What SPLS Block Size in REAPER?

bouldersoundguy said:
But there's a better solution to new tracks being offset. There should be a manual setting that you can use if the automatic detection isn't right. The normal procedure is to feed the output of the interface to an input and play a short, sharp sound from the DAW (like a snare hit) and record it back to another track. Then you can zoom in and measure the precise offset and enter that value into the manual record latency correction. It may take a positive or negative value to make it right. It's a simple, though slightly tedious, procedure. If it works you won't have to nudge tracks to match.
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Strange. It is uncorrectable. I must nudge.

Screenshot 2021-05-28 001403.jpg
 
If you zoom in and select from an obvious peak on the original track to the same but offset peak on the copied track, it should show you the ms selected just above the transport controls. It only shows you this while you're actually adjusting the selection, which is annoying, but you can get the info you need there. Then go to Options -> Preferences -> Audio -> Recording. At the bottom of that panel you'll see a checkbox for "Use audio driver reported latency." To the right of that you can enter ms or sample values to manually adjust the offset. If your new track is 25ms late, enter 25 into the Input manual offset ms box. Record a new loopback sample of a couple of snare hits (or whatever, something with sharp peaks). Zoom in and measure again. Add or subtract the new amount to the value. If you originally put in 25ms but it's still 2ms late, change the setting to 27ms. Record a new clip, zoom in, refine the number. Once you're zooming in so tight you see dots on the waveforms, you can start adjusting by number of samples.

On my system (Win10, Scarlett 6i6 first generation) I found that letting the driver report the latency still left new tracks 4 samples late, so I added 4 samples to the Input manual offset box and it came out perfect.
 
Manually changing the numbers effects it. I will need to start over again on something..Try again fresh.

When I laid three or more tracks that latency added up. The third would be farther from the second, as I recorded listening to the monitored playback.
 
Last weekend I was gonna record a bit, but get this...My Asus laptop's touch pad felt hot. I smelled burning plastic. It was in for servce 3 times since June. Hinge failure, VIA board failure, Battery and harness failure, and now this..Still under warranty I hit the ASUS now button and printed a label. Yersterday I got an email saying that my unit is not up to ASUS standards and it is being replaced with a new one.

This is the one they are sending me back as the other is unavailable.

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See anything that would make it not record music?
 
My last effort I got two to line up. SPLS block setting does nothing. It is the latency adjustments.

What about recording analog instruments to an analog medium first, like 1/4" tape? Then bring it in to the DAW. To try and fix my thin sound. I admit I am HPFing it to counter proximity boom. 40-120hz. But even before the HPF it is thin, but boomy. How is that combo possible?

Another garbage recording that sounds like my crappy room, but worse.



Does a hardware mixer change the sound? Set flat.

Are you using a gate? For me it sounds like gating keeps the track contents under pressure. If you follow. It does not let the air out.
 
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Thin? are you referencing your guitar tone? Or the vocal, or both? Not hearing any room sound in the guitar - but definitely in the vocal. Listening with decent earbuds, there is no boom, but the guitar tone is not great. Does the recorded guitar tone sound like what you hear in the room? If not, then you need to work on mic positioning.
 
There is nothing wrong with the sounds, if they were in a full mix. Bass and drums would fill it out. These are the types of sounds you need to get the instruments to cut through a dense mix..but this isn't a dense mix.

However, if you want it to just be guitar and vocals, you are going to need some lower mids.

I'm wondering if your monitoring isn't lying to you. You seem to complain about boomyness and proximity effect a lot more than average and your mixes are always really bright. That's what is making me wonder about your monitoring situation.

What are you using?
 
Farview said:
What are you using?
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Monitors? The daw is output to a Klipsch studio system. Klipsch is well known high quality audio.

The PA is monitored on 2 way cabinets horns and 15", so monitors are much different from my 8 track to the interface and DAW. I commonly refer to the 1/4" as sounding fuller. I thought it could be the Mackie mixer.
 
When you guys sing/play, are you looking at the meter? I am. Usually I would just belt it or bang it out. Contantly I am trying to modulate my volume. good ?
 
I wouldn't use HPF to counter proximity effect. The curves don't match at all. I would use a low shelf. Here's one example of a proximity effect curve.

w100_beta57_2sc.jpg
 
LazerBeakShiek said:
Monitors? The daw is output to a Klipsch studio system. Klipsch is well known high quality audio.

The PA is monitored on 2 way cabinets horns and 15", so monitors are much different from my 8 track to the interface and DAW. I commonly refer to the 1/4" as sounding fuller. I thought it could be the Mackie mixer.
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So, you are using home stereo speakers and PA cabinets? It's no wonder you are having a hard time. Home stereo speakers are made to sound good, not be accurate. That means different things for different companies, but mostly an emphasis on deep bass and high treble.

PA speakers are designed to throw sound as far as possible and fill up space, and again, not to be accurate.

You have a monitoring problem, which is getting in the way of knowing what is actually getting recorded. This is causing you to go down all these rabbit holes, trying to fix things that aren't necessarily broken.

Also, I agree with BSG. You don't need a high pass filter for proximity effect, you want a low shelf. You want to dial back the extra low end, not completely get rid of it.

When you set recording level, make the loudest noise you can with the instrument/your voice, then adjust the level so it doesn't clip. Don't try to modulate your volume to accommodate the recording level, set the level to accommodate your volume.

A mackie mixer will not make anything sound 'fuller'
 
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