Class D at low volume


Hi,

How do class D amps behave at low volume levels?  My question is general rather than related to a particular amp.  I know there are exceptions, but as a rule, SET’s and class A SS excel at low volume.  What about class D?  Is low volume performance of class D predetermined, all else being equal? Do class D amps have a comfort zone?  Do they distort more at low volume or is it uniform throughout?  For the purpose of this question I am only referring to analog input amps and not the ones that take in PCM (e.g. NAD M2).

The second part of my question is as follows.  I’m interested in some higher end commercial class D amps from the likes of lab.gruppen, powersoft, mc2, XTA, etc. due to their network-ability as in, I can control them, DSP them, and stream digital, all via RJ45, at the same time dispensing with all the extra boxes and cables.  But, they are all of very high power from 100‘s to 1000‘s of watts.  Does this mean that in a domestic setting at low volume they operate much closer to their noise floor or is this different with class D?

Thanks
serge_s
My only caveat in recommending Class D amps over others currently might be in regards to the very highest frequencies, tne ones over 12Khz or so that we perceive as "air" in teh sound. I am 50+ and used to hear fine up to 20khz but now I am more limted, as are most people as they age. So I am not a good one to be able to compare how various amps might perform at the highest frwequencies. If one looks at Class D amp measurements, the highest audible frequencies is the final frontier for the technology to conquer on paper. In practice though I do not hear anything to be audibly deficient or missing though I suspect the latest and greatest Class D technology to have the technical edge from a pure bandwidth perspective. Noise control as well as mentioned. To the extent either might matter in practice that is. I have never done a focused A/B comparison on say how well teh best Class D reproduces things like cymbals and air for example compared to others, but I would expect it to be a reasonable competition as best I can tell.

When I was young and heard clearly up to 20khz, I also tended to be very sensitive to anything that was not going on well at teh higher frequencies.

It would be interesting to do a study of amp preferences based on age factor.
It's my understanding that class D power conversion modules typically use high levels of global feedback. The input signal is constantly being compared to the output signal (thousands of times per second) prior to the output signal being amplified. The output signal is only sent to be amplified once it matches the input signal and any necessary adjustments have been made. This high level of global feedback, which designers of more traditional amp topologies typically try to avoid at all costs, is a mainstay of class D amp design. My theory is that this high reliance on high levels of global feedback is responsible for class D amps' astounding neutrality and their performing so close to the ideal of 'a straight wire with gain'. Also, my theory is that this design results in excellent frequency response at all volume levels.

As I've stated I have no technical training on amplifier design or electronics. The above is based on personal reading about class D amplification combined with any small amount of rational thought and common sense I may possess. I may have this entirely wrong, however, and would welcome comments and thoughts on my theory from those with more technical knowledge on class D amps than myself.
while it is true that class-D amps use feedback to make them work & while you are right in stating that without global negative feedback a class-D power would not work, I don't think it is correct to correlate the use of global negative feedback to a class-D power amp's sonic quality.
The way class-D architecture was invented/designed/formulated, global negative feedback is part of its entity. So, just because you read global negative feedback you shouldn't relate it in the same way as you would to GNFB in a class-A, AB power amp. The class-D architecture needs GNFB while class-A, AB architectures have topologies that can do with little or no GNFB.
The class-D power amp is a continuous-time, discrete-voltage pulse width modulated system. The 1st gen of class-D power amps used (& still use) analog/linear power supplies (like the type you see in class-A, AB amps). And, now I'm observing that the next gen of class-D power amps are using switch-mode power supplies (which are themselves class-D power supplies). I think the audio SMPS has finally developed to a point where it is has a low enough noise floor & can handle large currents in a compact size.

Sonic qualities of a class-D power amp have to do with
* power supply design
* noise attenuation at the final output (the amp binding posts) - correct choice of filter
* managing the switching noise in the power output transistors i.e. reducing the switching noise impact on the analog circuits that form the overall class-D power amp
* routing of noisy & quiet signals

just to name a few items.
Class D (PWM) can be easily constructed without negative feedback at all. NGF improves linearity, bandwidth and output impedance like in any other class of amplification.

While global negative feedback in class A, AB is going over many stages of amplification, that create delay (thus producing TIM distortions), class D amplifiers have only one stage - a modulator that drive output switches (Mosfets). This modulator converts voltage to duty cycle of the output frequency. In simplest case it can be created using ramp generator and comparator but current modules resemble more of Delta-Sigma A/D converters. In short one analog quantity (voltage) is converted to another analog quantity (duty cycle) to end up with voltage again by obtaining average value of duty cycle. It is usually done by common mode choke and capacitors (Zobel Network), leaving about 1% of switching noise on the speaker wires. Amount of output power is controlled simply by setting amplitude of switched DC voltage. Frequency of this remaining noise is too low for the speaker cables to become antenna for electromagnetic coupling, but direct capacitive coupling is still possible.

As for the type of power supply - first generations of class D amp also has SMPS power supplies REF1000 being one example and my Rowland 102 being another. Ice power modules were available from the start in two different varieties - with or without SMPS. These switching supplies used to operate around 50-100kHz to preserve efficiency, but newer designed by Rowland run at 1MHz to make filtering easier (very difficult to design). I would say that good efficiency, very quiet SMPS operating at 1MHz able to deliver close to 1kW is a masterpiece.

One advantage of SMPS powered amplifier is often ability to operate at universal voltage and to tolerate any amount of DC. In fact my amplifier can be supplied by DC only up to 400V. Another advantage is regulation. SMPS have line and load regulation. Amplifier with SMPS keeps composure during power peaks since voltage does not sag, like it often happens in linear power supplies.
I think the audio SMPS has finally developed to a point where it is has a low enough noise floor & can handle large currents in a compact size.
The Acoustic Imagery Atsahs use the Hypex/Ncore SMPS 1200 switch mode power supply and in my system they sound at least as good as similar powered, high quality solid state, Class A and Class A/B amps I have owned. They are dead nuts quiet and I notice absolutely no listener fatigue. When I listen, I sometimes wonder why some are trying to use linear supplies with the NC1200 amp module, when it sounds this good with the stock SMPS.
03-03-15: Kijanki
Class D (PWM) can be easily constructed without negative feedback at all.
I wasn't aware of this. how does the PWM system know that it's tracking the input if there is no feedback? Can you please point me to some references?
but current modules resemble more of Delta-Sigma A/D converters.
so, in current class-D modules, they noise shape the input music signal to move the noise out of band & the resulting pulse train from the delta-sigma block is used to drive the power switches? I suppose that delta-sigma technology makes sense for audio because the ear is very sensitive to distortion & noise? what frequency is the delta-sigma clocked at - also 1MHz like the new gen SMPSs? thanks.