I understand the signal flow for SONAR X3 fine, but it strikes me that the fader, gain control and effects bin are not in the line to the actual recorded track. How would one record with compression? I know it is a dumb, and probably easily answered question, but I am from the old skool (read: 20 years ago) world of recording, and things were a little different then, beginning with the fact that generally we'd bring mics into many channels to go to other tracks sometimes, using effects, faders and all on the initial channel to affect the input. I'm assuming there is a way to do that in a DAW that is basically the same, but figured rather than search the world and web for a day, I'd ask those in the know here.
 

In DAWland, FX can only be applied after A/D conversion. Theoretically, you could apply FX to the digital audio stream before writing it to file (and I gather some DAWs do allow "printing" FX like this, but since the great advantage of working in digital is non-destructive processing/editing, there's little value in that. Input Gain is a little oddball in that it is only applied to previously recorded clips, and not live input whereas FX are applied to both. I'm not sure why that is.

Thanks for the response brundlefly (great name, btw!).
 
I realize, and as you note, that enough editing/comping is possible so that effects and such are not necessary as often, especially with unlimited tracks, melodyne, take lanes, etc. However, one thing I can't get past wanting to use is light compression on vocals and acoustic guitars going into the channel. I realize I can just keep the level down to give myself a safety net, but, at least in the old days when I used 32 tracks of digital audio tape in my studio, there was still a relevance to "tape saturation," as it were. Is that no longer the case? Seems my mind can't quite get past the idea that I still want a strong signal level for recording so there is more information accurately captured. Might just be that old guy cap thinking though.
 
Also, I wondered, is there no way to send something through a plug-in compressor on another channel and then back for recording on the input track? If possible, that would certainly be a work-around if I insisted on keeping old habits! lol

Coming in, you couldraise a standalone instance of a plug such as T-Racks, Ozone,or, I believe Amplitube, and set the DAW to record the output of the plug as opposed to the instrument input.

Depending on you specs, you may notice latency/timing issues you'll need to correct.

Given unlimited tracks, I like to record both processed andun, but you're almost certainly going to see phase and comb filtering issues mixing processed and unprocessed tracks.

I'm not sure if what you ask is impossible because it never occurred to me to try it. I can't think of an advantage these days in recording any process live.  As brundlefly pointed out,  seems like it would tax your CPU and defeat the advantages of working in digital. In the analog days it was important to record a very hot signal to maximize signal to noise ratio.  In the digital world, as long as you have a decent level, clean signal, it is best to avoid pushing the levels too hot. Of course now that you have mentioned it, i have to try. :)

Digital tape can't have tape saturation unless the recorder is badly broken. Keep in mind that anything over 0 dB will clip, meaning it will go no higher. For your needs I would recommend a hardware compressor that is placed before the A to D conversion. 
 
 

John
Digital tape can't have tape saturation unless the recorder is badly broken. Keep in mind that anything over 0 dB will clip, meaning it will go no higher. For your needs I would recommend a hardware compressor that is placed before the A to D conversion. 
 
 

 
I do have occasions when  i need to use a compressor when recording like live event video where I have people of varying, unpredictable sound levels.  For that i use hardware.

John
Digital tape can't have tape saturation unless the recorder is badly broken. Keep in mind that anything over 0 dB will clip, meaning it will go no higher. For your needs I would recommend a hardware compressor that is placed before the A to D conversion. 

 
I used the term "Tape Saturation" only as a description. More accurately, in the days of digital tape, a hotter signal meant a more accurate "read" of the data. So you would still want to aim above a certain level threshold going onto tape. And obviously, in a situation with a difficult noise floor situation it could also be important, depending, obviously, on what one was recording.
 

Jay Tee 4303
Coming in, you couldraise a standalone instance of a plug such as T-Racks, Ozone,or, I believe Amplitube, and set the DAW to record the output of the plug as opposed to the instrument input.

Depending on you specs, you may notice latency/timing issues you'll need to correct.

Given unlimited tracks, I like to record both processed andun, but you're almost certainly going to see phase and comb filtering issues mixing processed and unprocessed tracks.

By Joe I think he's on to something here! That might be worth a try, if for no other reason than to know I could do it if I had to!
 
Oh, and btw, one of the major reasons I would like to be able to do this is that I am working on voice over stuff myself, and when I'm recording I don't want to worry about levels at all once I've set them. Currently I have to record with plenty and plenty of extra room on top, so a compressor, or even a soft limiter, would help nicely.
 

John
Digital tape can't have tape saturation unless the recorder is badly broken. Keep in mind that anything over 0 dB will clip, meaning it will go no higher. For your needs I would recommend a hardware compressor that is placed before the A to D conversion. 
 
 

  I agree with John on this.  Even now, in the digital age, there are plenty of engineers that put a hardware compressor in front of their converter.  Not only for safety reasons, but because a good pre-amp or compressor can thicken a sound and make it sound better going into the computer.  It's perfectly valid as long as one doesn't overdo it.  If you have the hardware, I would recommend experimenting until you're satisfied with the results.  Many of the classic recordings of the past sound so good today because engineers sometimes had to make hard DECISIONS, whereas today, so many want to play it safe and fix it in the mix.  The end result of this being homogeneous recordings that have a certain sameness to them.  There are exceptions, of course.

As was already stated, there is really no point to applying a software compressor to an audio stream before writing it to disk. The only thing you could possibly achieve is degrading the audio signal permanently. The point of doing compression before recording in the old days was indeed to keep the
level higher without clipping the tape. Not only do today's 24 bit converters generally have plenty of headroom and a super low noise floor so that you can record at almost any level, but applying a software compressor to the signal will always, by definition, happen AFTER A/D conversion (when the only place where it MIGHT have made sense is BEFORE). You will literally achieve the exact same effect/sound by applying the compression after recording/file writing, with the exception that you'll be able to change your mind later and switch it off, or alter parameters.

In other words, if you really need to compress before recording because your signal is super unstable (and you're afraid your A/D converters don't have enough dynamic range, which would be relatively rare) the only way is still to use an analog (hardware) compressor.