[Wiljen]

I've been reading and haven't found anything to suggest that frequencies this low are being used, but would it be possible with Class D to match the PWM with the PCM of the original recording?  It makes sense to my little brain that making the pulse width of the amplifier to be either the same width of original A/D or a multiple thereof would be the best possible mechanism as it would mean each block that was originally created during the recording would be reproduced as a single unit by the amp.
 
What am I missing?

 

[pfzar]

<YES>  
 
I re-read your post.  Its a bit clearer to me now.  So you are asking if you had a 16/44.1 signal you would want the amp to Switch at 44.1?  
 
 
read this. 
 
http://www.soundandvision.com/content/class-act-amplifier-classes-explained

 

[Wiljen]

Yep you got it.  I was thinking either 44.1 or a multiple of 44.1 so each of the original segments wouldn't be divided as it was amplified.  I am reading the article now.

 

[Wiljen]

Ok, I read through the article but it still doesn't explain how the pwm frequency is decided upon when designing a class D system.  I think that is the part I am looking for.

 

[spruce music]

There are some amps that basically do the DA conversion in the output stage at some multiple of sampling like Tripath based amps. In general most class D doesn't as there is no advantage if they did. Usually they pick switching rates high enough a simple output filter will filter out switching noise while still passing the amplified analog signal. Maybe 100 khz or a bit better. Some do switch at higher rates.

 

[pfzar]

A bit more to read.  
 
http://www.analog.com/en/analog-dialogue/articles/class-d-audio-amplifiers.html
 
https://www.google.com/patents/US20130223652
 
http://www.ti.com/lit/ds/symlink/tas5756m.pdf
 
The TI part is using the I2S clock to generate its switching freq.

 

[castleofargh]

  I've been reading and haven't found anything to suggest that frequencies this low are being used, but would it be possible with Class D to match the PWM with the PCM of the original recording?  It makes sense to my little brain that making the pulse width of the amplifier to be either the same width of original A/D or a multiple thereof would be the best possible mechanism as it would mean each block that was originally created during the recording would be reproduced as a single unit by the amp.
 
What am I missing?

how would you do that? let's say you have to reproduce 20khz sine, the pulse must rise fast enough to attain max amplitude within that delay. now with a pulse that fast, how do you reproduce 20hz? the signal will rise at the final amplitude in 1/20000th of a sec or faster, and then what?  and how do you reproduce quieter signal?
the principle is a up/down switch, you have to make it so fast that you can up and down around the amplitude you wish to follow so that the general shape is right. that means a switching rate way way faster than the actual signal you're trying to make. also as Spruce said, when it's that fast, you can easily low pass all the extra moves from the pulse because they describe an ultra high frequency noise far away for the music range.
 
in short, a class D amp isn't a discrete DAC.

 

[Wiljen]

I said right up front I was probably missing some simple reason why it wouldn't work.  Now at least I understand what the reason is.  That's all I was looking for.