> Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true?
No. Watch the Digital Media Primer and Digital Show & Tell, Monty (the article's author) explains how digital sound encoding works and why the analog signal will be reproduced beyond your ability to notice imperfections either way: http://xiph.org/video/
And that's for native 24/192, in your case since it's been upsampled the 24/192 signal can't have more information than the original 16/44.1, since that's all the information that went into it.
The engineering reason to upsample is to simplify the analog anti-aliasing filter on the other side of the conversion by giving it a wider transition band to work with. It also means one analog filter can handle a wide range of samplerates.
masklinn is right though—from a consumer perspective, there's no real reason to care how the conversion's done. Hopefully your DAC was designed by a team that knows what they're doing and will handle whatever rate you feed it.
Of course, if you have one of those obnoxious devices that can't actually clock below 48kHz or a mixing daemon configured to run at 48kHz, you're stuck oversampling anyway, and depending on your environment that might be done anywhere between amazingly well and linear interpolation. Many years ago I was stuck playing that game on Linux—in that case, there's an audible benefit from using a better designed upsampler in your player.
But if you're equipment's not broken, it's a waste of CPU cycles.
No. Watch the Digital Media Primer and Digital Show & Tell, Monty (the article's author) explains how digital sound encoding works and why the analog signal will be reproduced beyond your ability to notice imperfections either way: http://xiph.org/video/
And that's for native 24/192, in your case since it's been upsampled the 24/192 signal can't have more information than the original 16/44.1, since that's all the information that went into it.