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Old 12-22-2009, 05:20 PM   #1
Lokasenna
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Default How to deal with interface latency (delay/lag)

I just did a bit of experimenting to get my system set up, all nice and perfect, and I figured the information might benefit someone else.

What is latency? In the realm of digital recording, latency is the time it takes for a signal to open the door, come in, take of its jacket and boots, and sit down in Reaper's living room for a chat. Most equipment in your recording chain, especially analog (non-digital) stuff, won't add a noticeable amount of delay. However, digital things often will, and your computer's audio interface will typically add a fair bit.

Depending on how powerful your PC is, "a fair bit" can mean anything from 5ms (well below what most people can distinguish, but still enough to drive hardcore studio nerds insane) to 500ms ("Jesus, this drummer is horrible"). You may have noticed this if you've ever used guitar software like Amplitube, Guitar Rig, Revalver, or any of the sexy free plugins floating around the internet. That weird half-second between hitting a chord and hearing it through the speakers? That's latency, and that's what we're about to fix.

Well, sort of.

The first thing you want to do is mess with your interface's settings, particularly the buffer size, in order to minimize the latency you have to deal with. Buffer size refers to the size of each bundle of information that gets sent to Reaper. A smaller buffer means Reaper will receive incoming audio faster, but it will also put more of a strain on your computer. Conversely, a larger buffer is easier on the PC, but will make Reaper wait around longer until each bundle is full.

The rule of thumb here is that you want the buffer size as low as possible without causing any hiccups, glitches, or stutters in the audio signal. Trust me, if it's too low, you'll hear it.

We should now have a reasonably quick signal path going into Reaper. If you plug in a guitar, whatever you play should be coming right back out of the speakers with a minimum of fuss. (If your system simply can't handle a low latency, your best bet is to monitor through a real guitar amp while recording a direct signal into the computer, and then put it through Guitar Rig afterward).

Go ahead and record something simple, a basic rock beat, the riff to "Smoke On The Water", whatever - something easy. Record it to a click, as tight as you can. Now have a look at what you've recorded. Chances are it's not as tight as you thought it was. In fact, if you haven't already spent some time dealing with latency, I'd bet good money that the signal is significantly off from the click.

I think it's time for an illustration:

http://stash.reaper.fm/oldsb/745259/...sation%201.png

Edit: Big image, so I've changed it to a link.

I can hear people asking "What the hell am I looking at?" Simply put, this is how you deal with interface latency.

When doing performing the tests described below, set the track's Record Monitoring to OFF, or you could quite easily mess up your speakers with feedback. I take no responsibility for any inadvertent damage.

The first track, labelled "Source", is a simple click (In Reaper, Insert | Click Source). This is the most accurate signal to use for examining your latency, since it's all tidy and perfect.

Track two, "Interface", demonstrates the latency my interface is adding. To measure this, I simply ran a patch cable from the interface's output to the input, played back the click source, and recorded it coming back in. As you can see, the signal is significantly late. Exactly how much is a question we'll get to in a moment, but I'm not quite done here.

Just for fun, I wanted to see if there was a notable delay being added by the speakers and/or the two feet of space between my ear and the speaker. "Speaker" was recorded with a microphone against one of my monitors, and as you can see it did add a tiny amount of latency. However, the difference is so small that it might as well not exist. "Chair", recorded through a microphone placed beside my head, shows a bit more added time - but anyone with some basic physics already knew that.

The speed of sound, in air, is 1100ft/s. As my head is in the ballpark of two feet from the speaker, the signal should take an extra 1.8ms to get to me. Guess what? It does. Now, in the grand scheme of things 1.8ms doesn't make a bit of difference. Nobody, anywhere, ever, is capable of noticing 1.8ms of delay. At least, not by itself. But, as a guitarist, 1.8ms is enough to annoy me. Why? Simple: Doubling.

Pretty much anyone playing hard rock or metal these days will double (or quadruple) their guitar tracks. That's just how you get that big huge guitar sound we hear everywhere. In most cases, guitarists want their doubled tracks to be as tight (played in sync) as possible. The farther apart the two tracks are, the sloppier the sound. A little sloppiness might be fine for AC/DC and the Stones, but with today's ultra-perfect, ridiculously fast technical metal, the bottom line is all about inhuman precision. Given how simple it is to fix, I'd just as soon not have an extra 1.8ms lurking in the shadows of my mixes, tempting me to poke around and clean it up.

That's all just my opinion, however, and in most cases fixing the interface latency alone will be enough to keep you happy.

