04-17-12: Last_lemming
...............
What im ultimately interested in when it comes to how "D" amps work is if there is a loss of information like some say when comparing Vinyl and CD players. The Class D amps work a the principle of an on/off nature similar to how cd players work with 0's and 1's. My understanding is that class D amps use PWM (I think I have the acronym right) thus it would seem their sound is "interpreted" if you will just like CD's have to "interpret" between the bits. Is my thinking right on this or am I off base? Im I losing audio information in a digital amp? My understanding of the CI D200's is that the switching is analog, but the on/off concept still remains.
.............
04-17-12: Mapman
............
I'd say it is accurate to say that the sound of any amp (Class D or otherwise) is "interpreted". The mechanisms used by different types of amps vary however.
................
Since no amplifier is perfect, I would say yes, however, again same true of ANY amp design.
Class D amp technology is relatively new and innovative.
ROTFLMAO!!! Last_lemming, Mapman thanks much for providing the laughs today at lunch time......
This is a classic case of the bling leading the lame....
Yes, class-D amplifiers do use a principle called Pulse Width Modulation or PWM. Class-D amplifiers are, what one would classify, discrete-time systems. They are essentially a mix of both analog & digital hence "discrete-time" as opposed to being purely digital wherein the signal/data would be digital from start(input) to end(output). An every day example that touches our lives would be a digital signal processor in our smart-phones - digital data in, digital data out, then, a D/A conversion & we get brightness modulated on the LCD screen/phone call translated to voice/keypad push converted to user-asked-for action, etc.
The technology for class-D audio power amplifiers is new to the field of audio but it is a very old technology overall. Class-D amplifiers are basically modified switch-mode power supplies (SMPSs) that are appropriately modified to modulate a music signal (rather than track a reference voltage). As many of you already know, you home desktop computer & your laptops extensively use SMPS. That big metal perforated box into which the power cable connects on your home desktop computer is a large SMPS box. SMPS power supplies have been in existence since the late 1960s & early 1970s. Back then SMPSs had very low bandwidth in the 100s of KHz. Recently, with the advancement of technology, SMPSs can have upto 6MHz of bandwidth. And, they do if you look at National Semicondutor's, Maxim's, Analog Devices' catalogs. SMPSs are now extensively used in smartphones today - there must be atleast 6 SMPSs in a Apple iPhone or a Samsung Galaxy S2, etc.
So, the technology is old but the advancement lies in making the noise performance very, very good for audio applications because the human ear is very sensitive to noise in the presence region (1KHz-5KHz).
In a Class-D power amplifier the music signal is an input voltage reference (moving reference, of course) & there is an analog filter that averages this input signal. This average is compared against an internally generated ramp signal. When the ramp signal is above the average signal, one output power transistor is on while the other is off. When the ramp signal is lower than the average signal the other power transistor is on & the 1st power transistor is off. So, now you can see that the time that the on transistor is on varies each time - it depends on how long the ramp signal is above the average signal. This is where you get the pulse width modulation (PWM). The 'pulse' being how long the transistor is on.
Needless to say one power transistor is P-type supplying current into the load & the other power transistor is N-type pulling/sinking current from the load. After the output of the power transistors there is an analog filter to cut down the spurs that are created by switching transistors. If these spurs were not cut down they would create a very noisy output & totally destroy your listening pleasure. The other equally important reason for filtering is that the on/off pulses of the power transistors do not resemble in any way shape or form the analog music signal. If you filter/average these pulses then the averaged signal does accurately represent the input music signal.
So, you can see where the "digital" nature of the class-D amplifier comes in - power transistor either fully on or fully off. If you look at power transistor classification, this action is categorized as class-D (we all are very familiar with class-A & class-AB power amplifiers which are prolific in the audio market). You can also see where the analog nature of the class-D amplifier comes in - the analog filter averaging the music signal. The 2 systems are meshed together as a whole hence class-D amplifiers are not fully digital & they are not fully analog. They are discrete-time meaning that at specific time spots a certain action takes place (one power transistor turns on & the other turns off). But if you look at time on a continuous basis you see that they analog filter has a continuous-time analog output voltage. So, this system is analog & digital all at the same time.
So, the analog filtering post power transistors is an interpolation & there is some loss of signal as the filter smooths out the on/off pulses but the key here is that if the on/off pulses are, say, 10X faster than the highest frequency audio signal (20KHz) then, music signal is oversampled high enough that there is no loss of information (per the Nyquist criteria) & you have sufficient number of data points to reconstruct the analog music signal (per Shannon's theorem).
So, if the class-D amp is designed correctly, information-wise you should not be losing any information. And, BTW, neither do CD players lose any information!
Hope that this helps....