How do I adjust latency in Ardour 6.7?

Hello - I recently installed 6.7 on Ubuntu and I’m getting some latency. How can I adjust this? Is there an easy way to do this? Thanks!

I assume you monitor through Ardour

Mic → soundcard in → ardour → soundcard out → speaker/headphone.

In this case the lower the buffer-size in Ardour Menu > Window > Audio/MIDI Setup.

but keep in mind that the smaller the buffers, the higher the chance of dropouts (in case your system cannot keep up). Most systems are fine with 256 sample/cycle / 48kHz. Smaller buffers often requires tweaking the system.

Watch the “DSP load” top-right in Ardour’s status-bar.

PS. Also make sure that you do not have any latent plugins in the session which may delay the signal further.

Sorry, I should have given more info.

I’m using a Presonus interface that I plug into my USB on my computer. My OS is Unbuntu 20.04Lts. I am also using JACK.

Would changing the buffer size in this case work?

Also I tried adjusting the buffer size now and it did not work. Anything else I can try?

What did you do? How did you establish that the change had been made? What did not work?

I lowered the buffer size as instructed above all the way down to 64 samples and there was still a latency. I tried each sample size in between and there was still a latency.

You cannot eliminate latency. For most people, 64 samples is fine, since there’s 3msec anyway between your head and typical near field monitors, and 10 msec between a drummer and bass player.

Most audio interfaces only accept very specific buffer sizes, often powers of 2. You can’t pick arbitrary values.

Exactly what are you doing that you say ‘there is latency’? Are you recording an instrument, playing a virtual instrument, listening to playback and there is phasing? What is it making you say ‘there is latency’? What are you hearing precisely?

As of now most of the answers above assume you are monitoring a live input of audio through your sound card and Ardour, which going down to 64samples for your buffer size puts you well below the 10mS that even most trained professionals can hear at almost every sample rate.


Is it an annoying latency or did you just notice with the help of for example Jack Control that it’s latency in the system?

In case that’s annoying: Is it annoying all the time or is it when you need to monitor when you are recording? You might want to put a mixer before your sound card if the card does not have zero-latency monitoring.

Some singers will have problems even with low latencies under 10 milliseconds when monitoring with headphones because the bones, flesh, and anything else in the head interferes with the sound from the headphones. In that case, a mixer or a sound card with zero-latency monitoring is essential.

Another scenario is latency when you play something with a MIDI unit that triggers sound from a virtual instrument in the computer. It is true that in a band or orchestra, the distance between some players and even for example your own amp might do so you have a latency of 10 milliseconds (that’s around 3.4 meters distance) or more, so latency is natural. But in general, one wants a short latency as possible when playing an instrument so it feels natural. I’m not comfortable with latencies over 10-15 milliseconds so I sometimes use 64 frames/period but mostly 128 when recording something that’s triggered by a virtual instrument. But that is my preference, and I use real-time kernels in order to have these low latencies without xruns.

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