| partyjups |
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. |
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| subwo1 |
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. |
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| km |
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? |
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| silvermoon |
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 |
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| phase_accurate |
I know that one should not make any judgement from specs alone. But switching amps that take feedback directly from the output stage shouldn't be built or used. They are hopelessly outdated ! ;)
And there are in fact cheaper solutions around that don't have to be bought in multiples of ten pieces !
Regards
Charles |
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| silvermoon |
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! |
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| silvermoon |
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. |
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| Lars Clausen |
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 |
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| Jan-Peter |
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 |
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| Jan-Peter |
Lars,
Our amps are switching at +/-450kHz.
Just curious;
Why are you claiming feedback before the outputcoil would be better? For what for a reason would this be better?
Your modules looks nice, for shure that you can go so low in impedance.
Regards,
Jan-Peter
www.hypex.nl |
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| Lars Clausen |
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 |
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| Jan-Peter |
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 |
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| Bruno Putzeys |
| quote: | Originally posted by Jan-Peter
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 |
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| phase_accurate |
| quote: | | 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.
| quote: | | 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 |
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| Bruno Putzeys |
| quote: | Originally posted by Lars Clausen
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! |
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| Lars Clausen |
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. ;) |
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| mikeks |
| quote: | Originally posted by Lars Clausen
Bruno: I would never throw Otala at you, in fact i have never read his papers. |
You've missed nothing! :smash: |
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| mikeks |
| ...how do you class-d folks measure loop gain in your systems? |
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| ashok |
Hi Partyjups,
Could you please email me a copy of the application notes.
My email id is ashokm(at)sify.com . Please change the (at) to (@).
Thanks.
Ashok. |
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| phase_accurate |
The most common misconception is that people are believing that the response to a pulse can't immediately be seen at the output of a filter. And this is WRONG.
If it really were like that, mankind wouldn't have flown to the moon, there were no modern aircraft with autopilot and much less of electronic assistance in cars etc etc
Regards
Charles |
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| Bruno Putzeys |
| quote: | Originally posted by mikeks
...how do you class-d folks measure loop gain in your systems? |
It's quite pointless to measure loop gain. Due to the sampled nature of a class D amp, you design for an exactly known and desired loop gain.
First you calculate what loop gain characteristic you want. Then you design the loop filter to do precisely that. Then you build your amp. If the closed-loop gain, THD and/or output impedance aren't what you expected, you've done something wrong. Conversely, if the measurements do match the calculations, loop gain is certain to match the design within a fraction of a dB. |
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| phase_accurate |
Maybe it is the gain of the modulator/switching stage combination that Mike is curious about.
For the classic PWM amp it is simply the ratio of output_supply_voltage/peak_voltage_of_triangular_reference for instance.
Regards
Charles |
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| Bruno Putzeys |
| quote: | Originally posted by phase_accurate
Maybe it is the gain of the modulator/switching stage combination that Mike is curious about.
For the classic PWM amp it is simply the ratio of output_supply_voltage/peak_voltage_of_triangular_reference for instance.
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If the rule is generalised to use dV/dt (at the comparator inputs) and switching frequency instead of pp amplitude of a triangle wave, it goes for non-classic-PWM amps as well. Again it's not something you measure. Mathematical truths don't need physical verification. |
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| mikeks |
| quote: | Originally posted by Bruno Putzeys
It's quite pointless to measure loop gain. Due to the sampled nature of a class D amp, you design for an exactly known and desired loop gain.
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Assuming your closed loop class d system takes an analog input, and outputs a nominal analog signal, then surely the applied feedback is necesarily voltage (shunt) derived, voltage (series) applied (AKA 'voltage' feedback).
Why would you consider it pointless to measure loop gain in such a system...?
One would have thought that in view of the significant phase shifts introduced by the output filter, (for feedback sampled after the filter), judicious measurement, or at least SPICE simulation of the loop transmission path would allow one to compensate for the filter's poles with precisely placed zero's in the loop..
...This should allow feedback to be maximized without compromising stability.... |
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| Bruno Putzeys |
| quote: | Originally posted by mikeks
One would have thought that in view of the significant phase shifts introduced by the output filter, (for feedback sampled after the filter), judicious measurement, or at least SPICE simulation of the loop transmission path would allow one to compensate for the filter's poles with precisely placed zero's in the loop..
