I've got several old projects that have all been recorded at 44.1 and I'd like to convert them all up to 48 without having to use the destructive processes outlined here :
Basically I don't want to lose all of the individual clips and the ability to go back and adjust crossfades etc.
After some research I found this thread discussing a workaround - - I need to use 48khz for pretty much the same reason - my axefx 2 only operates over spdif using 48khz.
So I've managed to convert the sample rate of all a project's wav files up to 48khz and reload it and everything seems fine at first. But what I've worked out is that Sonar's project files would seem to have stored in them the exact sample length of each wav file it uses, and it ends up truncating anything beyond that. Its ended up that I have several silent parts at the end of each section of the song throughout.
How to reproduce: - I start a project at 44.1khz, and record lets say 48 seconds of audio, which equals 2116800 samples. Save and close the project, convert the wav file from the Audio folder to 48khz using voxengo r8brain, then reopen the project which is now running at 48khz. Playback is fine to start with, but audio ends at 44.1 seconds in (which at 48khz is 2116800 samples), and you can see it in the computed waveform picture for the clip.
Is there any way or workaround to get Sonar to recheck and recalculate based on the current wav files what these lengths are?
I know that saving to a bundle file will trim the actual wav file so that doesn't work. Copying the clip and pasting into a new project doesn't work either. Its like I need the opposite of the 'Apply Trimming' command.
Does anyone at cake have a solution to this? It'd be so great if there was a second dropdown box under the 'change bit depth' in the utilities/change audio format menu to change sampling rate.
Using Sonar Platinum btw
Thanks!
Glenn