ASIO latency

[giallanon]

Hello. I'm using HD500x and reaper to record some stuff (and slow it down to listen to my mistakes)

Reaper says I've a latency of ~95/190 ms and, to me it seems way too much.

For example, at 150bmp 16th notes, you're going to play 10 notes per second that means 1 note every 100ms that means that with 100ms of latency, I'll be always "one note" behind while recording.

 

How is your latency? Maybe I need to download something to upgrade the ASIO inteface

[Rewolf48]

Are you using the ASIO drivers?  These should latency between 3 and 10 ms using Reaper; the 100 to 200 ms are what you typically get using the Windows Sound drivers.

 

I just noticed you mentioned updating the ASIO drivers, but the latest Line 6 ones are installed by Monkey so this is hard to get wrong, but you can muck things up using ASIO4ALL. These times are so long that it look as though Reaper is simply using the wrong sound connection

Edited July 22, 2015 by Rewolf48

[giallanon]

I'm using the HD ASIO driver that's why I was asking.

Anyway, switching to ASIO4ALL drivers, I get a latency of 2.5ms.

Problem is that with ASIO4ALL enabled, I can't hear any other sounds, just the guitar and the output from reaper (and this is not a fault, it's how ASIO4ALL works).

 

But if you guys have nice latency using HD500 ASIO driver, I guess I should too... I'm gonna investigate further

[giallanon]

 

PS.

I gave a look at my Reaper (v4.78) and in upper right corner says: 44.1kHz 24bit WAV : 2/2ch 256spls ~9.1/18ms ASIO

 

 

With POD ASIO driver selected, mine says 44.1 Khz 320kbps MP3 2/2ch 4096spls ~96/190ms ASIO

 

Quite different, expecially the 4096 spls

[giallanon]

Reducing the 4096spls to 256, improved the latency to ~8.1/16ms

Thank you

[phil_m]

If you're monitoring through the POD, there shouldn't be any more latency than you would have if you're just playing through the HD itself. How are you monitoring?

[NucleusX]

A sample buffer size of 4096 is going give anyone grief, anything over 512 will. Keep in mind you will have

to increase the buffer size again if you decide to increase the sample rate to say 96khz. If you set too

low, you will hear glitch artefacts. If you set too high, and the latency increases. Its a balancing act.

[giallanon]

Yes, I've resolved.

I was aware that in order to reduce latency you've to use small buffer, BUT the ASIO config windows that comes with ASIO POD driver is not very clear.

As you can see from the picture, there's a nice slider near the bottom of the window (red arrow) that talks about latency and buffer size.

This is the first thing I've noticed; I set it to the left and close the window.

 

What you want to modify instead, is the buffer size near the green arrow. This will reduce the latency, not the slider.

 

 

[chuberto]

I always have my settings to give the highest latency. Yes, the highest. Everyone sits there chasing the smallest possible but if you're using hardware monitoring, Gearbox or going through a POD then you're not getting any benefit for running it like that. I keep it high and can then slap plugins, impulse reverbs and any other CPU hungry effects all over the place without any worries at all.

[duncann]

I always have my settings to give the highest latency. Yes, the highest. Everyone sits there chasing the smallest possible but if you're using hardware monitoring, Gearbox or going through a POD then you're not getting any benefit for running it like that. I keep it high and can then slap plugins, impulse reverbs and any other CPU hungry effects all over the place without any worries at all.

 

Very good point. I usually have my soundcard set to 10ms. Undesirable audio effects start to happen when set to anything lower than 4ms.

[giallanon]

I use the pod at night so that I can play guitar thought the pod, have a metronome coming from the pc ( or a backing track, or a drum machine) and listen to the whole mix with the headphone attached to the pod. In the meanwhile I may want to record myself playing and in this case, a low latency helps a lot because I must play in sync with the backing track/metronome I hear coming from the pc into my headphone

[Rewolf48]

I was just about to say the same!

[mazuwa]

With real time vst plugins you must compromize between minimum sample buffer and security against cracks.

What I do is minimizing the numbers of active DAW tracks and fx during recording for reducing CPU.

I do not use same fixed values, it depends

Finally I recommend to analyse (zoom in) the recorded test wavs to see if there are cracks inside.

For playback and mixing only reasons I increase the sample buffer.

Are you using dedicated music PCs? If not, unplug your LAN cable (don't deactivate LAN adapter for some softwares do need it), deactivate antivirus software and other music PC optimizations.

 

If you want to measure the real latency of all components in sum you can download this RTL Utility and let the signal take your way of choice over your DAW and through the plugins.

http://www.oblique-audio.com/free

Attention: You need a loop e.g. from input 1 to output 1 and DO NOT FORGET to unplug your headphone and monitoring cables before that- or your ears are gonna die!

Inside RTL Utility you can select the drivers, frequency, bit depth, inputs and outputs.

Great to compare between different software, drivers and devices.

With POD HD ASIO driver set direct monitoring level to zero and use DAW monitoring for measuring instead!