Magnatec class d amp sulution

Hi I just wonder if any one have used this class d sulution
http://www.magnatec-uk.co.uk/classd_amps.shtml
from magnatec.
they cost as follows if you have a registred company.I belive,
SDV1015-600 (600Wrms amplifier module) cost £102 each for a batch of 10
SDV1015-600 (600Wrms amplifier module) cost £69 each for a batch of 100

SDV1015-300 (300Wrms amplifier module) cost £71 each for a batch of 10
SDV1015-300 (300Wrms amplifier module) cost £48 each for a batch of 100

So It,s not very low cost.

I have an application note from magnatec with some interesting
circuits that for protection / compresion

but don't know how to attach it here but i can sent it bye e-mail if any one is interested.
 
One reason it is so expensive, I suspect, is that each amplifier is basically two amps bridged together. The audio is output out of phase, while the switching frequency is output in phase so as to cancel at no load. When the input audio is increased, the switching frequency at the speaker increases with the audio.

With a standard Class D configuration, using only one amplifier, the switching signal is at a maximum when the audio is 0. By the way, Tripath uses neither of these methods as far as I can tell, but I think the way Magnatek does it is similar to the way the Texas Instrument's TDA200X series does it.
 
hi.

why do you think its expensive?

i mean compared to say the b&o or the lc audio modules.

full bridge amplifiers do not have to be more expensive than half bridge ones , the lc audio is half bridge and seems much more expensive to me :)

yes please email me the memtioned applications.

and feel free to contact me (again) directly if you have soecific questions regarding class-d amplifiers.

bye k madsen - www.cadaudio.dk

ps. didnt you order the lc audio modules you talked about a couple of months back?
 
These amps are brilliant!

I've been using one of Magnatec's 600W modules in a bass guitar amplifier for onstage use for a little over a year now. It has run faultlessly night after night with some 'robust' handling from the roadies! I run it into 2x Ashdown 600W 4x8" cabs at 4ohm load.

I used to use a Peavey 450W amp that was heavy and expensive, and I will now never go back to analogue amps! The difference in sound with the new Magnatec amp is amazing - in particular the bass end is MUCH punchier and clearer.

I contacted Mr Bacon at Magnatec to discuss options when I bought the unit and I ended up buying the module that is described in the application note you mention, complete with all protection circuitry around it etc. It came mounted onto a little bracket/chassis with speakon and neutrik connectors (nice touch). I also bought the matching power supply.

Compared to the cost of equivalent systems, this was extremely cheap! It is also incredibly small and light, fitting neatly into a 1U 19" rack case, and weighing in at under 3Kgs (better than the Peavey's 26Kgs!).

Call Magnatec now, and get one of these systems to try out - they're absolutely fantastic!

:D
 
Hopelessly outdated? Anything older than a year or so in the electronics industry is outdated, and anything older than about 30secs in the audio industry is the same! However, these amps work - they sound good - they're 'affordable', and they're easy to get hold of. Just because something is old, does that mean it 'shouldn't be built or used'. How about valves? They too are surely 'hopelessly outdated' although they are obviously still very much in demand! Of course, the audio industry is proliferated by opinions - this is just mine.

Also, please note that as I mentioned: I bought ONE of these units, not 10.

You say that other cheaper options are available. Please let me know where since I am yet to find a comparable system that offers better value.

Best regards,

Ed.

P.S> No I am not affiliated with Magnatec, I am just very pleased with their product!
 
Thanks for the info - these look good. Some quick questions though:

1) Any heatsinking required for continuous operation (at maximum power) ?

2) Any plans for a matching switchmode power supply? Weight is a prime concern so a largbe toroid is simply out of the question.

3) Any plans on releasing a higher power version?

Many thanks,

Ed.
 
km: I see my name being mentioned :D I have looked with great interest on the Magnatec page referred to above, it looks neat.
I would say the SDV1015-600 compares more or less with our ZAPpulse 2.2SE.
Our solution costs £123.18 (@1 pcs) compared to Magnatec's £102 (@10 pcs) (quoted from above). So Magnatec is cheaper than our card.