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The output filter is an integral part of this mathematical study, as it is an integral part of loop gain. It is surprising that apparently you were assuming I forgot about this.
Poles in the filter become zeros in the error transfer function as a matter of fact!
Ergo:no such thing as explicitly compensating poles by zeros need be done. If it were, it would be impossible to do, as merely changing the load would obliterate the compensation scheme. So, we must thank mathematics for the fact that it happens automatically.
Class D amplifiers with correctly designed loops are extremely insensitive to changes in load or filter values.
I would like to invite you to perform simulations to see this for yourself. You will also appreciate the fact that to try and find the correct transfer function, by measuring actual hardware or even by simulating it, is a hopeless endeavour. The reason why so relatively few amplifiers with good post-filter feedback are made, is precisely because designers are afraid of doing the analysis first.
You will understand therefore that the phrase "to compensate for the filter's poles with precisely placed zero's in the loop" suggests you yourself are rather new to the subject of class D loop control. This is not a problem - I welcome anyone willing to become acquainted with the subject - but to try and teach me (or charles) how to roll a control loop while obviously being unexperienced at it oneself is not expedient to a fruitful exchange.
I hope you will understand my mild irritation and rest assured that no offense is meant. |
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| mikeks |
| quote: | Originally posted by Bruno Putzeys
You will understand therefore that the phrase "to compensate for the filter's poles with precisely placed zero's in the loop" suggests you yourself are rather new to the subject of class D loop control. This is not a problem - I welcome anyone willing to become acquainted with the subject - but to try and teach me (or charles) how to roll a control loop while obviously being unexperienced at it oneself is not expedient to a fruitful exchange.
I hope you will understand my mild irritation and rest assured that no offense is meant. |
Actually, a working analog/digital class-d design was the subject of my dissertation many years ago....(scored 80% overall by the way...which was the highest mark attained on that course for such since it's inception)....
This included an indepth quantitative analysis of control loop (DSP and analog) schemes......including practical verification.....
Indeed, placing singularities in the feedback path to ameliorate those generated by the filter in the foward path has been quantitatively and practically demonstrated in JAES....I just cannot remember the author of the article at the moment.... :scratch: |
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| Bruno Putzeys |
| quote: | Originally posted by mikeks
Indeed, placing singularities in the feedback path to ameliorate those generated by the filter in the foward path has been quantitatively and practically demonstrated in JAES....I just cannot remember the author of the article at the moment.... :scratch: |
Indeed, the degree of complication people are willing to go through while missing the obvious is a constant feature of class D design. Especially published papers tend to bear this out. |
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| mikeks |
Hi Bruno,
A simple low THD design would of course be interesting....
Complexity versus relatively poor linearity is the most discouraging facet of full-range class-d arrangements...
Simpler concoctions may be implemented for sub-woofer applications of course....
In respect of audio power amplifiers, i am a linear class-B/AB person really, but would be interested in your class-d design philosophy....and the sort of linearity obtained by its implementation... |
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| Bruno Putzeys |
| quote: | Originally posted by mikeks
A simple low THD design would of course be interesting....
In respect of audio power amplifiers, i am a linear class-B/AB person really, but would be interested in your class-d design philosophy....and the sort of linearity obtained by its implementation... |
Have a look around the threads that bear the word "UcD" in their title and check out www.hypex.nl who sell modules based on the concept. A spec sheet with plots is available there and will show the smoothest and most load-invariant frequency response you'll have ever seen since long. Effective output impedance is 20mohm+500nH, significantly lower than that of the boucherot networks typically found on linear amps.
Lower THD figures (-110dB) are readily realised, but so far the improvement on sound quality produced by this has proved zip (not a surprise considering loudspeaker distortion), so for the time being there are no plans offer these "improved" modules commercially.
Cheers,
Bruno |
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| mikeks |
| quote: | Originally posted by Bruno Putzeys
....Lower THD figures (-110dB) are readily realised...... |
You'll forgive me for taking this with a vast pinch of salt...
