Upsampling synths to 96 or 192 - worth it?

similar thread http://forum.cakewalk.com/FindPost/3046633

Excellent summary, Jeff. The only thing I'd add is I don't think 192kHz is worth it...then again, that's what I thought about 96kHz before I started experimenting...

Anderton
Excellent summary, Jeff. The only thing I'd add is I don't think 192kHz is worth it...then again, that's what I thought about 96kHz before I started experimenting...

I just bought a Babyface so for the first time the 192kHz option is on the table...doubt I'll use it though. I tend to use a lot of synths in my projects and should think things will get a little choppy fairly quickly, even though I have a decent rig. 

Anderton
 
Sounding better and being more accurate are not always the same thing; you might not want all those highs on warm pads. However, you'll get an even better sound if you apply a lowpass filter to the more accurate sound rather than let the computer reduce the highs for the wrong reasons, while adding in foldover distortion as a "bonus." To me, the wooliness of the foldover distortion is more problematic than the lack of highs anyway.

I've noticed this when using the oversampling mode of Ohmicide. I always have to use the post-effect LPF to tame the highs, and then it sounds great. 

One point that's being glossed over (but became apparent in the other thread) is that it is useless to upsample to 96K if the synth already has a "Hi Quality" or "2X oversampling" mode. Most do.

In short, NO! no one can distinguish between the 2 and if they do, they got lucky. 
 
If you want to bog down your DAW and waste Disk space, then YES! It's worth it!!! 
 
Stick with 88.2KhZ or 44.1kHz
 
Now, if a plugin has an oversample possibilities, then this is different, as oversampling in the limiting stage, (if your limiter has it) is a good idea at times. But this is oranges and kangaroos or grapes and apples or somehting.
 
Cj

There are different modes remember you can be thinking about this.  One mode bringing signals in through your sound card and create sessions at either 44.1K and 96K. And perhaps rendering the 96K version down to 44.1K.  I am not so sure there would be so little difference even in doing that.   Many plugins and things will probably sound better at the higher sampling rate too.
 
The other mode is using your virtual instruments and playing and sequencing them.  But when it comes to playing and performing on a virtual instrument there most certainly is a definite improvement working at 96K with some of them.  Still after doing even more experiments with this I still prefer the 96K sound.  I have become a bit hooked on it now.  My next setup will most definitely be 96K and 24 bit all the way. Apple's Mac Pro will handle it all with ease though.  The only way to do it.  But you need a DAW that runs on a Mac.
 
I can see the advantages of working all at 96K and 24 bit right from the get go and just convert the final master down to 44.1K and 16 bit at the last minute.  It always has been and still is the best and most professional way to produce quality audio.  The lower latency is also going to reveal itself in the way it feels under the hands of a great player.
 
We can work at playback resolutions now more so all the way through the audio production process and get great results. But once you get used to working at higher resolutions I think you will agree it sounds better and feels different.  We used to do it all with high end tape machines and vinyl was the playback medium at the end of that. (vinyl is inferior as well compared to a big multitrack playing back at 30 ips)
 
We can still do it in the all digital world too.  Work at 96K 24 bit all the way and create our playback medium last in the chain. The good news is the playback medium is much better now.  Why would working at a higher res not sound a little nicer like it always did.  It does.  But we can work at payback res now all the way from start to finish because many projects we do sound great that way and would hardly benefit from working at a higher res.  But then other things certainly do, like playing and creating music with 'Prism'.  (Native Instruments Virtual synth and a very unusual one at that!)  It is very obvious.  Just listen to the two files in my first link posted above and I think you will agree.
 

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
 
 
 
 
 

This is indeed an interesting thread. One thing that bothers me slightly though is: why is it not possible to change the sample rate of a project once it's started? Is there some technical reason for this, aside from the obvious that any audio clips would have to be converted? I have some really synth-heavy projects in 48kHz which I'd love to try converting to 96kHz, but I'm not that keen on the amount of work it would take to recreate them from scratch, especially given the amount of automation on some of the tracks. 

scook
It is possible to change the sample rate once the audio is converted see http://forum.cakewalk.com/FindPost/2934357

That too involves starting a new project. I realize you can change the rate of audio and import it to a new project - I just wondered about any technical reasons for Sonar not being able to change the sample rate of the same project. Is it just a feature limitation? I imagine it would be a non-trivial feature to implement, but without any information about why it's not possible, I don't know, it seems possible.....

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? 

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. 

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

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
 
 
 

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?