But then our module includes:

1..the output filters, and also
2..an effective short circuit protection. The high temperature shutdown has proven worthless, as it works much too slow in case of a short circuit of the output. Further we also include a
3..DC servo to keep the DC level under 20 mV, so the module is suited for electrostatic speakers like Quad ESL or any Martin Logan model.
4..Our module will drive 2 Ohms loads (And even 1 Ohms loads!) without parallelling 2 modules.
5..You don't need to add the +/-10V supply for the inputs, with a ZAPpulse like you need with the Magnatec's.

All in all i don't think our solution is all that expensive km. At least not when compared with other options. And should you wish to buy 10 or 100 units, i will surely match these prices listed above ;)

Phase Accurate: Did you ever build anything in the real life, or just comment on everybody else's work? I don't think your comment on Magnatec's modules is quite justified. The reason why we or Magnatec or many other's use the feedback directly from the output stage is not, that we are too dim, old or outdated to take it after the filter coil. :D It might be because we have good reason to believe it's a better solution.

Jan-Peter: Very interesting, what is the switching frequency of your UcD modules?

All the best from

Lars
 
Ed,

Ofcourse you need some cooling, all Class-D amplifiers have an efficiency of 90%-95%. For a bass guitar an aluminium plate of 15x15cm will be enough.

At the end of the year we will have available something between 700 to 1000W.

At the moment we have no plans for a SMPS. SMPS are still less reliable as with a traditional toroidel transformer and not to forget the price is much higher.

Regards,

Jan-Peter

www.hypex.nl
 
Jan Peter: Thank You :) My main argument for taking the feedback before the coil (and BTW also to have only a single feedback loop, which results in slightly higher THD measurements) is to keep the time delay in the feedback loop as low as possible. This gives me lower TIM, and better sound quality (my main concern - more important than good THD data). Another thing:
If you take an amp with feedback after the coil, the time delay and also switching frequency will get longer (lower freq) as you add capacitive load on the amplifier. I imagine if you add a few uF across the output, the freq may get low enough to destroy the amp with blind current in the coil .. ? This problem is non existent with amps that take the feedback before the filter.
One last thing is that by letting the filter coil determine the switching frequency, as happens when you take feedback after the coil, you might have trouble syncronizing the module with other modules in a multichannel setup. Please correct me if i'm wrong, there may be some way around this, that i am not aware of ;)

I think both systems have pro's and con's. One solution may be better in some applications, while the other is better in other applications.

BTW: you stuff looks really nice !

All the best

Lars
 
Lars,

Nice we like each other products ;) ;)

We use the total time delay of the amplifer, included the delay in the filter to create the selfoscillating. In such a way the whole systems becomes a liniear gainblock. Because of the feedback behind the outputcoil, the impedance of the outputcoil is in a way removed from the output. And will be below 0.010 Ohm. Because of the feedback behind the outputcoil we don't have a peak in frequency response around the LC frequency.

In never tested several uF at the output of the amp, but several 100nF will not be a problem. The amplifier sees already a big capacitor at the output ;)


In a multichannel setup we create slightly different extra delays in the feedbackloop to create a difference of 35kHz in every Class-D amp. We already did this in a 2-way and 3-way active studio monitor.

Regards,

Jan-Peter

www.hypex.nl
 
Jan-Peter said:
In a multichannel setup we create slightly different extra delays in the feedbackloop to create a difference of 35kHz in every Class-D amp. We already did this in a 2-way and 3-way active studio monitor.
www.hypex.nl
I never do that... On our 2x250W board, difference tones are well below 10uV (you need to sift them out with an FFT to find them).

Admittedly it takes some practice.:D
 
Did you ever build anything in the real life, or just comment on everybody else's work?

I developed a class-d amp 13 years ago, when info on class-d amps AND suitable components were still very scarce. The schematic can be found on this forum.
It didn't use feedback from the filter, so it was a "first-timer" like all the other ones.
Even though it wasn't intended as audio amp it sounded very nice.

