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That's what I use to export, I'm not having problems exporting though. I just inquired because something always just seems "different" about a track when I export it, but I could just be crazy.

Oh and why never the analysis files?

cause they're pointless and just take up space on your computer. live will create shitloads of analysis files. they're annoying as fuck.

they do have a purpose but i forget what it is because it's not very important, it's some really specific function that you won't use most of the time iirc, probably why i forgot.

Yeah, I've seen them in my sample libraries. I thought they were just for quicker loading if you use the same sample again.

ah, maybe that may be important to you, in which case ignore my advice

MP3 only works in 16bit anyway so when you convert to MP3 you WILL get 16bit, even on 320s.

Other than that, it's pretty much all been explained, truncation and all.the good thing is the truncation happens at the last stage so actually the damage being done is in the nature of going from the high quality sound in your DAW down to the lower quality in a WAV file (your DAW operates at 32bit FP or 48bit integer, the latter is around 16 times better than 24bit audio and the former is probably over 9000, its such a ridiculous number better).

It's the bits really that make all the difference.

Another reason could be the plugin/s you are using (for example sytrus) is set to render at a higher oversampled rate (a setting like this is usually made visible within the plugin) though this is more specific towards sounds in a song being different rather than an entire track.

dubunked wrote:export settings for ableton (since i know ur an ableton user):

file type: wav (aiff is fine though)
sample rate: 44,100
bit depth: 24 (i've heard 16 is fine though, the larger the bit depth, the larger the file i believe)
dither: triangular
NEVER CREATE ANALYSIS FILES

i agree with this, pretty much exactly what i use aswell

gen_ wrote:MP3 only works in 16bit anyway so when you convert to MP3 you WILL get 16bit, even on 320s.
(...)
It's the bits really that make all the difference.

Wikipedia wrote: During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples. If there is a transient, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

MP3 stores data in the frequency domain, not in the time domain. You're talking about something you don't even understand.