Sample rate conversion Options

[MannyCalavera]

Hello everyone.

I have a question regarding sample rate conversion algorightms.
I don't know how they work, but I guess you guys are more familiar with this

Let's say I have a 48 kHz wave file that I want to use in something I have to mix
with some 96 kHz material at certain places. This requires the 48 kHz file be resampled
and in the end back to 48.

So in short: does resampling 48 --> 96 --> 48 change the original or can the original
be restored 100% ?

I'm using Wavelab for this.

[Wombat]

I admit i have no clue about some things you talk but to summarize my findings i still don´t see anyone convincing me there is upsampling without lowpassing, so changing EVERY single musical bit in the process.

On http://en.wikipedia.org/wiki/Upsampling they talk about the 2 ways of implementing
1. Add zeros between each sample -> to my understanding all resamplers do
2. Filter with a low-pass filter which, theoretically, should be the sinc filter -> that no resampler does because, also from Wikipedia:

"The second step calls for the use of a perfect low-pass filter, which is not implementable"

[saratoga]

I admit i have no clue about some things you talk but to summarize my findings i still don´t see anyone convincing me there is upsampling without lowpassing, so changing EVERY single musical bit in the process.

On http://en.wikipedia.org/wiki/Upsampling they talk about the 2 ways of implementing
1. Add zeros between each sample -> to my understanding all resamplers do
2. Filter with a low-pass filter which, theoretically, should be the sinc filter -> that no resampler does because, also from Wikipedia:

"The second step calls for the use of a perfect low-pass filter, which is not implementable"

Although you can never make a perfect sinc filter, its possible to build ones very very close to perfect, such that any difference is below quantization noise except for a tiny region right around the Nyquist limit, which of course won't include audio anyway because of the anti-alias filter on the original recording ADC.