Best way to deal with latency

Let me put it this way…let’s say I bought an RME Babyface Pro FS. I would not be using the pristine mic pres, I would not be using the internal effects, I would not be using 22 of the 24 channels and on and on. I would only be using the DAC feature of this interface. Can anyone explain how the DAC of the Babyface is different functionally from the Apogee Groove?

I cannot tell you what is different in the conception, but with an RME you probably won’t need to reduce the buffer to 64 to feel comfortable with the latency (e.g. beside the DAC they build their own USB hardware and driver). I use an RME UCX with a buffer of 256 and even with percussive sounds I don’t feel better when reducing to 128. I don’t feel nor ear the difference. Difficult to say but it could be that your apogee “looses” time somewhere which makes it important for you to reduce your buffer at 64 to feel right with the latency. But, at 64, your have more chances to get crackles.

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@LilyM Several users have reported that asio4all seems to work better than the apogee asio driver with some software than others. It might be worth trying, just in case.


Ah yes…marketing … I suppose I could claim that you can use a bicycle to get from New York to California … but wouldn’t you rather take an airplane? :slight_smile:

Here’s the thing. If you want to play some existing music (an MP3 file, say), then if there is a 16ms delay between the time that you press the PLAY button and you start to HEAR the music, you will not notice that delay.
Similarly, if you are working with a DAW that is playing back tracks, remember that other than the track that you’re actually recording, the information needed to play those tracks already exists and so the audio can be created in advance. In other words, a DAW can look forward to see what will need to be played and create the audio in advance so as to have it ready.

But Gig Performer is not a DAW. There are no tracks — it is intended for real-time use. That means that there is no way to look forward since there’s no way to know what to process until you actually press a key on your keyboard or pluck a note on your guitar. In that situation, anything more than a few milliseconds of delay (latency) will be noticed.

Now, there are several contributing factors to latency.

  1. The sample rate and buffer size define the inherent latency in the computer itself. For example, if you are using a sample rate of 44,100 Hz and a buffer size of 64 then the inherent latency is 128/44100 which is about 1.4 milliseconds.
    What that means is that the entire collection of running plugins must generate 64 samples of audio within that 1.4 ms deadline. If your computer isn’t fast enough to do that, then you’ll hear clicks and pops and you have to either increase your buffer size or decrease your sample rate, the former being normally preferred to retain decent sound quality.
  2. The latency of the audio interface gets added to that inherent latency and that gives you the total latency (NB double the whole thing if you have audio input from a guitar or vocals that has to be processed). If the audio interface (and/or its drivers) adds significant latency here, that will add to the delay. The less latency introduced by the audio interface, the higher buffer size you can use in your computer (for the same delay), giving the plugins more time to meet that deadline.

Check out this blog article we wrote for more details on this topic.


If the awfull ASIO4All works better than the Apogee driver, I am worrying about Apogee users


I really appreciate the detailed responses. I do have a basic understanding of most of this, but I understand now that while indeed the DAC aspect of any audio interface is the same, they are all just playing back audio and generally at a “higher” fidelity… there is playing back audio and then there is playing back audio.

If I am correct in understanding , if you were playing back audio from a DAW that had MIDI tracks and plugins in it, the DAW could preload or look forward to the notes and sounds from the plugins and be better “prepared” (is this really true in the case of plugins? Makes sense with audio). Therefore if an audio device were marketed for basically this use it would not need to be particularly low latency (what I assume would be a well written native driver).

Therefore the Apogee Groove probably does not have a well written driver and is adding its own inherent latency to the equation forcing me to go to 64 samples to compensate for that.

I did find the “safe” 16ms setting you referred to. This is what Apogee calls the USB device streaming settings. I was not aware of that and unfortunately the laptop is in transit for show tonight. I assume that is different form the ASIO settings and if indeed the default is 16ms, Apogee shows these settings going all the way down to 1ms, then this could be of some real help to me. Thank you for that.

I can not find that with Google search, Apogee and ASIO4ALL

No idea about that. I only reacted to what @Hermon said, as ASIO4ALL driver introduce a lot of latency.

Yes - all AD/DA converters perform the same operations albeit at different speeds/quality. Some internal processing may also add latency so if your device has some post processing built into it - that will add some latency too. For example the Apollo Twin Duo is a great sounding interface, but because its running internal plugins it adds latency itself by running those plugins.
On the other hand - A PreSonus Quantum2 (nod discontinued sadly) has a really, really lowe latency with phenomenal quality.

A bunch for sale on Reverb…

I wonder why it was discontinued - they’re still selling the Quantum…what’s the difference?

As far as I know, the only difference is the number of channels. Everything else is the same. As to why they dropped it - it may have been too expensive to make or maybe there was no demand for it.

Ok Understood. I guess there was just a misunderstanding for me because it seemed people had a negative view of Groove because it was for playing back audio. Yet that is what all DACs do, in all interfaces.

It is indeed possible Groove does that at lesser speed/quality than needed, I’m just not sure what an opinion on that would be based on. I understand the view that it’s possible I am experiencing sluggishness due to latency in Groove and that is causing me to run at 64 samples. Yet at the same time, I am running at 64 samples! With some heavily loaded racks, NI Kontakt, BX3, Diva, Amplitube 5, and just some very occasional pops, which would lead me to believe Groove is actually doing a pretty good job.

I should also say that I also experienced this same sluggishness with other audio interfaces, though nothing as high end as RME. But I had an Echo Audio, which was fairly high end for the time and also have used a Focusrite Scarlet… both of which could not come close to handling 64 samples and I had to settle for 128, and even that was unable to handle my trial of BX3 at the time.

I searched my browser history for the reference to asio4all. I found the first three links about it but I can’t find any trace of the last one that pointed out that asio4all had a lower latency.

Absolutely - I personally do not have one nor have I tried it. It may very well be on-par or better than some other intefaces. We’re just making sure that we all understand that there are differences.

“Diva” in particular is a very “CPU hungry” plugin as far as I remember so running it at 64 samples may be what’s creating the occasional pop.

Diva has built-in multi-core support which may help but it certainly is very CPU hungry