If you take an amp with feedback after the coil, the time delay and also switching frequency will get longer (lower freq) as you add capacitive load on the amplifier. I imagine if you add a few uF across the output, the freq may get low enough to destroy the amp with blind current in the coil .. ? This problem is non existent with amps that take the feedback before the filter.

You will not have a deviating switching frequency with carrier-based class-d amps (i.e. PWM) like mine was and the Magnatec also is.
Since you will not have really large capacitive loads in real life (and veeeeeeeery seldom purely capacitive ones !) it will not be a large problem with amps like yours either.

And the delay of the filter is definitely NOT a problem. I made the fatal mistake to use my imagination only over the years in order to find ideas how it could be done. But it is definitely better to use imagination AND maths to come to conclusions for how to do it. In the meantime I do not only know how one could use after-filter feedback takeoff with first-order PWM loops but also high-order delta-sigma loops.
The absolutely easiest solution for feedback takeoff from the filter I came up with, I use to call "the simple tweak". It can be applied to any class-d topology where the feedback signal is going into an inverting integrator. It can be used with other topologies as well but it would then be a little less simple. From the measurements that were made by Stereopile, I assume PS-Audio does something similar.

Regards

Charles
 
Lars Clausen said:
My main argument for taking the feedback before the coil (and BTW also to have only a single feedback loop, which results in slightly higher THD measurements) is to keep the time delay in the feedback loop as low as possible. This gives me lower TIM, and better sound quality (my main concern - more important than good THD data).
The problems you quote are typical of control structures where the post-filter feedback is added as an afterthought instead of being an integral part of the solution.

While it isn't obvious at first to solve the "time delay" problem, it's very amenable to the use of lead compensation. The whole UcD concept revolves around uh... an extreme case of lead compensation.

Thus executed, the sonic tables turn. It is my experience that amplifiers without post-filter correction all have a sense of glassiness in the top-end. This is often confused with tube-like warmth, but is a definite detraction from neutrality/transparency.

TIM is simply a restatement of an amplifier's slew rate capability, and its ability to remain linear when brought close to its slew rate limit. In linear amplifiers, distortion often already starts increasing when you're only getting near the slew rate limit. In class D amplifiers, the mechanisms responsible for this is not present. Therefore, as long as power bandwidth exceeds 20kHz, there is no correlation between slew rate and sound quality. In general, I have little sympathy for the still mythological status of TIM. Still today I get people charging at my desk, waving a copy of Otala's paper, proclaiming they know the source of "solid state sound" now. It's only an intermod measurement, nothing more!
 
Bruno: I would never throw Otala at you, in fact i have never read his papers. However it seems obvious that the 'older' (in microseconds) your feedback signal is, that you attempt to align with the current input signal, the higher the mess in signal transients. Maybe not of any importance when you are measuring response to nice sinewaves in the lab. But when we are talking music, it's a whole different ballgame. And i think that it is also obvious that if you take this feedback signal after the coil, then it is nessescarily a little bit 'older' and has 90 degrees of phase shift compared with the direct connection to the output stage. This 90 degrees at fc which is example 90 kHz, makes the voltage (which is what you use to feedback) delay 90 degrees at 90 kHz, or roughly 3.5 microseconds more delay after the coil than before. But OK i agree if you are using a high order sigma-delta feedback loop, the coil delay would be insignificant, as the 4-5th order feedback filter will probably have much higher delays. I am not convinced this is a better way to go.

Your remark about 'as long as power bandwidth is higher than 20 kHz, there is no correllation between slew rate and sound quality' i guess it is people like you who put MC4558 opamps (dual uA741) in modern CD players, because they can just meet the 20 kHz, and so there is no reason to go for higher slew rate. I will only add, that i don't agree with this kind of engineering.

But let's not start the old discussion about whether mathematical or intuitive engineering is better. ;) Anybody will claim their own way of doing things is the best, and much better than anybody else's.

Phase Accurate: Thank You - that answered my question. ;)