I use an M-Audio ProFire 610 audio interface with Ableton and I was wondering if anyone had any insight as to what is the best way to oversample individual tracks. Before I export a mix I will typically freeze and flatten my MIDI tracks to save Live's CPU% so I can listen to the track in its entirety without it glitching before exporting it. I prefer this method because it is quicker then exporting the whole MIDI track and putting it back in my set as an audio track. Occasionally I will record the MIDI track to an audio track internally, depending on which 3rd party software I am using.
So my question is as follows; assuming that I have my audio interface set to my desired bit depth and sample rate during each of these processes's, which of these methods is the best, and which is the worst for oversampling? Also if I freeze and flatten a MIDI track without my audio interface, do I lose the ability to oversample it effectively at a later time?

 

in fl studio i just save as new version render and listen then make adjustments that's only cuz i run 2gb ram but are you trying to bounce tracks out in wav format or something im not understanding what you want to accomplish

 

http://www.dailymotion.com/video/xifyzl_ableton-live-tricks-oversampling-high-quality_tech

 

after freezing you can move (or "copy" when you want go back to mididata) your freezed midiclip to an auditrack, it is already rendered.

 

awesome hint

 

I’ll try to better explain what I am seeking to accomplish here. Since I typically convert all my MIDI tracks to audio tracks before exporting the final mix, I want to know if I need to set my audio interface at 24bit192kHz, during the process of converting the MIDI tracks to audio tracks, if I want those components of my mix to be oversampled at 24bit192kHz on my final mix.

If I don’t have my audio interface connected while I convert my MIDI tracks to audio tracks, I would like to know if I lose the ability to oversample those audio files since they have already been converted from MIDI to audio at a lower bit depth and sample rate.

As far as I know, the purpose of using an audio interface to oversample audio is so that when the audio file is converted to the standard CD quality of 16bit44.1kHz, it will still retain the same digital information of the bit depth and sample rate that it was oversampled at, thus creating a smoother signal and (somewhat) better sound quality (I think?) I wish I knew more about this topic.

Any thoughts, comments, answers, corrections, etc. are appreciated...

 

well, live is internal recording with the samplerate setup in perferences and there you will have only the choice of your audiointerface is offering

the bitrate is not so important, you usualy level your tracks, that you dont rise 0db (=16bit). but lets make a scenario: you are recording your guitarplaying and you play a great solo you never will be able to play again. but accidently sometimes the level rises over 0 db. with a 16-bit interface the a/d converter gets overload , the result is a digital clipping and the record is not useable. there it is good to have a 24-bit a/d converter to be able to record more than 0db. internaly most DAW work with 32bit to avoid clippings.

the samplerate is important. let say you will record a sine with 15khz, with samplefreq 44,1khz you will get about 3 samples of you sine. when you reconvert to analog with these 3 samplepoints you will get everything else than a sine. its more like a pulse/square, so your audio ist distorted. a higher samplerate will lower the distortion.

you can see the sine, the grey part is the result after a/d d/a convert, already very diff from orginal (sine)wave

well, why recording in 192khz when later the audio has to be converted to 44,1khz, because its standard rate every %§$"/§$ CD- and mp3 player play?
A: while poduction you may decide: wow, now i (or maybe the masteringengineer) like to route this fat bombastic killa chickenshaking guitarsolo thru a 1176 outboardcompressor, then back to daw to record and mix. there you will have lowest distortion with the highest possible samplefreq thru d/a a/d convert.

its always good to get best quality as possible, but to hear most of this better q you will need the golden ears

result: best is to record in max samplefreq as you can. its also possible to oversample a 44,1khr recording later, but oversampling is calculating steps between the stored digitalwords. just a calculating! when you record with high samplerate you will get the better result of your recording.
the effect get more strong the higher the freq to convert. sadly dont find a graphic with extreme freq like the 15khz freq by 44,1khz samlingrate with just 3 "points" to rebuild the signal, but when you maybe understand my writing you see it will be a big difference between to record a wave with more samplerate than oversampling a lower sampled wave (more true point to rebild, the calculated points from oversampling could be very different)


sorry for my english