When I am connecting the shield from the Mogami cable to the chassis, what is the best method to do that. Should I strip the cable, de-braid the shield and twist it into a wire long enough to reach the chassis or can a wire be soldered to the shield and then run to chassis or is their a better method than these two options?
Thanks in advance.
Thanks in advance.
There will be much more proficient techs here than me - but I would say best practice would be to keep the shield as close to the termination as possible and the route from cable termination to chassis as short as possible. Regards AJ
Seems...
To me that the input cable shield could most easily be terminated to the XLR body tab, right?
Shielding and ground are an interesting topic, considering that most audio cables have the shield terminated to pin 1 solely at the source end of an XLR cable... Typically components use unshielded input wires internally, but considering the additional RF which is likely to be present in a class D amp with SMPS, shielding the input wiring, as well as keeping it as short as possible, and routed away from the modules, is very likely a good practice.
I am curious though about connecting the shield to ground (chassis as ground, not necessarily house/AC ground though). Since the SMPS600 is grounded to the chassis, I wonder if this could couple hash from the SMPS to the input cable shield?
And, is it really good practice to have pin 1 connect to chassis ground? Could not this couple hash from the grounded SMPS600 back to the source?
Not worried about loops here, as AC ground is lifted in my components, but I do wonder about RF travelling around on shields getting to places I'd woudl rather not have it...
To me that the input cable shield could most easily be terminated to the XLR body tab, right?
Shielding and ground are an interesting topic, considering that most audio cables have the shield terminated to pin 1 solely at the source end of an XLR cable... Typically components use unshielded input wires internally, but considering the additional RF which is likely to be present in a class D amp with SMPS, shielding the input wiring, as well as keeping it as short as possible, and routed away from the modules, is very likely a good practice.
I am curious though about connecting the shield to ground (chassis as ground, not necessarily house/AC ground though). Since the SMPS600 is grounded to the chassis, I wonder if this could couple hash from the SMPS to the input cable shield?
And, is it really good practice to have pin 1 connect to chassis ground? Could not this couple hash from the grounded SMPS600 back to the source?
Not worried about loops here, as AC ground is lifted in my components, but I do wonder about RF travelling around on shields getting to places I'd woudl rather not have it...
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I was commenting on the method of termination. The shield should go to pin 1 of the xlr and then be linked to chassis, sometimes the tag on the xlr may not give the cleanest contact to chassis so a separate ground point could be preferable but ideally close-by.
I understand that chassis should be at signal ground to be effective as a shield, I would hope the SMPS ground wouldn't introduce hash.
I understand that chassis should be at signal ground to be effective as a shield, I would hope the SMPS ground wouldn't introduce hash.
No, the shield on the internal connector wire should go to ground/chassis, and pin one should connect directly to ground/chassis.
Otherwise, any rf coming in on pin 1 would be coupled directly to the internal shield wire of the wire running to the module. Remember the wire running to ground/chassis is an inductor, especially at rf frequencies, making it a big resistor between ground/chassis and the shield.
The rule is that pin 1 *always* connects directly to chassis/ground and nowhere else.
cheers,
Alan
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I've been thinking about feedback and class-D characterization and the practical impossibility of measuring all relevant variables of a given piece of gear, and have a few speculations to share about possible directions forward. Here's the broader context I see. Because an amplifier's component parts are the very things the distort the signal, we have two routes of dealing with this distortion. We can go the simplicity route (L'Amp or SET, etc) to minimize parts-count and optimize their quality to minimize distortion, or we can go the complexity route. Simplicity is problematic because it has no global means of correcting distortions created by the parts it necessarily uses. Complexity is problematic because the more parts in a given circuit, the greater the accumulated distortion at its output. Feedback must therefore be applied.
But feedback across an analogue circuit is a non-perfect solution, particularly in the time-domain where propagation and phase delays and anomalies, not to mention distorting effects of the feedback circuit itself (which remains outside feedback correction), will remain as distortion on the output.
