It is true the method described in the documentation uses a new project file but the method described in the link works with the existing project. From the link:
 
"Then when you re-open the original project (or a copy of it) with those re-sampled files in the audio folder, SONAR will automatically set the project rate to match the audio it finds, and you're done."

scook
It is true the method described in the documentation uses a new project file but the method described in the link works with the existing project. From the link:
 
"Then when you re-open the original project (or a copy of it) with those re-sampled files in the audio folder, SONAR will automatically set the project rate to match the audio it finds, and you're done."

Right, so what you're saying is that if you have a project that's only synths, you could theoretically change the sample rate of the project by inserting a dummy audio clip and then converting it to the new sample rate? 

If there is no audio at all in the project, the sample rate of the project may be changed directly. Open the project and change the sample rate in Preferences > Audio > Driver Settings. You should notice the change in sample rate of the project in the transport module.

scook
If there is no audio at all in the project, the sample rate of the project may be changed directly. Open the project and change the sample rate in Preferences > Audio > Driver Settings. You should notice the change in sample rate of the project in the transport module.

That setting is only for the default rate of new projects, is it not? Therefore it does not change the sample rate of the current project. 

As long as the project contains no audio, changing the value will change the current project sample rate too. Try it and see the sample rate change in the transport module. You may even hear the sample rate change in the interface, mine makes an audible pop when the sample rate changes.

Ah OK, it's a little confusing that the setting is labeled "default sample rate for new projects" then. 

Wouter Schijns
Sander, great to read your experience/thoughts, hope Sharke will get something from this thread
 
under Preferences, when you set higher rate/lower buffers.. Sonar specifies a roundtip with lower latency
I understand that as latency you will have in music out of Sonar (plugin latency not calculated inn).
I think I hear a latency when exporting a song, then A/B listening to Sonar and the exported file.
the wav file sounds not as real/live as the Sonar project, it has slower attack to my ears....the difference is just small...can only hear it on headphones.
those things don't come through latency ? or maybe the Media player reduces the sound quality ?
a saw a post somewhere from a mastering engineer, he likes the overall DAW latency under 10 m/s
 
 
 
 
 

The latency figure there shows you the latency between you taking an action (playing a synth note, moving a fader or recording audio) and Sonar processing it/outputting audio. This only has effect on live behavior, because it is basically a discrepancy between what you're doing "live" and the existing track. When you're bouncing the track you're not doing anything live, there is no discrepancy, and no useful meaning for the term "latency" on the bounced track. I have no idea why a mastering engineer would prefer low latency, usually mixing and mastering don't require low latency, only tracking (and playing live) does. Perhaps you (or he) misunderstood the context of the question or statement.

There are many reasons why a song could sound worse when you play it with WMP. You could be using a different soundcard, you could have recorded at 96/24 and bounced to 44.1/16, there could be a bunch of "sound enhancers" active in WMP or in your soundcard's app, and so on. The one thing I guarantee doesn't cause it is latency.

Great, so with the r8brain converter I was able to quickly batch-convert the audio from a project folder I made a copy of up to 96kHz and thus convert the project to 96. Everything sounds great - the only problem is that a part I was using the AAS Player for (a plucked string patch) sounds absolutely awful now. It has these extra harmonics which ring out. Here's a couple of audio clips which demonstrate - I've turned off all effects and then exported them both @ 44.1kHz:
 
From the original project @ 44.1kHz
https://www.dropbox.com/s/3cgvc05tei6mziz/AAS44-AAS%20Player%201.wav
 
 
From the upsampled project @ 96kHz: 
https://www.dropbox.com/s/c2s26z1121j7w7r/AAS96-AAS%20Player%201.wav
 
 
 

Sharke are you going about this the correct way. This the way I would do it:
 
Create your initial session at 44.1K get your VST synths playing at 44.1K as well. Get the music all correct. (At this point you may want to render these parts out at 44.1K audio as a backup) Leave all the other 44.1K audio alone. It is only synth parts you are interested in.
 
Save all those synth parts you want to up sample as Midi tracks. ie Midi data only.
 
Have a 96K synth rendering session now:
 
Create a new session at 96K right from the get go. Insert the same VST's and load them up with the same presets. Set the tempo of the session the same as it was at 44.1K.
 
Now import those midi tracks you generated at 44.1K and get those midi tracks to play your VST's now except this is all happening at 96K.
 
Render out all these VST's to audio at 96K.
 
Down sample that rendered 96K audio now to 44.1 K again and drop them into your original session.
 
Compare them with the VST parts you renderd out at 44.1K and you will find often the 96K down to 44.1K versions sound better.
 
Any other way is not correct. There is no point in rendering out any audio from VST's at 44.1K then up sampling to 96K and back down again. Not sure of you are doing that but that wont work.
 
The idea is to get the midi parts playing these VST's right from the start at 96K. It is only the Midi parts that allow you to do this only. You are not messing with the original session audio at 44.1K. Leave that all alone. The idea is to just remake any VST parts at 96K and down sample them back to 44.1K.

Hi Jeff,
 
What I am ultimately trying to avoid is having to set up the synth parts from scratch. It's easy enough, of course, to save the MIDI parts and drag them into a new project, but I'll sometimes have 3 or more automation lanes going on for each synth as well. So I'd have to copy and paste all the automation into the new project - quite a hassle, and I never seem to get on well with having 2 projects open at the same time for some reason 
 
Have a listen to the two audio clips I posted - one is exported from a 44.1kHz project at 44.1kHz, and the other is the identical part exported from the 96kHz project at 44.1kHz. It seems that 96 did something nasty to the sound which persisted when I downsampled it to 44.1. 
 
So I'm thinking that the best thing to do would be to create a copy of the whole project folder, batch convert all the audio in the copied version (in order to force the whole project into 96), figure out which synths sound better at 96 and export those at the lower sample rate. Does that sound like a plan?