Latency

LeadPaint

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I have found out that I get a 12 ms latency when monitoring my guitar through my computer without any VST's or other effects running.

Interface: Tascam US-1800 (latency set to lowest option; 49 samples)
Computer: Windows 7 64-bit, Intel Core i5-750 2.67GHz, 8 GB RAM, Samsung 850 PRO SSD


1. Is this a correct way of measuring latency?
  1. I plug my guitar into my interface.
  2. I set up my DAW to monitor the guitar.
  3. I connect the headphones out back into a line in.
  4. I simultaneously record the straight guitar and the "phones out > line in".
  5. Between the two recordings there is a difference of 12 ms, both when using guitar amp simulators and monitoring the dry guitar signal.
I have attached the recording. Left is straight guitar. Right is "phones out > line in", first without effects, second with Amplitube, third with LePou amp sim + cab impulse loader.


2. Is there anything I could do to improve this? First I thought it was the amp sim that was causing the latency, but I get exactly the same latency when monitor the dry guitar signal.


3. How much latency do you find I acceptable? When playing fast stuff, 12 ms makes me feel slightly out of time.
 

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If you work on 1ms being a foot, then I think it means more to my brain - I'm having to put up with a 22ms delay on one pice of kit at the moment, and it's a little annoying. Normally it works out on the screen as 11-14ms, and this I can work with. 12ms is quite short, and if it's causing grief, it's probably being increased by something else in the system that isn't registering - as 12 feet from a guitar amp is hardly unusual - so maybe something else is delaying the audio and not registering?
 
You can disable tracks you're not working on, and it might help with latency. Also setup your PC for recording -- there are instructions online for optimal settings just google it. I'd upgrade to 16gb of RAM if $ allows and your pc takes it.
My setup usually gets 3ms input latency, but last night on a busy track I noticed it was 23ms.
 
You can disable tracks you're not working on, and it might help with latency.
That shouldn't actually change the latency. It might allow you to get away with less latency, but you'd still have to change that in the interface settings.
 
You can disable tracks you're not working on, and it might help with latency. Also setup your PC for recording -- there are instructions online for optimal settings just google it. I'd upgrade to 16gb of RAM if $ allows and your pc takes it.
My setup usually gets 3ms input latency, but last night on a busy track I noticed it was 23ms.

In the test recording I made there were no other tracks.

I'll try and find some instructions for optimal settings, I hope that gives me a couple of milliseconds. But I wonder if that will change anything if the latency is caused by the conversion of analog to digital and back to analog, I wonder.
 
In the test recording I made there were no other tracks.

I'll try and find some instructions for optimal settings, I hope that gives me a couple of milliseconds. But I wonder if that will change anything if the latency is caused by the conversion of analog to digital and back to analog, I wonder.

Check how many windows services are running, etc. Disable internet when recording. Windows runs a ton of awful/boggy services that aren't even needed. They're all going to use resources. Does it affect your latency? Maybe.
 
Check how many windows services are running, etc. Disable internet when recording. Windows runs a ton of awful/boggy services that aren't even needed. They're all going to use resources. Does it affect your latency? Maybe.

Before a recording session, I close down everything on my machine, eject any external drives I don't need, and turn my WiFi and Bluetooth off. Then I restart. When the computer boots back up, I load only my DAW.

Amp sims, like all plugins, will increase latency. Good amp sims will significantly increase it because of the amount of processing they do.

Still, I got fed up with latency as well. I started using Audiobox VSL software about 2 years ago. While it has some bugs, the "near-zero latency" with monitoring effects is magnificent. But that wouldn't work if you're tracking using an Amp sim (the VSL software monitors at the USB port, not in the DAW).
 
Also, what is your buffer size? That's probably what's causing it.
 
Before a recording session, I close down everything on my machine, eject any external drives I don't need, and turn my WiFi and Bluetooth off. Then I restart. When the computer boots back up, I load only my DAW.

Amp sims, like all plugins, will increase latency. Good amp sims will significantly increase it because of the amount of processing they do.

Still, I got fed up with latency as well. I started using Audiobox VSL software about 2 years ago. While it has some bugs, the "near-zero latency" with monitoring effects is magnificent. But that wouldn't work if you're tracking using an Amp sim (the VSL software monitors at the USB port, not in the DAW).