How does one break this logjam of necessary distortion, whether of simple or complex circuits? Why not eschew mere simplicity, retain complexity where useful, but apply global corrective feedback in the digital sphere? Here's where a proper characterization of class D amplification can I think help. Class D, imho, is not merely analogue, and not merely digital, but is both. The pulse output on class D is digital---it is discrete pulses, after all---and given the digitally representative nature of those pulses, one is there offered an access point to apply a global solution at the very source-level that comprises the musical signal, the information-form of that music signal itself.
So, why not this: digitize the output waveform, compare that to a digitized version of the input, extract the difference (= distortion) and apply that difference in negative digital form to predistort the digital pulse sequencing on the class D output. This could be done real-time. With a sufficiently flexible and subtle algorithm, I don't see why in principle this could not be made to work.
Comments?
But feedback across an analogue circuit is a non-perfect solution, particularly in the time-domain where propagation and phase delays and anomalies, not to mention distorting effects of the feedback circuit itself (which remains outside feedback correction), will remain as distortion on the output.
How does one break this logjam of necessary distortion, whether of simple or complex circuits? Why not eschew mere simplicity, retain complexity where useful, but apply global corrective feedback in the digital sphere? Here's where a proper characterization of class D amplification can I think help. Class D, imho, is not merely analogue, and not merely digital, but is both. The pulse output on class D is digital---it is discrete pulses, after all---and given the digitally representative nature of those pulses, one is there offered an access point to apply a global solution at the very source-level that comprises the musical signal, the information-form of that music signal itself.
So, why not this: digitize the output waveform, compare that to a digitized version of the input, extract the difference (= distortion) and apply that difference in negative digital form to predistort the digital pulse sequencing on the class D output. This could be done real-time. With a sufficiently flexible and subtle algorithm, I don't see why in principle this could not be made to work.
Comments?
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As they say...
Rules were meant to be broken:
"The rule is that pin 1 *always* connects directly to chassis/ground and nowhere else."
So, how does this work with components which are not designed this way? I know of a few high end manufacturers who prefer to keep their balanced gear floating: no pin 1 connection to chassis at all, no AC ground connection to anything, pin 1 is signal ground only, and the chassis is left entirely floating as well. It seems to me an nCore amp may work perfectly well in this connection scheme: no pin 1 connection to anything, pins 2 and 3 connected, shield of amp (internal) input cable connected to chassis. AC ground no connection, with class 2 construction. This assumes a true balanced signal for the source.
What I am wondering about, is that if pin 1 in the nCore amp connects to the chassis, and at the source is the signal gound alone, couldn't there be a transfer of RF from the nCore chassis, to the signal ground at the source? My assumption inherent in this question is that the manufacturers who do not ground the chassis to AC or pin 1 do it for a reason: presumably to avoid RF pick up into the signal ground?
Rules were meant to be broken:
"The rule is that pin 1 *always* connects directly to chassis/ground and nowhere else."
So, how does this work with components which are not designed this way? I know of a few high end manufacturers who prefer to keep their balanced gear floating: no pin 1 connection to chassis at all, no AC ground connection to anything, pin 1 is signal ground only, and the chassis is left entirely floating as well. It seems to me an nCore amp may work perfectly well in this connection scheme: no pin 1 connection to anything, pins 2 and 3 connected, shield of amp (internal) input cable connected to chassis. AC ground no connection, with class 2 construction. This assumes a true balanced signal for the source.
What I am wondering about, is that if pin 1 in the nCore amp connects to the chassis, and at the source is the signal gound alone, couldn't there be a transfer of RF from the nCore chassis, to the signal ground at the source? My assumption inherent in this question is that the manufacturers who do not ground the chassis to AC or pin 1 do it for a reason: presumably to avoid RF pick up into the signal ground?
Serengetiplains
Class d is analogue
Read Bruno's papers
And a complex circuit does not mean more open loop distortion. The additional components should be added to improve open loop linearity - ie. reduce open loop distortion, and therefore reduce closed loop distortion.
ncore has LOTS of feedback
Btw, what you describe is zetex/nad class z as used in the m2 and m51
Class d is analogue
Read Bruno's papers
And a complex circuit does not mean more open loop distortion. The additional components should be added to improve open loop linearity - ie. reduce open loop distortion, and therefore reduce closed loop distortion.
ncore has LOTS of feedback
Btw, what you describe is zetex/nad class z as used in the m2 and m51
if you don't want the connection to have an external ground reference, don't connect the input cable shield to pin 1. Equipment still has pin 1 connected for proper operation in other situations.