....at what frequency is this figure quoted?
...and of even greater importance, what was the measurement bandwidth?
.....If the measurement bandwidth was the typical 80KHz used for linear amps., and we assume you mean -110db at 1KHz, then this class-d amp. outperforms the majority of linear class-ab designs...
Even assuming that residual carrier signal at the class-d amp's output did not cause the input stages of of the distortion analyzer to slew limit, this figure is unlikely in the extreme.... |
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| Jan-Peter |
Mikeks,
Hereby a measurement of a UcD Class-D amplifier with an extra integrator loop. Measurement bandwith was 22kHz.
This is not implemented in a product, but a test to take a look how low we can get the THD......
Regards,
Jan-Peter
www.hypex.nl |
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| mikeks |
| A schematic of the class-d amp. to which this performance is attributed would not be remiss....? |
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| Bruno Putzeys |
| quote: | Originally posted by mikeks
You'll forgive me for taking this with a vast pinch of salt...
....at what frequency is this figure quoted?
...and of even greater importance, what was the measurement bandwidth?
.....If the measurement bandwidth was the typical 80KHz used for linear amps., and we assume you mean -110db at 1KHz, then this class-d amp. outperforms the majority of linear class-ab designs...
Even assuming that residual carrier signal at the class-d amp's output did not cause the input stages of of the distortion analyzer to slew limit, this figure is unlikely in the extreme.... |
This figure was quoted at 1kHz. Measurement BW is 22kHz, but you will recognise that going to 80kHz would make noise dominate the measurement, but not add any further harmonics to the measurement. In fact, for this reason, the 110dB pertained to THD, not THD+N.
Concerning slew-rate limiting on distortion analyser inputs, you will understand that this would require a very large residual.
The AP analyser will analyse at least up to 200kHz full-scale audio signals, The carrier is at 400kHz. It should not elicit any slew rate problems as long as it is better than 6dB suppressed. It is suppressed by better than 40dB.
AP's do potentially have a problem with the presence of a carrier, in that the autoranging circuits base their decisions on the full-bandwidth signal. At low amplitudes the carrier will force an unnecessarily high gain setting, adversely affecting the device's resolution at low signal levels. The AES17 pre-filter used to measure D/A converters (most of which have large outband noise levels causing similar problems) remedies this.
On the AP2 this problem is not overly dramatic. JP's plot was made without the use of an AES17 filter. |
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| Bruno Putzeys |
| quote: | Originally posted by mikeks
A schematic of this class-d amp. to which this performance is attributed would not be remiss....? |
You can find the basic "UcD" schematic by looking up the patent application. Add 2 orders of integration. You should not find this hard to do. |
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| IVX |
| quote: | | You can find the basic "UcD" schematic by looking up the patent application. Add 2 orders of integration. You should not find this hard to do. |
Bruno,
however basic UcD is sounding better yet? |
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| JohnW |
mikeks,
You seem to doubt that Class D amps can perform technically if not sonically. I’ve posted FFT’s on the UCD180 thread and others, that show the performance of both my Digital and Analogue Class D designs that are limited by the performance of the AP2 and R&S UPD05 test systems.
I’m in the process of designing a new measurement system that should allow me to measure the full performance of my designs.
John |
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| skippy |
Jan-Peter,
You say: "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."
If this is the case, why do all the 300-500W Class D DVD receivers coming out of China have SMPS? Or are you only considering the case of low-volume home build?
Regards,
Skippy. |
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| Jan-Peter |
Skippy,
The volume of this products is >100.000 or even >1.000.000.
I am happy with a traditional transformer...
Regards,
Jan-Peter
www.hypex.nl |
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| Bruno Putzeys |
| quote: | Originally posted by skippy
Jan-Peter,
You say: "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."
If this is the case, why do all the 300-500W Class D DVD receivers coming out of China have SMPS? Or are you only considering the case of low-volume home build?
Regards,
Skippy. |
We've had tons of these over here for analysis. If you run all channels at full power, they shut down within seconds. Some of them will deliver 2 channels of full power for prolonged periods of time, but not all.
Therefore, a 6x50W receiver is not a 300W device, far from it. |
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