Recording Latency Manual Offset in REAPER
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- Опубликовано: 25 авг 2024
- If you’ve ever recorded something thinking that the timing was perfect, only to find that on playback its all a bit off. Well it could be you need to adjust the recording latency offset in reaper’s preferences.
Whenever you record from an audio input, reaper uses the audio device drivers information about latency to automatically put what you recorded into time. It’s usually pretty close but if the driver isn’t telling reaper the right thing, the timing will be slightly off.
In this video I'll show you a simple way to detect and correct any additional latency your interface isn't reporting.
Text version of this tutorial is on the blog:
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over 2 years later and this video is still a life saver. thank you so much. i thought i was going crazy for a while.
3:04 "We don't want to make a feedback loop."
Me: Suddenly wants to make a feedback loop
I plugged in old speakers I don't care about... it was LOUD. could definitely tear a speaker on it.
OMG thank you so much... you resolved all my latency issues... there is no way I would have sorted it without this video. MANY thanks!
I didn't have the time or cables to try the loopback test. So I went with 65 samples and set up everything else the way you said. I have a Scarlett Solo interface. I was absolutely blown away. I detect no latency, or at least it's not enough to bother me. Before making the changes it was terrible. Thanks so much!
I just put the value of 65 to check if it will eventually work... and it did... perfectly. I deleted this annoying delay accidentally i would say. Thank you, sir! :)
So, I got my 18i20 all set up and started recording. A ton of latency. Watched this video, followed it step by step, BAM! Zero latency!(that i can hear) Thanks for the hook up!!!!
Did you make the input gain 0?
It's incredible how complicated DAWs are. I feel like Forest Gump over here.
You don't have to go through this crap with other DAWS. It's just reaper being overly complicated and hides simple functionality behind a hundred steps. RUclips how to fix latency issues on the specific DAW you're on if Reaper isn't the one you're using.
@@seraglukemicky19 I am using Reaper, but have not bought it yet. It allows you to try for free.
@@seraglukemicky19 Are you sure about that? What if I told you that most of the DAWs don't care if your recording is 37 [with my TC SK48] or whatever samples off? It's just that Reaper gives you an option to have everything correctly recorded to a 1 sample, especially if using external hardware like I do. "Pro Fools" is notorious about having all kinds of syncing problems, for example. It's a piece of shit DAW. I wonder what DAW you're using that is so perfect. Try the hardware feedback loop and see for yourself how good it is. ;)
@@dddux Lol, I use Abelton It is significantly better. It allows me to reduce samples at a good enough level to record without any issues. Most people don't really care about reducing samples to 1 they just want no latency.
Reaper has forced me to close and reopen it so many times its not even funny.
Production isn't an elitist's niche anymore it's become accessible to everyone so things such as solving latency issues and basic functionality should be easily solvable and not hidden by a hundred other steps. In addition, DAWS have become less of an Engineers playground and is now sharing said playground with composers and beatmakers who's aim is to write, record AND mix not JUST mix so they are designed for those people in mind, unlike Reaper which looks to be designed specifically for engineers. Imagine Deadmau5 trying to use this thing lmao. Doesn't he use a lot of hardware too? I wonder why he doesn't use Reaper. Probably because he isn't interested in wasting his time going through several things just to fix something that you can fix in the preference section alone in Abelton.
That's a problem with Reaper. It advertises itself as a cheap/free alternative to what's out there and in doing so you market yourself to broke beginners/intermediate-level not experienced people who are interested in the detail and little things that give you more "freedom" than other DAWS. Who cares? We'd just like not to gouge our eyes out trying to figure out something as basic as latency lol.
It doesn't matter how much freedom Reaper gives you in the long run or when you learn it to its full capabilities. The type of shit i mentioned is exactly the things that put this DAW on the bottom of many people's lists. You don't see Rick Rubin, Phinneas O Connel, or any other award-winning producer use Reaper everyone's using Abelton, Cubase or Logic so there goes your "perfect" DAW
@@seraglukemicky19 I've tried Abelton lite, isn't really for me. I don't think any DAW is really "better" than the next (except crap like FL or Bitwig) they all pretty much get you where you need to be as long as you are proficient at them.
I couldn't figure out how to check the routing button.(PC)
Lucky for me my main issue was a stupid one. I just needed to use the correct ASIO driver for the 2i2.
I'm sure its still off by a dozen samples but I can live with this for now.
