Hi! I conducted the following tests to determine my latency. Any thoughts on which measurement to use when offsetting latency w/in DAW (Reaper is my go to at this time) to get as close as possible to sample accuracy?

Reaper (latest version)
Mac M2 Mini/Sequioa
Audient iD14mK2 Interface

1. Loopback Test (UNCHECKED - use audio drive reported latency)
Sample Latency@48k/64=+366

2. Loopback Test (CHECKED - use audio drive reported latency)
Sample Latency@48k/64=-38 (negative latency . . . apparently over-reporting)

3. RTL Utility Test
Sample Latency@48k/64/429 (reported)/370 (measured)/7.708 (milliseconds)

 

This looks like a great start to a rabbit hole. Why do you need to know this?

 

Reduce latency . . . get it as accurate as possible with no perceived lag.

 

You cannot reduce latency without changing the audio interface, software, or settings in your DAW. The latency is a constant based on those factors. There is nothing for you to calculate. If you want to introduce a recording delay (negative or positive), that is a different purpose, over and above the latency compensation in the DAW. But in your prior posts, you mention wanting to record guitars, etc while multi-track recording. In which case, they will all result in the same latency anyway.

Round Trip ping measurements are used when sending audio from the DAW, through external hardware, and recording it back (aka Print Loop) . Like you would with an effects rack, console mixer, and so on. Again, prior posts suggest you are not using any external processing at all.

I'm not trying to give you a hard time. But I do think you are asking about something you have read something about and have not actually tested at all. That's why the term "sample accurate" has already disappeared from your concerns.

 

y tho? just make music brother

 

God bless RME

 

if your tracking analog instruments or digital plugins in a controlled environment is it really holding you back? Unless your playing Live and relying on your Gear thats 100% digital then I could i understand you chasing the fastest obtainable signal..but thats me

 

You need to be playing an external instrument, while recording the sound at the DAW and monitoring the output of the DAW playing back along with other channels while you play. Yesterday he was asking about how to use a Wave Editor as a multitrack recorder. Now he is sending Headphones Mixes? I don't buy that at all. This is looking at numbers on a screen that other people have told you are important. And very small numbers already. 7 milliseconds is close enough to zero it won't even matter.

 

Haha agreed, im with you 100% brother. To quote you on "This is looking at numbers on a screen that other people have told you are important"
i cant tell you how many times i have overheard or debated the same thing and experience and my ears once i learned to trust that and not waste time on "if'its the standard

 

I suggest ditching digital and going full analog if you really care about latency that can't be perceived. Otherwise, I am 100% sure even if you scrape half of that RTL you will not notice any difference. There are better excuses around for not using your gear (and it's a pretty good one at that, I have the id24, and people would kill to have one of these back in the day).

 

Most of the time that turns into nothing but a 5 page thread that is "let's compare audio interfaces again and show off how little we actually know". Maybe practice our typing skills with 100% useless answers.

I like this one, even though the very first post says it is a Mac.

But this one is equally useless. " I just stopped by to let you know I have a more expensive interface than you do".

All we need now is someone to tell us about the 30 years they have been doing this as support for some things which have no facts behind them. A clickbait trifecta in the works.

 

Well, all I'm trying to do is see (one last time) if I can get going on a DAW before going analog. While customizing, I realized there was a small latency. So, I'm trying to compensate. I could be LAZY and just say, "Oh, it's good enough." Or, I could make a small adjustment and have near zero delay. Hence, here I am . . . then, once set up and reasonably comfortable, I'll "JUST MAKE MUSIC."

 

Never needed anything below 7 ms for RT-tracking, which is pretty basic value for standard ASIO driver.
When I jumped into digital world with all the virtual instruments in 90's, my soundblaster got 25ms latency, yet I manage to handle it

 

But it is a Mac. His situation with REAPER and Audient interface is going to be using Core Audio, not ASIO drivers. There is nothing that can be done about choices of ASIO driver. The buffer is set as a value in the DAW. Using an "offset" is going to be in REAPER. He wants to add a manual offset for this. It's on the bottom of the Reaper audio preferences window. In Logic it is called Recording Delay, but it's the same basic thing.

You can use the reported latency detected/assumed by your DAW, and delay/advance the input/output individually, or do it manually. If you really think you need this, the only way to get it even close enough to be useful is to set values and listen/record.

 

LatencyMon checks if a system running Windows is suitable for processing real-time audio and other tasks.
www.resplendence.com/latencymon

44,1 kHz - 256 Samples

 

Funny how people didn't get past the third line or any of the multiple comments where it's pointed out it's a Mac M2. lmao

Maybe a Presonus Quantum can scrape down the latency, but once again, I believe the reduced latency is fully imperceptible in practice. 7ms is normal latency.

 

Thank You shinjiya!

Audient USB Audio: --> Studio One --> Input latency 7.30 ms --> Output latency 8.50 ms

 

in MacOS, you're stuck with CoreAudio engine (regardless of what audio interface you got),
luckily Reaper is able to explicitly request specific buffersize to operate (in its settings),
if I remember correct, default CoreAudio buffer is 512, which indeed is too much,

you have to use certain numbers (lower means lower latency, but questionable stability),
try 256, then 128, then 64, not sure if 32 would work at all - whatever lowest will work without any dropouts and CPU overloads (you should observe performance monitor in Reaper directly, especially RT %)

don't bother with latency offsets, Reaper is designed to automatically keep things in sync when recording,
you would only use offsets when combining various outboard gear, cameras, or using specific inserts, or certain plugins on input/monitor paths incompatible with anticipative fx processing

 

really depends what exactly you're tracking,
for ex. drums are loud enough to give you "zero latency", but electric drums setup with roundtrip higher than 10ms combined (input+VSTi+output) is unusable for tight beats in tempo,
or for ex. vocals, podcasts, woodwinds - those are also tricky because people usually hear themselves "from inside", e.g. inverted phase basically, so again 12-15ms roundtrip (input+output) what you probably call 7ms (either reported input or output) is borderline comfortable,
for guitars and other instruments, latency happens even on stage playing live (because those amps are not glued to your ears), but again in studio environment less is better (giving you more headroom for some plugin effects and amp sims which have some latency on their own)

 

Based on @Radio input, my input (id24) and the OP results, it's safe to assume 7ms is input+output. Mine is the same. This is in line with most other interfaces in the market. The most performant ones should be at around 5ms, which is just 2ms less. I think latency is currently as perfect as possible and thankfully we have zero latency monitoring for things like voice.

Edit: screenshot from Studio One