Engelmann's MyFormatConverter Basic is today's GOTD, so thought a few notes might be helpful.
Natively audio is analog, but there's no way to record & store fully continuous sound waves digitally -- much like video, digital recording means a series of snapshots in time, & those samples are stored in audio files. The 2 most common sampling frequencies are 44.1 kHz & 48 kHz, though it can be higher or lower. 44.1 and 48 are a bit if an oddball choice -- the math doesn't work to easily convert between them. You can do the conversion in audio software, and it'll probably sound OK, but if you're after best quality record at the sampling rate that's native to the sound card or audio chipset, and if or as possible, leave it at that.
There are also issues converting audio to different bit depths... just like a photo with 24 bit color will look better than a 16 or 8 bit version, 32 bit audio has more data than 24 bit or 16 bit audio. It's recommended that at the final step after any editing you apply something called dithering when you've changed the bit depth to try to get rid of any noise patterns or resonance. Increasing bit depth or sampling rate after the audio's been recorded will not increase the quality because as with enlarging an image or video, the data just isn't there.
Working with video in today's 7.1 audio world you can wind up with problems trying to save a file in Windows' native & lossless .wav format, which works well up to 2GB, and may work up to 4 GB, but that can be very iffy. Wave 64 [.w64] is a decent solution, though not all software supports it. As possible you want to record & edit uncompressed audio that's usually stored in .wav or .w64 files. Compressing audio [& decompressing during editing] loses quality and takes processing horsepower, so you want to do that as a final step to shrink the file size.
izotope[.]com/en/learn/digital-audio-basics-sample-rate-and-bit-depth.html