Saying your equipment will never need a different input configuration is egotistical design and does not interface well with the real world.
Alan
Chris, I'm not going to sidetrack into digital-analogue semantics, but I will say this, and just once to keep from going OT. The 1-bit pulse train (call it what you will) is discrete, which is all I mean by "digital." Because it's discrete, it is a mathematical representation of the music signal and, as such, can be mathematically manipulated at source in a manipulation that includes the entire amplifier---or the entire stereo, if one is adventurous. In such a manipulation (call this feedback if you want), there by definition exists no time-problems applying the manipulation. It's pure. It gets to the very source of the signal itself.
I am not sure that made any sense. Sounds like what you are talking about is calculating some sort of global, static inverse of the transfer function of the amp/system, and applying that to the signal. That has nothing to do with feedback.
This is already ot
If you use a digital feedback (or feedforward) system it's no longer class d
To use feedforward you need to predictably know about the system characteristics under all operating circumstances and component tolerances in order to provide predictive compensation for distortion
And therefore you use feedback
And you can choose to use digital feedback if you wish
Except it probably won't actually be digital - there will be some form of adc element to provide fine control
It's not pure it's just different
If you use a digital feedback (or feedforward) system it's no longer class d
To use feedforward you need to predictably know about the system characteristics under all operating circumstances and component tolerances in order to provide predictive compensation for distortion
And therefore you use feedback
And you can choose to use digital feedback if you wish
Except it probably won't actually be digital - there will be some form of adc element to provide fine control
It's not pure it's just different
Semantics aside, that's it, Julf. A Zetex designer captures what I was thinking in the Zetex thread. He uses almost the same language I used, which confirms I was at least on some realistic track:
The digital circuits then predistort the input to the PWM modulator which drives the actual power output stage to compensate for the timing and amplitude errors in it, such that it has the same performance as the reference DAC and the error signal tends to zero.
The end result is an amplifier with performance at the speaker output pretty much as good as the best DACs available have at the DAC output -- not my opinion here, this has been demonstrated by both measurements and listening tests. In fact, Zetex are also applying just the modulator chip as an 8-channel DAC, with performance up there with the best on the market.
I'm not guessing here about how this works, I designed the low-jitter (picoseconds) clock and PWM output stages that produce this reference PWM DAC output
that sounds like what has been called predistortion, this is a technique that has been used for many years. applying it to other systems (ie digital amps) is nothing new.
later I add:
I know nothing of the Zetex amps, just the description provided, which appears to not convey to me the correct topology. According to Bruno, see below, it is not predistortion.
Alan
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And predistortion only works when the non-linearity is predictable and non-dynamic.
We've had this discussion in this thread before. An approximate diagram of the Zetex amp is given in http://www.hypex.nl/docs/papers/AES124BP.pdf, page 51.
As you can see there is no predistortion, unless you want to call the signal internal to the feedback loop "predistorted", in which case ALL feedback amplifiers use predistortion.
I would be interested to see those measurements. The ones I made are seriously at variance with that claim.
As you can see there is no predistortion, unless you want to call the signal internal to the feedback loop "predistorted", in which case ALL feedback amplifiers use predistortion.
I would be interested to see those measurements. The ones I made are seriously at variance with that claim.
That, however, is true.
Yes, so one would have to come up with some way of tracking real-time, some sort of dynamic, adaptive algorithm, perhaps one that learns on the fly, but in any event one that receives then recalibrates the comparator pulse output to negate distortion products.
Thank you for digging out that reference, Bruno.
Enough speculation for now.
Thank all of you for your answers to my questions so far. I have one more question.
Can the Mogami shield, Nampon wire and the XLR 1Q all be connected to the same chassis screw or should they be connected to different points near each other?
Sorry for the simplistic question, but I am very close to finishing and I want to fire this thing up.
Can the Mogami shield, Nampon wire and the XLR 1Q all be connected to the same chassis screw or should they be connected to different points near each other?
Sorry for the simplistic question, but I am very close to finishing and I want to fire this thing up.
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