I don’t have my s2400 yet (it is still in transit somewhere), but am reorganizing my patchbay to make room for it and had a few questions I was hoping someone could help me with.
Inputs: If I use line ins 1 and 2 it gives me all the options for recording, correct? I can choose via software whether it will bypass the antialiasing filter or not?
Outputs: With the last update it sounds like tracks can be assigned to the same channels without choking each other. So my plan was to connect the main outputs to my patchbay as well as the channel 1 and 2 outputs. So as long as I didn’t assign anything to channels 1 and 2 everything would play through the main outs, correct? And I could also assign whatever I wanted (even the whole mix) to go through channels 1 and 2 if I wanted it to go through the dynamic filters instead?
If you have the 8 individual outs connected there will be no sound through the main output. It is hardware switched so when a cable is inserted to an individual out that signal will be removed from the main out.
So if you want to use some kind of stereo summing of the s2400 in your patch you better use the headphone output for that. But that will not have the analog output filters in the signal.
As for the inputs 1/2 have the input filters and 3/4 doesn’t, it is not software switchable in the machine so you just plug into the one you want to use. However you can have them both patched in your patchbay and you can choose which inputs to record in the S2400.
Since the last update you can assign tracks to output channels freely, so if you just want everything to go out a stereo pair you can just set every track to output 1/2 for example.
Good points Melker, just a clarification that Input Line 1&2 is software switchable. In sample mode at the top left of the display you can change the source/rate/depth. From page 73 of the manual:
Line 1&2 and Phono 1 inputs can be routed through the Classic anti-aliasing filter that is required for sampling at 26KHz, so those inputs have the option of sampling at either 26KHz or 48KHz. Line 3&4 and Phono 2 inputs can only be sampled at 48KHz"
To HBIII’s point, input Line 1&2 gives you all the (non phono) recording options.
So are inputs 1/2 always routed through the anti-aliasing filters and you can just choose to record 26khz or 48khz, or when you choose 48khz they are no longer routed through the filters?
Also, are there any sound differences when routing out of outputs 7/8 vs the main outputs?
I dont like using headphone outputs for my main outs as the built-in headphone amps on gear are never that great sound-quality wise.
Are you sure the input filters are switched in software to? Isn’t there only the option to record in 26 or 48kHz but the signal still goes through the same input circuit?
Since the manual singled out that the Classic anti-aliasing filter is required for 26Khz but made no mention of anti-alias filtering at 48Khz led me to believe that the input filters are switched in coordination with a change in sample rate.
This could certainly be a misinterpretation on my part.