So, having discovered this evil delay in our signal, how do we fix it? Thankfully, the companies who manufacture interfaces are one step ahead of us. Most interfaces can make an educated guess as to how much latency they're adding and pass it along for Reaper to work with.

Here's where you want to look:



Checking off "Use audio driver reported latency" will have Reaper move all of your recordings back by whatever amount the interface says. But does it work? Have a look at the fifth track, "Interface, Reported Latency". In this case the interface is clearly trying to do the right thing, but is nowhere near the right number.

Every interface is different. Some might be spot-on with their guess, others might be worse than mine. If this option works for you, that's awesome.

It looks like we'll have to solve this problem ourselves. In order to get a measurement of our latency, as accurate as possible, here's what we'll do:

1. On Reaper's toolbar, turn off Snap To Grid (the magnet icon).
2. Right-click the Transport bar and tell Reaper to give use Samples as a time base. (A sample is one digital "tick" of audio information, and is the most precise unit of measurement you can use)
3. Zoom in nice and close, so the original and delayed clicks cover most of the screen.
4. Very carefully, drag a time selection from the start of the original to the start of the delayed signal.

Your screen should resemble this:



On the Transport bar, the third Selection box tells us how long our selection is. In my case, with "Chair" as our target, the delay is 2160 samples.

Back in the Preferences menu, next to the option to use our reported latency, are a few boxes that let us do the same thing manually. Unchecking "Use reported latency" and entering your own measurement under Input | Samples will tell Reaper to ignore the interface, and just move everything back by whatever you tell it to.

Looking back at our first image, the final two tracks show the same tests as above, but now compensated for their respective delays. Manual compensation has clearly done the trick, which means any further timing issues are now entirely my own fault as a musician.

I think it's time for a drink.

Last edited by Lokasenna; 12-29-2009 at 12:17 PM.
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Old 12-22-2009, 06:23 PM   #2
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Thank you so much for posting this! I'll be checking things out tonight for sure
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Old 12-22-2009, 11:28 PM   #3
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Well, I'm all fixed up now. My Firepod's reported latency was was still ahead by 96 samples. With manual compensation I had to adjust it by 900 samples.

I did things a little differently, though. Instead of recording an instrument to a click, I just inserted a click source then connected a main output to a line input on my interface and hit record. Also when measuring, I zoomed all the way in (until you see the individual samples) and made my selection.


After that, I realized I could have just used ReaInsert's Auto-Detect feature. Though, every time I clicked Auto-Detect, it would give me a different sample offset. It seems to bounce around 895 and 900 (900 was the offset that I measured manually.)



Now that I feel all nerdy () it's time to record!


Thanks again for posting about this. I've been meaning to adjust my latency, I just never got around to it until now.
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Old 12-23-2009, 07:40 AM   #4
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Yes thanks for posting. I'll do a similar test with my home setup when time permits. This could be really useful.
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Old 12-23-2009, 07:51 AM   #5
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Great post! Bookmarked for linking when appropriate (and that's quite often!). Thank you!
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Old 12-23-2009, 11:07 AM   #6
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Awesome. Thank you for posting this. I just asked a latency related question in the Q&A Tips Tricks and howto Forum.
This covers what I asked and more!
Thanks again.
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Old 12-23-2009, 02:08 PM   #7
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Yay. Latency reported by my home card (m-audio delta 44) is reported correctly.

Next test will be the m-audio project mix and the allen & heath zed 16 at work... well after the holidays. Cheers .. (hiccough)
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Old 12-23-2009, 04:02 PM   #8
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Very KOOL!

Thanks
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Old 12-26-2009, 04:03 PM   #9
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Default am i doing something wrong?

Hi i tried doin your method,i did the time selection to see the delay, entered the number in manual offset but it dosnt seem to work for me no matter what number i put in (up to 2000000)theres still pretty bad latency.
Ive read over it a few times to make sure im not doin anything wrong and ive done everything you say .
can anyone help at all? Would be much appreciated
peace Splinta
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Old 12-28-2009, 04:35 PM   #10
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Thanks for sharing this : )
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Old 12-28-2009, 04:43 PM   #11
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Quote:
Originally Posted by mastersplinter View Post
Hi i tried doin your method,i did the time selection to see the delay, entered the number in manual offset but it dosnt seem to work for me no matter what number i put in (up to 2000000)theres still pretty bad latency.
Ive read over it a few times to make sure im not doin anything wrong and ive done everything you say .
can anyone help at all? Would be much appreciated
peace Splinta
Without more information, I'm not sure.

