I can only go up to 24 bit on the mixer…that will be ok ?
That’s fine. Virtually all audio interfaces & soundcards are 16 or 24-bits. (You can’t change the bit depth of the analog-to-digital or digital-to-analog converter but the drivers or software can up-scale or down-scale.)
I have Audacity set at 44100 24 bit…
Audacity’s [u]Quality Settings[/u] should be 32-bit float. Floating point is better for processing/editing/mixing and the conversion from 24-bit integer and back is lossless.
I should keep everything at 41000 bit rate ?
Don’t make “unnecessary” conversions and there is no advantage to up-sampling (Excpet for that floating-point processing thing). “CD quality” (16-bit, 44.1kHz) is better than human hearing. If you have a “high resolution” original and downsample to CD quality, people can’t hear a difference in blind listening tests.
The “pro studio standard” is 24-bit 96kHz. The additional bit depth gives you more resolution which may help when recording at lower levels. Since your mixer apparently has a 24-bit analog-to-digital converter you might as well record at 24-bits.
The only “downside” to higher resolution is bigger files. Uncompressed 24-bit files are 50% bigger than 16-bit files and 96kHz files are twice as big as 48kHz files, etc. Then if you want/need a different final format you can re-sample as the last step or during mastering (as a separate-additional step).
What settings should I use to export WAV files
What format do you need or want? If you’re making a CD, CDs are 16-bit, 44.1kHz, 2-channel stereo. As Jademan says, video is normally 48kHz. And most audio/video files are compressed, which means there is no actual bit-depth so there’s no harm is feeding 24-bit audio into your video editor.
and mp3s ?
The “best” MP3 quality is 320kbps constant bitrate or “V0” variable bitrate and Joint Stereo (assuming a “regular” stereo file). But we can’t really say it’s better unless it actually sounds better. Often a lower bitrate will sound the same and/or an MP3 can often sound identical to the uncompressed original (in proper blind listening tests). And a lower bitrate will give you smaller files. (I use V0 for my MP3s.)
With MP3s the file size is determined by bitrate. kbps = kilo_bits_ per second, so if you know there are 8 bits in a byte you can divide by 8 to get kilo_bytes_ per second. The actual file may be larger with embedded album artwork or other metadata. The original uncompressed bit-depth and sample rate don’t affect MP3 file size.
As you probably know, MP3 is lossy compression. If you open an MP3 in Audacity (or any “regular” audio editor) it gets decompressed. If you then re-export to MP3 you are going through another generation of lossy compression, and the “damage” does accumulate. So if you want MP3, compress ONCE as the LAST step. (AAC/MP4/M4A is immune to accumulated damage, but it’s still lossy and it’s still “bad practice” to do audio production/editing in a lossy format.)
…FLAC (and ALAC) is lossless compression and it has a couple advantages over WAV - Metadata tagging is more universally supported, and of course it’s compressed so you get files almost half the size of the uncompressed WAV.