I can't see no reason for an mass-produced digital amplifier to be more expensive than an ordinary amplifier.

Many people like tube amplifiers which feature very high THD (compared to clean Solid State designs) but still sound "good" (not transparent, but good).

Considering that modern solid state amplifiers are probably the least distortion introducing and most linear stage in any hifi (unless clipping or broken) I really don't see the next revolution in musical transparency coming from developments in any amplification technology.

These amplifiers seem like they could be fairly efficient and offer excellent distortion measurements. Having said that amplifier measurements don't seem to effect sound. Many people like tube amplifiers which feature very high THD (compared to clean Solid State designs) but still sound "good" (not transparent, but good).

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I can't see no reason for an mass-produced digital amplifier to be more expensive than an ordinary amplifier.

They are fairly similar technology to switch mode power supplies, just controlled by their input voltage, which means that they should be able to be produced super cheap.

Class D designs are prone to distortion, chiefly from imperfect power supply regulation and timing error. Since the output voltage of a Class D amplifier is directly proportional to the power supply voltage, any error in that voltage modulates the output voltage. Power supply variations in the amount of current drawn by the amplifier show up in the output as distortion. Instabilities in the supply itself, such as power line ripple, show up at the output as noise or hum. Building a power supply so that voltages remain rock steady in spite of fluctuations in output current is not a trivial task.   

although i may have misunderstood this digital amplification technique.

Class D amplifiers also use two transistors, but instead of amplifying analog signals, which can assume any value, they switch between just two voltage values (in our example) +40 and -40V. One transistor can connect the output to the +40V rail, the other to the -40V.
  Theoretically, neither transistor wastes any power. When one is fully on, all of the power supply voltage is dropped across the speakers and none travels across the transistor. When it's fully off, all of the supply voltage drops across the transistor, but no current flows through it. In both cases, the product of the voltage across the transistor and the current through it is zero.
  Of course, this Class D amplifier can reproduce only binary (two valued) waves. So to use it to amplify analog music signals, those signals have to be converted into a suitable on-off waveform. One way to do this is with pulse-width modulation (PWM). In PWM, the amplitude of an analog input signal serves to control the average percentage of time the transistor spends turned fully on, known as duty-cycle.
    The PWM signal is generated by comparing the analog input signal with a triangle waveform - one that continuously sweeps linearly from a low to a high value and back again. To do that, both signals are fed into an analog comparator whose output is high whenever the analog signal has the higher instantaneous value, and low when the opposite is true.
     The output of this comparator, then, is a waveform that has the information of the original analog signal and yet switched between just two values....In a Class D amplifier this PWM waveform acts as a binary control signal that switches the transistors on and off depending on the amplitude of the analog input. Changing the power supply voltage changes the amount of amplification.
      Of course, what the transistors produce is a higher-power version of this same, switching waveform, which would overheat the speakers dreadfully if allowed to reach them, containing as it does amplified versions of both the original audio input signal and inaudible higher-frequency components arising from the PWM process. So after amplification, that PWM wave has to be passes through a filter that lets lower-frequency signals through while weakening the higher-frequency ones. This the low-pass filter does by smoothing the switching waveform, in effect supressing the rapid changes in the output waveform and leaving only it's average value. At the same time, happily enough, it filters out noises caused by the switching process itself.

i think fast diodes are only really needed in more efficient designs (class AB for instance) where the "speed" of these components related to crossover distortion.

Does anyone have some info on what the different classes (A,B,C,D) means for amplifiers?

It reminds me of descriptions of DSD, and as always there is somthing unclear for me.

If so, why taking the average of the digital waveform, while it is the "primitive" function of it (taking -1 and +1 instead of 0 and +1) that represents the original waveform, and not the average ?