Hey ffshrine users.
This question goes to audio nerds who are skilled with audio mastering and especially wav files.

I actually tripped over a problem.
Im ripping soundtrack of a PS2 game which is working fine more or less.
I used PSound to extract all audio files and picked out the actual music with just sorting after the filesize, because music tracks are bigger and longer than just random sounds.
Captain Obvious said.

Well, but thats not the actual problem.
While extracting the files I noticed that most were saved at 24000hz Wav and only 2 files in 48000hz wav.
Probably because the last 2 being videotracks. But its not about the last 2 files.
Its about the ones that are saved in 24000hz.
Since I keep Lossless and Lossy files I always convert them to LAME MP3 CBR 320Kbits.
Those 24000hz files making problems to and LAME always saves them in 160Kbits obviously because of the sample rate which is not higher than 24000hz.
MP3 uses normally 44100hz or 48000hz. 24000hz seem to make some complications and forces the Bitrate to 160Kbits.
I want to have the lossless audio files in 48000hz which isnt a problem because the ripping tool has a function to force saving in specific sample rates.
So I used 48000hz. Btw, those files seem to have lossless 48000hz samplerates! Dont think im trying to convert 24000hz wavs into 48000hz wavs.
Check those pictures so you see im not kidding.
24000hz =
48000hz =

So far so good, the actual problem is that those 48000hz lossless files play at double speed.
Also noticeable because of the time counter in the picture.
I was able to fix this with using Audacity and changing the Track speed and pitch by -50%.
Now the file plays normal.
PROBLEM now is that we are back at those 24000hz. -.-
Because after saving it the file only contains 24000hz instead of 48000hz which is oviously because I changed the speed and pitch by 50%.
Changing only the speed doesnt fix the problem btw as some might think.
I also was able to change the channels from 2 to 1 and this fixed the problem too but I noticed alot of quality loss when comparing the 1 channel 48000 wav with the 24000 2 channel wav.

Now my question is:
Why is this happening and is there no way to save the file with 48000hz and the right speed?
I know, this is maybe a noob question. I never tripped over this problem.
Sure, I could just save it with 24000hz and let LAME convert it to 160Kbits but I would like having the lossless audio.
24000hz is a bad way to use music files.
What I also noticed is that those pictures look weirdly the same even though there is a different timeline.
Am I stupid or something? I dont really get it...
Could somebody tell me a way to save it with 48000hz and the right speed?
Is there even a way or do I have to life with the 24000hz file? I would find it nice to be enlightened.
Im really not understanding the problem actual...
Why is the pitch bound to the speed anyway?
What the hell my brain.
Sorry, im usually not working with WAV files.
They giving me a hard time actually.
I mostly use Flac and MP3 files.
I wanted to convert those WAVs to Flac anyway but with this issue its a waste and it sounds shit anyway.
I also listened the PS2 Rom tracks with using PCSX2 and they sounded better than my 24000hz rip fixes.
Maybe just imagination though.
Cant compare 48000hz because of the fast playing.

Thanks for every help!

As you have correctly deduced, saving as 48000khz isn't actually doing anything at all - it isn't doubling the sample rate, it's halving the amount of time it takes to play the same number of samples. Resampling is not the same thing as changing the sample rate. You are wanting to do the former, but your software is doing the latter.

I suspect that the music you're trying to export actually is stored at 24000khz which is why it is exported in this manner. If you really want output at a different sample rate, I would advise investigating SoX (SoX - Sound eXchange | HomePage (http://sox.sourceforge.net/)) or if you fancy a GUI, try Foobar2000. The built-in converter has a resampler (Resampler PPHS) which will do what you need to do.

Thanks for the reply.

Im using Foobar2000 anyway to convert files and everything.
I also tried to experiement with the convert settings but I didnt found any sort of resampler function.
I see, its in those DSP things. What is this anyway? How to use it?
Sorry, never worked with this kind of functions.

EDIT: Ok I figured out how it should work.
However, what exactly should I use at the converter settings?
I tried DSP Sample Rate 48000hz and 24000hz, both with Ultra Mode thingy checked.
Didnt fixed anything. I also tried to use dont reset DSPs between tracks.
Didnt worked either. Do I do something wrong?
Dont really get how this function works.
Btw, I noticed that my Wav files say they use PCM Codec.
However, my converter uses Wavpack. Is this a problem?
Maybe thats why it doesnt work?

But anyway, you know that I want to keep the 48000hz but with the right speed and pitch.
Sampling them back to 24000hz isnt what I try to accomplish.
24000hz files sound bad. Or are those files like that and I cant change it? Are the ripped files by default 24000hz and I cant get them to work correctly with 48000hz? Would be pity. But to be honest I dont really understand anything anymore.
Those wav files are like a riddle to me. What has the hz rate to do with the actual playback speed of the file.
Same with pitch. Never worked with those things and I never had a problem with those with MP3.
*Head hurts*

I had some similar occurrences in ripping a PS2 game the other day. I was using MFaudio to rip the audio files and convert them, but when I opened the converted files they were playing twice as fast and I figured it had something to do with the sample rate. Low and behold, opening the raw PS2 files in audacity (because somehow it works, half the time) revealed that their sample rate was actually 22050Hz, but MFaudio was saving them at 44100. So I changed the output to 22050 and now they all play fine.