Well, the latency is also 12 ms when monitoring the dry guitar signal, so it's not the amp sims that cause the latency.

---------- Update ----------

Also, what is your buffer size? That's probably what's causing it.

Buffer size is 49 samples, the lowest option with my interface.
 
The US-1800 is capable of true zero latency monitoring, but you can't use it with live sims. Here's my suggestion:

Get a Pod and a DI. Run your guitar through the DI to the Pod and send both to the interface. Use the Tascam's simple input monitoring to listen to the Pod while recording. If you don't like the Pod's tone you can reamp the direct signal with a sim.
 
^^^You'll end up hearing both the Pod and the DI sound while tracking. The Pod might be loud enough to mask most of the DI, but I'm reasonably sure that it wouldn't be acceptable to me in most cases. Certainly worse than a delay equivalent to standing halfway across the room from your amp.

I play live through my US1641 and amp sims all the time. I regularly run other people through the same system (yes, four or five at a time) and nobody ever complains about latency. Occasionally a very sensitive vocalist might feel it, but almost never, and those that do just think I've got a bit of reverb or something on the voice.

12ms does sound like a lot. 49 samples at 44.1K sample rate is 1.1ms. Double that for input and output and you should still only be at 2.2. You've installed the ASIO driver, and set it to lowest latency in its control panel, but have you specified in your DAW to use that driver and are you sure the DAW is not somehow overriding or "requesting" some other buffer size.

As mentioned, buffer size is the only factor that actually affects latency time. Even with a bunch of tracks and a slow hard drive and a bunch of background processes on a slow CPU with barely enough RAM, it'll either work at the latency specified by the buffer or fall apart into stuttery, clicking and popping and dropping out mess. It will not just sound fine with more latency unless you specifically go change the buffer settings.


Edit - Well, yeah... Some plugins add latency, but it's not actually completely necessary except for those that need to "look into the future", or really do a whole pile of processing. Neither PodFarm nor GuitarRig add latency of their own from my observations, and the OP seems to have shown that (his version) Amplitude doesn't either.
 
^^^You'll end up hearing both the Pod and the DI sound while tracking. The Pod might be loud enough to mask most of the DI, but I'm reasonably sure that it wouldn't be acceptable to me in most cases. Certainly worse than a delay equivalent to standing halfway across the room from your amp.

Isn't there a mixer for the monitor output of some sort? Otherwise, how can you get a decent headphone mix without compromising the recording levels?
 
^^^You'll end up hearing both the Pod and the DI sound while tracking. The Pod might be loud enough to mask most of the DI, but I'm reasonably sure that it wouldn't be acceptable to me in most cases. Certainly worse than a delay equivalent to standing halfway across the room from your amp.

I bet you could run the direct recording level a bit low and the Pod a bit high. A distorted tone tends to be naturally compressed so it can be run with a hotter average level without clipping. But, yeah, that could be a deal breaker.

The reason I didn't think of that hangup is that I generally have a big analog mixer in front of my interfaces which allows me to control monitor mixes in analog. But since mixers are bad for recording then the excellent, latency free monitor mixes they make possible must also be bad.
 
Isn't there a mixer for the monitor output of some sort? Otherwise, how can you get a decent headphone mix without compromising the recording levels?

There isn't. It just passes inputs to the input monitor bus as they are and you get to balance "all the inputs" against the playback using the mix knob. It does offer true zero latency monitoring because it's an analog path.

Honestly, for a 2-channel interface this is pretty much doable but for a big interface like that it's pretty stupid. I know some here do just fine with them but having no more control than that for the monitor mix is a deal breaker for me. Then again so is input monitor latency. Anything over 5ms is unacceptable, especially when 0ms is easily attainable.
 
If I would get a Pod, I could monitor straight from the Pod and have no latency issues.

[...]

12ms does sound like a lot. 49 samples at 44.1K sample rate is 1.1ms. Double that for input and output and you should still only be at 2.2. You've installed the ASIO driver, and set it to lowest latency in its control panel, but have you specified in your DAW to use that driver and are you sure the DAW is not somehow overriding or "requesting" some other buffer size.

[...]
Reaper is set to use the interface's ASIO driver and to not request a buffer size. I've attached a screenshot.
 

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