Would you be able to do up a screenshot like the ones I made? Or, if you want, save your test results as a project and upload it so I can have a look?
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Old 12-28-2009, 10:49 PM   #12
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Lokasenna.... veeerrrryyyyy nicely done!

And well explained, too.

What you did was one of those acts of genius (and kindness) that causes me to shake my head and say "Why didn't I think of that?"

Duh...

Anyway, thank you so much for taking the time to help in such a tangible way.



ZoD

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Old 12-29-2009, 04:17 AM   #13
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Great post, will surely be useful to link to for newbies.

One nag: Could you edit the big screen shot to not be so big... please... would make for much easier reading.

Thanks

EDIT: Many thanks for editing out the big pic, makes for much nicer reading. Kudos...
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Old 12-29-2009, 10:43 AM   #14
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Quote:
Originally Posted by mastersplinter View Post
no matter what number i put in (up to 2000000)theres still pretty bad latency
As explained in the original (great) post, the first step is to adjust buffer size to make latency as small as possible without causing clicks and pops. When that has been set, as good as you can get it on your PC, then some fixed latency time delay will always be present. It is important to make that latency delay as small as possible. When adjusting buffer size you really are reducing the latency time delay of your system - but there will always be some latency present.
That's OK though. See below...

On an old low-spec PC the lowest achievable latency may be massive. On a powerfull PC with a good ASIO driver the latency may be very small. Either way, you can record good in-sync audio. Having very small latency makes some additional recording methods possible. See below....

The second step is to tell REAPER what size the fixed latency delay is, on your system. REAPER can then make automatic corrections by time shifting each new recorded item so recordings are in exact sync with previously recorded tracks. As explained above, "Use reported latency" will use a value reported by the driver software. Entering a manual correction in ms or number of samples can be more accurate than the reported latency. Both options do the same thing though. They shift the new recorded material, after recording, backwards by some fixed time interval. This can exactly correct for the delay that was present when the recording was being captured.

Notice however that the system latency was still present during the recording process. This is where the limitations of having large latency and the advantages of having a very small latency are seen:

1) With massive latency...
The automatic latency correction in REAPER can still exactly cancel-out the delay, after recording. You could have a latency of 523ms, and REAPER could exactly shift new recordings backwards by 523ms.
Perfect? Not entirely.
You will still hear the latency delay if monitoring the new material you are laying down by listening to the new material, as you play it, through REAPER. That makes playing impossible!
The solution is to monitor your playing externally, before the signal goes into the PC for recording, before it suffers any latency delay. On an electric guitar input you will need a headphone socket at the guitar interface into the PC. Or, if playing acoustic guitar, or drums, or singing, recording through a microphone, then you don't need to monitor the recorded signal because you can hear the live sound. Listen on headphones to hear only the existing tracks being played by REAPER.

2) With very small latency...
Aha. This is very nice. Now you can monitor the new material being recorded by listening through REAPER. This means you can now insert some fx in a track and hear the results of the fx applied to your input as you play. You can record the dry un-fx'd signal, or the fx'd soud, or even both at the same time if you user track routing and arm multiple tracks.
As in case 1) above, some latency delay will still be present. If it is, say, 5ms or lower then it will be far too small for you to hear when monitoring - so it will not mess up your playing when recording new tracks. Also, when you hit the stop button, REAPER will shift the recording backwards by the tiny 5ms correction.
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Old 12-29-2009, 02:55 PM   #15
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Original post stolen (shamelessly) by myself for the Wiki, http://www.cockos.com/wiki/index.php...delay.2Flag.29

Nice one...
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Old 12-30-2009, 02:59 AM   #16
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Quote:
Originally Posted by MikeLacey View Post
Original post stolen (shamelessly) by myself for the Wiki, http://www.cockos.com/wiki/index.php...delay.2Flag.29

Nice one...
Good job, thanks.

I think the wiki is under-used and we should promote it more, that is, link to there instead of here, as well as move good stuff like this over there.

The "tricks" to disable wireless and ACPI seem to be missing from the suggestions list. Should these not be in there? For me, those have absolutely had the most impact on my latency.
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Old 12-30-2009, 04:25 AM   #17
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Quote:
Originally Posted by Fabian View Post
The "tricks" to disable wireless and ACPI seem to be missing from the suggestions list. Should these not be in there? For me, those have absolutely had the most impact on my latency.
Absolutely! You ok to add them in? PM me if you need a hand.

