The Wavewizard method is slow but failsafe, it can't go wrong unless something is done wrong, so you can use it as a test to check which part of your process is getting screwed up (decoding, remuxing, or encoding).
In essence NTSC is 29.970, but pulldown removal makes it 23.976.
I would use avisynth, avs2wav and aften for this job,
with this script (audio.avs) for decoding and timestretching:
Code:
NicAC3Source("audio.ac3")
TimeStretch(tempo = (23.976*100/25)) #takes audio from 25 fps to 23.976 fps while maintaining the pitch
ConvertAudioTo16bit()
and then, with the command line or in a .bat for loading the script and feeding it to aften:
Code:
avs2wav.exe -f wav -i audio.avs -o - | aften -b 448 - stretched_audio.ac3
It does everything at once, like you wanted it. No intermediary files, saves time and hdd space while decoding, timestretching, and reencoding, retains the pitch. And the tools are free, small and easy to use.