Edit:
I figured out what I was doing wrong, I made the assumption that Input was the first pull downs in the ReaInsert, once I swapped those it worked.
Thanks for your technical expertise, liked and subscribed!
really cool. I always did it the way you initially wanted to demonstrate (with zooming in). This video, of course, shows an easier and faster way.
Thank you Brother! you resolved my problem about latency issue on my reaper using guitar.
Here it looks like this:
System: Win 7 (Ultimate) x64
Processor: AMD Phenom II X2 560 3.30 GHz
RAM: 8.00 GB
DAW: Reaper v5.963 x64
Interface: Scarlett 18i8 (1st generation)
Driver: 4.54.104.715
Scarlett MixControl: v1.10b3
Firmware: 1056
ASIO:
16 = -88
32 = -89
64 = -89
128 = -88
256 = -89
512 = -88
1024 = -88
Thanks
Thanks man, I was kind of frustrated and found your video, 5 minutes later, problem solved.
dude thanks ,I was actually getting so depressed thinking my timing is sooo awful and there's no way to fix it because in my ears it sounds on time but the recording says otherwise and it turns out it was all because of the recording latency omg...
Yeah, that latency is ruining modern rock music , you can hear it in many recordings
same here, been playing for 18 years and i thought my timing was terrible, now i found why... 85 samples offset and now i'm doing much better. it's liberating to know i'm not utterly useless
Thank you so much!!! The "Request Block Size" hint works perfect
Thank you so much! You solved my problem. Windows 10 had changed my audio setting to WAV and the latency was horrible. Changing it to ASIO solved my problem. Many many thanks to you!
thank you. mine was -45 on the latency ping test. now it's perfect!
Wow, this was so helpful. I had so much problems with the waves. This is a game changer! Thank you so much!
oh man...! t
that is so much less of trouble than the other method, wich is truly a pain and made me revisit videos on this for the umpht tie. So glad I clicked this one first now, I can skip the rest now, this is absolutely superio and evaporates all the complicated bottlenecks
Excellent tutorial. Simple and straightforward. Many thanks.
Thanks much. I had been meaning to do this. I have a Focusrite Scarlett 18i20. I first tried using channel 3 -> 3 and the ReaInsert didn't detect anything. Then I switched using channel 1->1 and it worked as you said. Going to try recording something now and see how it works.
Just what I needed. Thank you
This was super helpful!! Thank you so much! 🙏🙏🙏
This is such a good implementation of this..... its so annoying in Pro Tools to do the same thing.
Nice video, well explained Someone finding this probably has some latency issues in Reaper, it might help to note you are discussing only one aspect of latency: the latency attributed to converting audio sound to digital and back again.
Anyone experiencing latency issues caused by plug in processing .... that is a separate issue. Can you do a video on PDC (Plugin Delay Compensation) in Reaper, how it works and when to use it and when not to? (ie what does "Preserve PDC monitoring in recorded items" mean exactly)
THANK YOU SO MUCH!! Instant fix. Thumbs up 100 times
very usefull info. thank you, Jon!
Thanks for the tutorial man. Greatly appreciate it!
Thank you, everything is clear and without much chatter.
Thanks a million!!! 👍👍Very much obliged!!!!
using a rocksmith cable. it worked great a few year ago but since i reinstalled reaper to a new version there is a .5 sec lag and offsets dont work at all.
I also use a Rocksmith cable and I noticed a small increase in latency with the latest three or four versions, but only when using certain plugins. For instance, Amplitube gives a lot more latency than Guitar Rig, for some odd reason
Hi, I’m a bit confused to as why when I run the test, sometimes I’ll get a reported ping of ~17 samples and other times close to 530. Which number should I adjust for and why would this be happening?
What if I have a negative sample rate?
Thanks! My setup just got 56 samples more accurate.
I have a problem. When I do the test, it says "0 samples", but I still have latency. I'm using FL Studio ASIO as my audio device. Any clue?
RME FF 800 @ 44100, 512 samples buffer is 105 samples.
love this one!
I sure wish there was a way to create an action to automatically change the requested block size, instead of having to open preferences every time.
Just using nudge set to milleseconds or samples to find the offset value worked better for me because my midi stuff would ping different values.
I have a problem with only midi imput latency, can I solve the problem with this? I tried it but didn't work
You my friend have saved my life
Good stuff nice and to the point :)
I am using Wave Out as my Audio Device, when I turn it to ASIO, there is no sound from my plugins, is it supposed to be like that?