Mike
P.S. My little boy wants to know why your avatar picture is a red pepper
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Old 12-30-2009, 06:45 AM   #18
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Quote:
Originally Posted by MikeLacey View Post
Absolutely! You ok to add them in? PM me if you need a hand.

Mike
P.S. My little boy wants to know why your avatar picture is a red pepper
OK, I've entered that stuff, maybe there should be a special section for laptops...

Why my avatar is a red pepper? Cuz I'm a Chile Head, man... http://www.g6csy.net/chile/index.html
Here is a pic of some of the chiles I've grown this season http://forum.cockos.com/attachment.p...5&d=1253306835. Alas, it's winter here now, no more fresh chiles, but I get by... and my kids think I'm crazy with all the hot stuff, but I'm slowly converting them too. Little by little I spice up the food I cook them, and one day they'll be as hooked on the stuff as I am. Thankfully, it's not illegal (yet(?))...
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Old 12-30-2009, 10:58 AM   #19
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Interestingly enough, spicy foods are known to significantly reduce latency in other... areas...
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Old 12-30-2009, 11:23 AM   #20
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Quote:
Originally Posted by Lokasenna View Post
The first track, labelled "Source", is a simple click (In Reaper, Insert | Click Source). This is the most accurate signal to use for examining your latency, since it's all tidy and perfect.

Track two, "Interface", demonstrates the latency my interface is adding. To measure this, I simply ran a patch cable from the interface's output to the input, played back the click source, and recorded it coming back in. As you can see, the signal is significantly late. Exactly how much is a question we'll get to in a moment, but I'm not quite done here.
Shouldn't your audio interface automatically know and adjust the roundtrip latency? AFAIK, most do.

With the few interfaces I've used (EMU0404 USB, M-Audio Audiophile Firewire, Echo3G PCI), the latency is compensated for and the recorded "click" plays back perfectly in sync with the source click, no manual adjustments required.

Somewhat related: I tested a "prototype" audio interface for a company, and they apparently didn't know how to write audio drivers, the interface would not compensate for buffer latency after recording. So the recorded click would be off by a certain amount at 64 samples, and even more at 128 samples, quite ridiculous! They soonafter sent the interface through production, without fixing this issue (good thing it is an esoteric interface that isn't very popular). It makes a decent paperweight tho.
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Old 12-30-2009, 12:28 PM   #21
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In theory, yes, they should all adjust themselves. A lot of interfaces don't, though - my Tascam US-122, for instance. I think my 0404 PCI needed a bit of help there too, though not anywhere near as much.
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Old 12-31-2009, 03:20 PM   #22
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Quote:
Originally Posted by Fabian View Post
Cuz I'm a Chile Head, man...
Cool
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Old 12-31-2009, 03:47 PM   #23
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Quote:
Originally Posted by AndyNewton View Post
was being captured.
If it is, say, 5ms or lower then it will be far too small for you to hear when monitoring - so it will not mess up your playing when recording new tracks. Also, when you hit the stop button, REAPER will shift the recording backwards by the tiny 5ms correction.
But how do you get 5ms latency for monitoring and is this round trip or 5ms each way ?

The best I can get using my Edirol UA-25 is 17.9ms round trip and it's enough to through off timing when using direct monitoring through Reaper when recording vocals.
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Old 01-03-2010, 03:18 AM   #24
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Default MIDI Loopback?

Is there any reason to do such loop back testing with MIDI?

Why I ask is because I did, and I get strange results...

I have two ASIO drivers, and it seems that what ASIO driver I use affects the MIDI latency (as measured by a loop back). Using ASIO4ALL, I get a large variation in when the sent MIDI events are recorded, sometimes they even arrive before being sent! With the other ASIO driver (Lexicon Lambda) I get rather stable MIDI recording, fluctuations in the 50 sample range (~1.1 ms).

What I do not understand is why the chosen ASIO driver should affect the latency in the MIDI loop back...
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Old 01-03-2010, 04:37 AM   #25
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Default Time Travel via MIDI... Call Nobel!!

Ok... That is more than strange, and very interesting. When you say "large" variations - what range of numbers are you talking about?
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Old 01-03-2010, 06:07 AM   #26
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Quote:
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Ok... That is more than strange, and very interesting. When you say "large" variations - what range of numbers are you talking about?
Yes, strange, but not unheard of: http://forum.cockos.com/showthread.php?t=45801

With ASIO4ALL I get the MIDI events recorded between, -202 (that's early!) and 440 samples (at 44.1kHz sampling) of the time they are sent. Some numbers in-between just to show how it varies: -138, 62, 184, 314, 395. So, with ASIO4ALL the range is 642 samples, roughly 14.5 ms.

