to be derived from some of Richard Sequerra's early, very expensive, speakers where the Met 7 component was only the central part of the speaker. He placed a ribbon tweeter on top, and a huge woofer underneath and charged 'big bucks', far more than Dave Wilson for what it cost him to produce them. Still, these speakers give 90+ percent of what typical sound quality is required, and they are easy on the ears, (mostly because they don't have extended highs). There are better speakers out there, but not at the price, (although Scott and I got them wholesale or free) because we personally know Richard Sequerra and are pros in the audio business, but even at some retail price, they were a bargain.
 
I have done some analysis of the spectral energy distribution of music. As I remember, except for hip-hop, its centered around 100 Hz. Hip-hop is lower. You can use Audacity for this as I remember. It would be worth the effort to update.
Thanks, that's interesting and usefully low
The other way to look at it is how high you can bring the subs (acknowledging their bandwidth and directionality) before you start localizing the sound. If that can get up towards 120-150 Hz, then you're really taking a huge displacement load off your main speakers, which can be realized with a 2-way, if desired.
Apparently we need to hear 4 or 5 cycles to identify the pitch by which time, in the average room, the waves will have reflected a number of times at these frequencies making localisation harder
 
Status
Not open for further replies.