Everytime I re open reaper on a completely blank project. Following this video exactly. The ping detect gives me a different number.
I use the wasapi driver on windows on that gives 0ms latency which is super cool.
My question is this - if you are ping testing a loop, aren't you seeing the sum of input and output? Meaning, some unknown number of samples latent on the sending and receiving path? Does it matter in the end? For instance, should you put 32.5 samples offset in send and receive?
Correct. My reasoning is, you are always reacting to what you hear from reaper, so if there’s a delay you play late, and the input offset fixes that.
Thank you so much!
when you used audiofuse, did you use any cable in this video?
so this latency fix method just need a cable?
Thank you so much! Works
Help please!! I use a USB mic with asio4all and I noticed the things I record are a bit early...how would I fix this?
If i dont have a TRS output what should i do?
Does it changes in busy project with a lot of plugins which have big latency?
I think REAPER adjusts for that after the manual offset is set. I haven’t checked
Thank you for this
Nothing happens when I press ping detect. Can't figure it out
how accurate is the auto delay compensation feature? for me it worked ok i guess, at times maybe i had a slight feeling of a delay but so small i didnt know if it was just in my head. would you recommend to always set it up manually for real exact precision?
Is it possible my difference was just 4 sample or I did something wrong?
I'm having trouble with this. Maybe because I'm using an older interface. I have a Lexicon Omega, which is still awesome for it's age. But following these steps, I end up with no feedback from ReaInsert. That is, in the Additional Delay Compensation box, I get ZERO when doing the ping test. I don't understand how it's supposed to work anyway, when I'm turning the input level completely down. But the Lex has it's own way of doing things, and when you're saying input level, it may be different on this interface. It has 4 inputs (1-4) all with their own separate input level knobs. Each of these channels can be assigned to USB input 1-2 OR 3-4, which one stereo group can be selected in Reaper. Lex has Out L & Out R. I took my cable and connected Out L to Line 3. In ReaInsert, these are the In/Out I choose. By your instructions, I put the input on 3 down completely, output is up all the way... nothing. how is it supposed to detect the signal if the input is down? don't I need the ping signal to be received into the line input so that it can measure the delay? is that not the point of connecting the cable from output to input? So I lowered the volume of the output and raised the input just slightly so that I don't get a feedback loop; just enough to get a signal through. I hit Ping test.. and it works. but man it's high.. When I set Lex at the LOWEST latency possible setting, I get 547 samples. So last question.. is that accurate? guess I'll punch it in and find out.
Mine came back a value of -39 samples. Does that mean anything, or can I follow suit?
Thank you!
I need help: My audio interface's (IK Multimedia Axe I/O) reported latency constantly chnges.
I've tested using the RTL utility, reainsert, as well as just recording a kick sample from the input to output of the interface and checking the difference in samples on the timeline in reaper. And with these individual tests there can be a difference of up to 80 samples the next time I run it.
This is all usually at 48kHz sample rate and 64 buffer size. So that 80 sample difference translates to a difference of over 1.5 ms latency.
Is this normal? If not, does anyone know what could be the issue?
is it possible that maybe I just got used to my interface's reported latency? I have tried doing this and kenny's way many times and I get it to be sample accurate and all that. But whenever I play, I find that I play earlier lol. Not sure if muscle memory. Today I spent all morning trying it out, I wanted to give it another chance and woop, I play more on grid when I don't manually offset
Weird! I guess just leave it alone then :)
@@TheREAPERBlog yea :(( this is probably why I suuuuuuck when playing real amps. I play super early
I have the problem when I do this on my UAD apollo twin X with antares on the insert its always a diffrent sample amount. Always around 200
Nice help man, but why by me was pink "-1" in one input of my soundcard, and in another "-18" samples. I have more channel for input. I have Roland Studio Capture. And I'm interested in how midi pink works, i have many external midi devices ? My midi latency even though it's about 6ms sometimes.
Thank you SOOOOO MUCH!!!!!!!!
Very helpful thanks :)
Could you teach the manual time selection ?
I tried but just displays zero (0) delay samples after I hit ping detect. I wonder why. Thanks for sharing anyway.
Been having latency issues over several years. Seems like the latency can vary from different sessions / computer boot ups. Any comment on why the latency can vary? Do you check the latency each session? Also, do other DAWs also suffer from these issues?
I have the same problem. Can't figure out what's going on.
I'm using a Blue Yeti USB mic and it has latency. I have no idea how to fix it because nothing I do changes it. I'm using the DirectSound driver.
