Project sample rate vs the audio interface clock

I used Audition for digitizing archival audio and I mostly work in 96 kHz because it's the current archival standard for analog audio, however I occasionally need to transfer DAT or ADAT tapes at 44.1 or 48 kHz. One thing I noticed when I went to change the project format was that my audio interface clock did not change and match the multitrack settings. I was able to change the clock manually but I had to go to the Audio Hardware preferences.
I find this behavior for a DAW very strange. If I have a multitrack open of one bit rate and my clock set to another, how does that work? How does Audition 6 play and record audio at differnt sample rates than the interface clock? When I open a multitrack that's 96 kHz and my clock is set to 48, what sample rate am I getting? The audio file being saved says 96 kHz but is it?
Audition 6
RME Fireface UCX
Mac Pro mid-2010

loneraver1 wrote:
If true, this is indeed strange. I have used just about every DAW under the sun for the last 10 years based on what my employer purchased and I have never seen a DAW allow you to record sample rates where the project didn't match the incoming clock.
There should be an option that locks the audio interface internal clock to the opened project. It makes it annoying any time I have to switch between recording in 96 kHz to 48 kHz. That's annoying on it's own right, but what's really annoying is that it's an extra step that I have to train our archival technicians who don't have a techical background in audio.
I can definitely say it's true for my set-up!
Surely, the "extra step" you refer to is not necessary if AA is resampling the audio to match the desired sample rate?
As for the "option" to "lock the audio interface clock to the opened project", for those cards, like mine, which cannot have their sample rate changed by the audio software, how is that going to work?  I suspect this was how AA CS5.5 and earlier worked: unless session/project and interface sample rate matched, no audio.  I much prefer the current "resampling" method; no necessity to have to think about soundcard/interface settings every time a different sample rate is being used.
FWIW, if I want to record something at 44.1 I always ensure the card and AA settings are 44.1; but for "playback" (or video editing which I do quite a lot of and which requires 48 - my video editor software will not "play" unless the audio sampling rate of the card is 48 - ) 48 is my "default" setting.
JMO!

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