ok got it. your point is that we'll have more precision at each exact sample point with r2r, and that's probably true(don't know enough in modulation to tell if they bother passing through the exact sample voltage or if it all becomes a bigger math problem where the aim is shaping the wave?).
my trouble with your point is when you take that for better fidelity. how could it be? that's forgetting a little too much about how everything will be wrong between each samples. we're trying to get an analog wave in the end, not the perfect staircases. we're all telling you the same thing here. I call it staircases because I'm a noob, RRod talks about zero order hold, that's the "I know what I'm talking about" way of saying staircases. cjl mentions the same thing as he doesn't understand why you would wish to keep those staircases. and stv014 tries to keep up with your theory, which is courageous as you started wrong and only accept half of what he tries to explain each time.
if you reject noise shaping, anti aliasing, and anything that in the end offers to make the analog output look like an actual wave, what's the point? the original signal was a wave, you certainly don't think you're having the best possible resolution by turning sine waves into staircases?
here I also went and did some... how did you call it? independent research ^_^.
http://en.wikipedia.org/wiki/Digital-to-analog_converter
from the start you've been saying you didn't want to filter anything. see how that's a tiny little problem?
but another way to look at it would simply be to ask yourself why they all bother with filters in the first place if it's a bad thing? or maybe you know something the DAC engineers don't?
castleofargh:
Let me start off by thanking you deeply for your criticism, without the likes of which scientific progress would not be possible. My response is as follows.
You are confusing "what the waveform looks like" with audio fidelity. No one has the right to say "what the waveform should look like," whether smooth, rough, staircased, sinusoid, or whatever. Visual representation in a graph is but one way of representing the real signal. The basis of the S-N theorem is that there exists something in reality called a "signal" which is independent of the way we choose to represent it (e.g. time domain voltage graph, FFT, sound pressure meter, what have you). My goal is simply to point out that the information contained in the signal is best preserved by a PCM encoding and R-2R conversion process, as sigma-delta conversion "smooths over" details that are present in the PCM encoded signal.
The good DAC engineers (the ones who design high-resolution R2R DACs) DO know this. All I am trying to do is present the information in a way that is accessible to Head-fi readers. Of course a DAC engineer from ____, whose revenue stream depends on fooling consumers into accepting inferior-quality DAC designs, will not ADMIT that this information is true.
The zero-order hold is another shortcut, like delta-sigma conversion, that circuit designers have used to simplify implementation, and while it doesn't affect the resolution, it negatively affects the frequency response. The ideal would be to have a Dirac impulse output at each sample of the PCM stream.
The fact that I do not approve of extra low-pass filtering, beyond that which is provided by "the finite bandwidth of the device(s)" in question, is something I am personally advocating to improve the current state of the art in hi-fi audio reproduction. By no means am I saying that existing approaches are invalid, just that they may be improved upon.
Here's to further criticism and discussion!
- m3_arun