DirectSound driver sure has a "nice" latency to it. It can't go lower than 512 samples. Hasn't Yeti got an ASIO driver? That way you could lower the latency heaps.
I got a "-60." What does that mean?
I did everything it said but when i run the test its 0. Im using a scatlet 2i2 as well. Do i need to have that audio fuse thing?
Helpful video but I am getting a negative number when I click PIng Dedect and it keeps changing so I can't get it down to zero! What am I doing wrong? FYI, im using a Behringer UMC 1820 Interface. Many thanks
Super video.Thasnk
Thank you 🙏🏽
I tried to adjust the output latency but it doesnt change anything; the monitoring is still super delayed. Any idea why?
Will it be the same result if I use loopback function inside my interface (built in), I use Steinberg
I have a scarlett 2i2 but it's RCA out... i have adaptors but nothing balanced.. Would i be safe to assume my result would be pretty much the same at 128/44.1 .. don't want to fry my input stage
In my case with Reaper and A UMC 22 interface the number is a negativ number, -17 and if I apply it to the box I would ear only in mono afterwards on my headphones.😬
Pod X3 Pro, buffer size 128 : 699 samples latency :)
I'm trying to follow these instructions but have encountered some problems. First, my track doesn't have a route button so where do I uncheck the master send? Secondly, when I enable the Cockos Reinsert, "none" is the only option in the Hardware Sends/Returns fields. How do I set them? Also, I'm confused by the references to the AudioFuse. I'm using a Scarlett 2i2. Anyway, sure would appreciate some advice!
It also works when I record more than one track?
When I record more than one track, some tracks timing have delay and some others don't have delay. Does anybody know what I'm supposed to do to fix it?
My audiobox comes out to 0 additional samples but when I sync my drum machine it's off atleast 1324 samples. What's the fix I got an arturia drumbrute impact as the master. It only works as a master using spp won't work on mtc.
Hi Jon, just wanted to ask you, do you have any great videos on recording midi bass or writing in in the MIDI Editor I want to create a track using Moto bass. Thanks. Matt
Is it normal to have different offset values with different buffer sizes at the same sample rate?
Nevermind I got it worked out....
if I reduce the request block size, it always corrupts my uadiobox usb driver.
Someone help bruh I just upgraded from usb mics to xlr and I’m hearing delay when I play my vocals back and I don’t know how to fix this.
Decrease ASIO latency AKA "Block Size" to 128 and see how it goes. My card has direct monitoring turned on, so when I sing or play into it there's 0 delay. When you're using a DAW for processing your vocal during recording, it adds ASIO latency plus the latency of your AD converters and that's what you hear, especially if your ASIO latency is 512 samples or bigger. 64/128 and even 256 samples is rather imperceptible.
What if you have audio files already recorded in an old DAW that you want to move to Reaper, but you need to manually set up different offsets for each take because they were different in the original DAW? How do you line up all the audio files correctly in Reaper? I have been trying to find how to specify the exact position in samples (e.g., measure 1 should start at sample 9024 of Take 1, sample 8096 of Take 2, etc.), but so far haven't found it. (Alternatively, if I could cut/delete the excess samples at the beginning of all the tracks for a given take, that would also work--but I couldn't find that, either.)
i need help ,,for latency problem ?buffers x sample
Hi, i Just pruchased reaper and im trying to correct the input. It doesnt matter how much i compensate, the delay is still exatlcy the same. I have the scarlett that is in your video. What am i doing wrong?
Bruh how do yhu do it without the cable ik there’s a way
Sorry I am a complete noob here. Why do you have to plug the jacks in loop for the focusrite audio interface? I see that you did not even connect this device to your computer!
i have a umc 18 interface using it on krk rokit 5s. I am having issues with setting this up and it is really a pain. please help. anyone know what the offset setting is? i am trying to record drums properly and i cant. :(
What if you do all that and it keeps coming up with that number?
Hey man, what if you're having the same issue but it's with a midi device only and your audio system is set to windows WaveOut? I seem to be having an issue where my recording of midi notes will be played on time but on playback it's not in time
💕💕
Is a negative 370 a normal number to come up or am I doing something wrong. I know its a noob question but I am a noob 😅
My interface has loopback built in. Can I use that? Or is connecting a cable more accurate?
I would expect that to be zero latency, since loopback wouldn't use the ad/da
Legend
Coul you tell us what the song of the begining is?
My processor is a 6700K and I can't seem to go below 512 samples.