bewerber2
But the point is that I always have the feeling that my bounced *.wav-files sound different than my DAW playback. That's why I asked this question. Since I am big friend of WYSIWYG, I am always frustrated when I export a project to a *.wav;
It might help to think of it in terms of photography. You take a 10 megapixel picture of something and then turn around and take the same exact picture of the same exact subject with a 20 megapixel camera. To the naked eye, both unedited photos will look more or less exactly the same. The difference comes when you start editing and manipulating. Even though both photos look essentially the same, the 20 megapixel version contains much more information about what original subject actually looked like. Therefore when you start editing and applying processing, you end up with a much better result from the 20 megapixel source than you do from the 10 megapixel version. Even if your end result is going to be 10 megapixels.
It is exactly the same with audio. You will be hard pressed to tell the difference between an unedited 16 bit recording and an unedited 24 bit recording of the same source. Unedited, both will sound exactly the same to all but the most talented ears (think like 10 people on the planet could tell the difference maybe).
But once you start applying processing, the more information your source contains (i.e. bits), the better your processors will be able to do their job accurately.
That being said, here's a way to consider it that is much more literal. The nature of audio and bit rates is such that 16 bit audio only actually uses all 16 bits when the audio level hits 0dB. If your signal is less than 0dB, then your audio is playing back at something less than a true 16 bits.
And since nasty god awful things happen if we exceed 0dB in the digital world, it stands to reason that most of our audio is going to end up being something less than 16 bits if we start with a 16 bit source and then mix so only the highest peaks approach 0dB.
However if we start with a 24 bit source, we've got lots of headroom to process and mix our output to something less than 0dB and still end up with a result that exceeds 16 bits. This then allows us to dither the end result down to 16 bits without losing any detail that would be detectable to most ears.
As for your belief that your 24 bit audio always sounds different after being truncated down to 16 bits, I can only speculate. Are you listening to the resulting 16 bit output on the exact same system with all of the exact same processing in the signal path? Or are you creating the 16 bit output and then listening to it on other systems?
And if the system and processing in between is exactly the same (i.e. same speakers and same everything between the software and the speakers), have done any true blind a/b listening tests where you try to identify which version you're listening to without otherwise knowing? If not, I would strongly recommend you do so. You may be shocked at how much our preconceived notions impact what we think we're hearing out of the speakers.