I have a bunch of cool samples I found and none of them are 16bit 48khz.
If I HAVE to I guess I could create a session and convert them in a DAW. But that’s a lot of work.
So I am wondering if there are drawbacks.
If I throw a bunch of samples into the AR does the conversion happen while I transfer and then stored permanently as the proper bit rate or it converts the samples every time I load one into project?
i just received my ar a few days ago and have been dealing with uploading samps the past 24 hours.
if you want a clean sailing ship of uploading, most definitely, convert.
long samples are a problem too, i’ve noticed that if you upload those individually, they’re a lot more successful.
finally, if you don’t convert aiff files, they’ll crash c6.
oh, and anything more than 48k will not upload.
44.1 and 48k will.
cheers,
ian
Yes always.
I always chime in on these posts to say that you should definitely pre convert to 48kHz 16bit with a program that has a good sample rate conversion algorithm.
Most programs out there, DAW’s included, and I’m sure C6 (since C6 barely works in general) are terrible at sample rate conversion and leave all kinds of nasty artifacts in the sample.
http://src.infinitewave.ca/ for evidence
I use Audacity because it’s SRC is very good, it is free and cross platform. You can set it up to process multiple files simultaneously so it’s a pretty quick process.
Oh thanks a lot for a free program link!
Oh thanks a lot for a free program link![/quote]
Your welcome, enjoy, it’s a great audio editing program.
SRC is not something to take lightly.
First - yes I agree with everyone ALWAYS convert before it hits the AR.
First of all, unless you do that, the transfer rate is super slow.
Second - as far as I know, the AR doesn’t really converts. It truncates which is not the same thing!
You are better off using a quality SRC programs.
The best bang for teh buck is to you use Voxengo’s R8T Brain Pro.
It’s not free, but affordable. It does WAY better than any free utility and I know several Mastering Engineers that still prefer it than other more expensive solutions.
Also - if you own Izotope’s Ozone it has a VERY good SRC on board.
Eitan
SRC?
SRC?[/quote]
Sample Rate Conversion
Hi there. Can you explain this a little more?
Truncating = throwing away.
When you simply save an audio file to a different format i.e. a 24 bit file 96 KHz into a 16 bit file 48Khz then there is no dithering (conversion of bit depth) nor Sample rate conversion.
The extra data is simply disregarded, or ‘truncated’ to fit the new Format.
Since there is no mentioning of any actual conversion on the input, I assume the AR is simply truncating audio data that exceeds its playback format.
Eitan
Just some technical quotes:
Sample rate conversion: doesn’t requires dithering. Just filtering of the frequencies above “Nyquist” to avoid aliasing.
Bit depth reduction/truncate: does require dithering to avoid artefacts added by quantisation error. Most noticeable with high dynamic range material at its lowest amplitudes.
In many cases you won’t perceive a thing but as guys said, the best is to process your samples. 
RX is a great choice and Amplitude will do a great job also, both have batch converters making the process of resampling several samples a breeze.
Happy resampling!
and make sure to normalize to no louder than -3dB!
eh, really/why?
i often go up to 0. most things sound pretty good this way.
Into the AR?
What kind of samples are you importing?
I tried normalizing to zero and man my samples sounded like shit in the AR. I usually go in with a lot of headroom - -6db to -3db.
Actuallly you can normalize to 0db but you have to keep sample level’s parameter at low levels.
A common mistake with the RYTM is normalize samples to 0db then set a sample level value of 127 assuming 127 is equal to 0db.
Remember always to use track level and amp levels at high values and sample level at low values and your samples will sound pristine.
I spent a whole weekend testing.
I’m sorry but that is not the case on my end. I don’t send my samples at that amplitude anyway.
It might have something to do with using C6/AR to resample at zero db though.
Actually this has been covered a few times in this forum with people coming up with the same conclusion as me.
If you’re coming up with different results, then it might be good to get to the bottom of it either way
I didn’t test C6’s resample. I tested with pure mono sine waves at 0db 44.1 16, and the harmonic distortion added by the RYTM was a totally acceptable and at expected levels.
I will re-test on the next days, this time involving 0dB and C6’s resampling.
I´ll put the results here and in a separate topic.
It would make sense to convert dynamic samples - not static waveforms.