And just to make sure... this is MIDI loop back recording. Also, this is with ASIO4ALL latency compensation set to 0.

Using the Lambda ASIO driver, numbers are more consistent: 138, 161, 184 (that's it, of the 12 tests at 3 different buffer settings only these three latencies turn up). So, with Lambda ASIO the range is 46 samples, roughly 1 ms.
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Old 01-03-2010, 10:32 AM   #27
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As you say, not unheard of; recently as well.
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Old 01-17-2010, 10:40 AM   #28
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Just done this and I seems to have made a great difference to my ear
Thanks
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Old 01-18-2010, 06:27 AM   #29
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Quote:
Originally Posted by Basil View Post
The best I can get using my Edirol UA-25 is 17.9ms round trip and it's enough to through off timing when using direct monitoring through Reaper when recording vocals.
i get about half that (round-trip) in my Edirol UA-25 so you should be able to do better

Quote:
But how do you get 5ms latency for monitoring and is this round trip or 5ms each way ?
you know, the edirol has direct monitoring

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Old 01-18-2010, 06:34 AM   #30
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measuring round-trip latency is easy with CEntrance ASIO Latency Test Utility
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Old 01-18-2010, 03:43 PM   #31
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Quote:
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measuring round-trip latency is easy with CEntrance ASIO Latency Test Utility
All I keep getting is an error message, saying "The driver requested ASIO reset." It doesn't matter what I do, it keeps doing the same thing...
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Old 01-21-2010, 01:16 PM   #32
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Default Awesome Latency Response from Lokasenna

Hey LokaSenna!! Now, THAT was a thread reply!!! Awesome info, insightful images and very easy to interpret!! Actually, last night - I installed ASIO4ALL - not installing it initially was my problem in the 1st place...

I now can record a 2nd, 3rd, 4th track, etc after recording my 1st (Drum Mach) Trk... Just now getting a bit familiar with REAPER and the effects, plug ins, etc.

Will post any other questions - but in the meantime - I'm excited to get home 2nite and get some sh*& done!!

Thx Again!
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Old 04-16-2010, 07:10 AM   #33
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LokaSenna,

Thank you for a valuable method to fix recording latency, I am using us-144 and having this problem for years, now it is all fixed.
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Old 04-16-2010, 10:24 AM   #34
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in case you guys don't know, centrance ltu is accurate to half a millisecond. just something to keep in mind when testing, especially if your interface's latency seems to be inconsistent. doing a loopback test in reaper is more accurate, and it's not hard to set up.
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Old 04-16-2010, 11:22 AM   #35
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One more note about "loopback testing". What you are testing there is BOTH the input and output latencies. To see why, we must first understand where this latency comes from: your first post was exactly right that it depends on the buffer size. That is the biggest and most easily changed source of latency.
You must remember that the same system is used on both the input and output sides of the ADDA conversion. Thus, if a loopback test shows 400 samples of latency, then the input latency is 200 samples and the output latency is 200 samples (assuming a perfectly symmetrical system, which rarely happens).
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Old 04-16-2010, 11:29 AM   #36
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i might add that using the click source doesn't seem as accurate to me as recording the output of a tone generator and using that as the source. i was coming up 4 samples off when using the click source for loopback testing / setting manual offsets - i know that's not much, but we might as well do it right.
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Old 05-14-2016, 03:19 PM   #37
animal other
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When I try to playback a recording, it sounds very heavily delayed. I might play some 1/8th notes and it plays back as an ocean of notes... Help! I'm going to bash my machine with a bat!
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Old 05-15-2016, 09:48 AM   #38
solger
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Quote:
Originally Posted by animal other View Post
When I try to playback a recording, it sounds very heavily delayed. I might play some 1/8th notes and it plays back as an ocean of notes... Help! I'm going to bash my machine with a bat!
Which audio interface are you using?
And what are your settings in Preferences -> Audio -> Device?
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Old 06-11-2017, 10:05 AM   #39
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Quick question ya'll. From the click test, I determined my interfaces latency to be 767 samples.

I set this number in the Recording Settings -> "Output Manual Offset".

Do I need to put anything for "Input Manual Offset"? If so, how do I determine it?

It is my understanding that when this thread was made, there was only 1 setting which controlled both, now we have two! Yay!
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Old 06-19-2017, 03:38 PM   #40
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Bumpity! Does anyone know?
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