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Added preamp to Focusrite, much latency later... is this normal

labeledbass

Supporting Member
May 24, 2015
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Basically what the title states. Have a 4i4, direct into interface, hardly any latency, added a sansamp rack mount in between my bass and DI input into the Focusrite and now a much more noticable difference in latency. I havent tried 1/4 inch outs on the preamp/interface, if for some reason unbalanced is less processing.

Is this normal? Really have to to do some tricks just to play along with drum track in DAW, and hear my bass live monitor.

Would reamping the preamp like a insert work better with (insert)

Thanks in advance
 
Are you sure the buffer setting on the focusrite software didn't get increased? Analog devices don't cause latency.
Still getting used to Reaper but I tried lowering as much as possible, didnt get artifacts but PC blue screened and force restarted. Only thing I changed was device>(checked "request block size") changed from 256 to 128, that caused PC reboot so I changed it back (I'm assuming that's where I lower it, the latency in the corner did show lower time before it restarted). I was under the impression if you ran the buffer low it just sacrificed the quality. Computer is a little dated, but nothing like a Celleron or anything.

Hopefully (and not) I'm remembering there not being a difference, I just dug out my preamp excited to play with a track and, through the DAW its simply not possible, fast at least. I did work out a way of listening to monitor in left ear and live from interface in right, I just have to align the tracks up after the fact
 
Basically what the title states. Have a 4i4, direct into interface, hardly any latency, added a sansamp rack mount in between my bass and DI input into the Focusrite and now a much more noticable difference in latency. I havent tried 1/4 inch outs on the preamp/interface, if for some reason unbalanced is less processing.

Is this normal? Really have to to do some tricks just to play along with drum track in DAW, and hear my bass live monitor.

Would reamping the preamp like a insert work better with (insert)

Thanks in advance

There is something else going on. The latency should be exactly the same with or without the Sansamp. Did you try - while leaving everything else the same - to just unplug your bass from the Sansamp and plug it into the Focusrite? Did you measure or did you hear the latency? If the Sansamp would have any latency that is
perceptible on its own, you simply would recognize it, whenever you use it.
 
There is something else going on. The latency should be exactly the same with or without the Sansamp. Did you try - while leaving everything else the same - to just unplug your bass from the Sansamp and plug it into the Focusrite? Did you measure or did you hear the latency? If the Sansamp would have any latency that is
perceptible on its own, you simply would recognize it, whenever you use it.

Making wonderful headway so far, I did a bypass and it was the same. Went to the ASIO config and matched the sample rate of the focusrite and DAW and now I'm down as far as 64, with not much latency at all, if any. I think Reaper was trying to do its automatic call on the control panel for changes and only reaper was changing, not the focusrite, hence the crash(s). At least I'm really hoping that was the issue, because so far it sounds great.

Just so I know I've gone too far down, what does it sound like when you've gone down to far with sample rate? Is it something that will show on playback? Thanks alot for all the help everyone
 
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Making wonderful headway so far, I did a bypass and it was the same. Went to the ASIO config and matched the sample rate of the focusrite and DAW and now I'm down as far as 64, with not much latency at all, if any. I think Reaper was trying to do its automatic call on the control panel for changes and only reaper was changing, not the focusrite, hence the crash(s). At least I'm really hoping that was the issue, because so far it sounds great.

Just so I know I've gone too far down, what does it sound like when you've gone down to far with sample rate? Is it something that will show on playback? Thanks alot for all the help everyone

It will not show on playback as long as it is at least 44.1 or 48.
 
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Is the sansamp doing digital processing? I thought those were analog...

Definitely analog. Sansamp into Focusrite is my recording setup as well. Technically I also run through my amp, but the Sansamp does all the heavy lifting.

Went to the ASIO config and matched the sample rate of the focusrite and DAW and now I'm down as far as 64, with not much latency at all, if any.

IME ASIO is the key if you're able to use it.

Just so I know I've gone too far down, what does it sound like when you've gone down to far with sample rate? Is it something that will show on playback? Thanks alot for all the help everyone

As @DirkP noted, there shouldn't be an issue as long as you don't go below the standard 44.1. I've heard some arguments for a 96kHz sample rate as it pertains to preserving transients when stretching or compressing samples. With that being said, Reaper has some cool settings aimed at preserving transients when stretching that I've only scratched the surface with myself - so it really might not be necessary after all.
 
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Still getting used to Reaper but I tried lowering as much as possible, didnt get artifacts but PC blue screened and force restarted. Only thing I changed was device>(checked "request block size") changed from 256 to 128, that caused PC reboot so I changed it back (I'm assuming that's where I lower it, the latency in the corner did show lower time before it restarted). I was under the impression if you ran the buffer low it just sacrificed the quality. Computer is a little dated, but nothing like a Celleron or anything.

Hopefully (and not) I'm remembering there not being a difference, I just dug out my preamp excited to play with a track and, through the DAW its simply not possible, fast at least. I did work out a way of listening to monitor in left ear and live from interface in right, I just have to align the tracks up after the fact
256 sample buffer size will be noticeable latency at 44.1k. 128 is passable, I run at 64 when tracking.
 
256 sample buffer size will be noticeable latency at 44.1k. 128 is passable, I run at 64 when tracking.

I've been using 256 (apparently) and haven't noticed any latency when tracking personally. I'm not arguing, as I'm sure you're correct, just trying to learn.

I did a test at one point, feeding my audio-out back into the input and recording the playback against the original track - lined up beautifully. Is this just the fact that the combination of Reaper and ASIO are good enough at self reporting latency and auto-correcting (when that option is selected)? If that's the case, it seems to negate the drawback of a higher buffer size fairly well.

Maybe I'm just being extra dense this morning, I only now just started drinking my coffee! ;) In googling the topic lower buffer rates seem to be consistently recommended - I'm just trying to figure out why, if the auto-correction seems to work so well.
 
I've been using 256 (apparently) and haven't noticed any latency when tracking personally. I'm not arguing, as I'm sure you're correct, just trying to learn.

I did a test at one point, feeding my audio-out back into the input and recording the playback against the original track - lined up beautifully. Is this just the fact that the combination of Reaper and ASIO are good enough at self reporting latency and auto-correcting (when that option is selected)? If that's the case, it seems to negate the drawback of a higher buffer size fairly well.

Maybe I'm just being extra dense this morning, I only now just started drinking my coffee! ;) In googling the topic lower buffer rates seem to be consistently recommended - I'm just trying to figure out why, if the auto-correction seems to work so well.

I'm not in front of my console at the moment, where is the auto correction?
 
I'm not in front of my console at the moment, where is the auto correction?

So... I don't have mine in front of me either, but if memory serves, it's under general options (not project options). I think there is a category there labeled record, and at the bottom of that screen, there should be a tick box to enable auto latency correction using the latency reported by the audio driver.

Something like that. I tried googling it real fast and it escaped me, but a few more minutes of googling would likely produce a few screenshots.

Ah - it's this:
reaperprefs.jpg
 
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Reduce the I/O Buffer Size? Bigger buffer size means slower latency times. 1024 sample rate is going to be slower than 32.This is adjusted in the DAW audio preferences.

And make sure you are using Direct Monitoring, which has near zero latency.
 
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