What is latency? (and why you don’t care)

Latency is a word you hear thrown around the digital audio world quite a bit. A lot of people use the word latency to mean a lot of different things, and often are not clear on what it does actually mean or how it will affect their home studio workflow. You will often hear salesmen or so called experts recommending this gear over that gear because you will get better latency. The truth of the matter is latency does not matter at all for the recording needs of nearly every home recording enthusiast. Most of you literally should not care about latency at all.

What is latency?

Simply, latency is the time it takes digital audio data to go through your soundcard’s software driver. Great, so what is latency? Latency is the amount of time between when your computer tells a sound to play and the time when your speakers make the sound, or the time between when you put a sound to be recorded into your computer and the time it actually gets the data to record it. Awesome, so what is latency? I thought you’d never ask…

There are two types of latency. More accurately there are two directions of latency. There is playback/output latency and recording/input latency. Often they are the same amount but sometimes they can differ. We are going to tackle them one at a time. Let us start by taking a look at what happens when you press play in your favorite home recording software.

Playback/output latency
daw playback latency

  • A. Press “play”
    Represents the moment in time when you press play in your DAW.
  • B. Read data
    All the tracks in your recording are stored as files on your computer’s hard drive. Before playing them back, your computer needs to read a little bit of every file. Depending on your hard drive speed, DAW software, RAM, computer speed, track count, etc. this step can take around 5-100ms. This is not latency.
  • C. Process data
    All those tracks, plugin effects, mix automations, etc. require a bunch of math to do what they do. Your computer needs to crunch all the numbers to work out what the final sound should be. This can take anywhere from 1-100ms or more depending on some of the same factors from step B. This is not latency.
  • D. Send to driver
    The computer crunched all the numbers but all it did was come up with more numbers. We can’t hear numbers. This is where the soundcard and its driver come in to play. Your home studio computer needs to push all these numbers through the soundcard so they can be converted into a signal which will later be turned into physical sound. This is done by filling up presized buckets (buffers) of data and sending them through the soundcard to be converted from digital to analog. The time it takes to push data through these buffers, convert them from digital to analog, and send them to the soundcard outputs is latency.
  • C. Sound is heard
    This part is pretty straight forward. Your speakers take the analog signal and move air so you can hear your music. This is not latency.

You can see there are a lot of things happening in the computer between the time you press play and when you hear the sound. Most of these things are not latency. Only the processing time associated with moving data through the soundcard driver is considered latency. This is true for recording too, but the steps are just a bit different.

Recording/input latency
daw recording latency

  • A. Sound is captured
    This sound can be captured through a microphone, an amp modeler, a set of V-Drums, or any other creative way you can think of to get sound into your home studio soundcard. This is not latency.
  • B. Convert sound to data
    The analog sound gets converted to digital data by your soundcard. The driver fills input buffers with chunks of data to be retrieved by your computer. The time it takes to fill these input buffers to be retrieved by the computer is latency.
  • C. Computation
    The computer grabs the data in the buffers, does some mundane computation involving streaming and creating a file, and sends this data toward the hard drive for storage. This is not latency.
  • D. Store Data
    The data is stored on your computer hard drive. Depending on the speed of your hard drive and the number of tracks you are recording, this could take from 5-50ms or more. This is not latency.

Again we can see that many things happen when recording a file that take some time and that almost none of them are latency. The only thing that is latency is the time it takes the data to get through the soundcard driver.

What effect does latency have on recording?

  • “You can’t use a soundcard for recording unless the latency is below 10ms” – False!
  • “With my brand X soundcard I can record with zero latency” – Impossible!
  • “My home studio computer is better than yours because my latency is lower” – Funny, but wrong!
  • “Professional quality soundcards all have lower latency than amature cards” – That’s fiction!

Let’s take a look at the timeline of events when we want to overdub a track in a home studio environment with 31ms of latency.

daw latency in the home studio

  • A. Your computer gathers up the sound to start playing a at 0:00.000. The sound data is sent to the soundcard buffers at this time.
  • B. By the time the data works its way through the soundcard and you hear it, it has been 31 milliseconds since the sound was played. At this point in time you are hearing the sound that was played at 0:00.00 but the computer is playing the sound from 0:00.031. That’s right, you are hearing sound that the computer played 31ms ago. When you play along with the track, what you are playing (the instrument in your hands) is already 31ms behind the music the computer is processing.
  • C. By the time your instrument’s signal has made its way back through the soundcard driver, another 31 milliseconds has passed. Yes, that means the sound you played is not heard by your computer until 31ms after you played it. Worse, the sound is not recorded until 62ms after the sound it was supposed to be played over.

This seems pretty bad. How could any of our home studio recordings sound any good if everything is off by 62 milliseconds every time we overdub? At 120 beats per minute that is about equivalent to a 32nd note. Not even the Rolling Stones are that loose. We still don’t care. As home studio owners latency does not matter to us. Why?

Why don’t we care about latency?

Computers can be the bane of our existence. They almost never seem to do what we want them to do, but always seem to have an agenda of their own. Even in light of the love/hate relationship most of us have with our computers, there is one thing we must never forget. Computers are good great at math! We can’t get around latency. It is impossible. Soundcard drivers have to do their thing. There will always be latency. Computer programmers know this and they build this knowledge into their software. It is a small matter for your home studio DAW to do a bit of math and correct all the wrongs latency introduces. Here is what your recorded track looks like in its raw form according to the real world timing of the events:

before home studio daw correction

Notice the overdubbed sound file doesn’t start until 62ms after the track we were playing over. The computer can easily calculate how long it will take the soundcard drivers to send sound out and bring sound back in. Once it does these calculations it knows the audio you played was meant to go over the audio that happened 62ms before you played it. Yes, the computer is like a time machine! It sends your recorded track back in time, using latency correction, to make sure it plays back right in sync with the prerecorded audio tracks. It ends up looking like this:

after home studio daw correction

Notice the tracks are now perfectly aligned. The principles of latency correction are well defined and understood. All pro level software for the home studio does this automatically for you. I can’t think of a single DAW package I’ve used that does not do this for you. Sometimes there are settings to turn this off or adjust it by manual amounts. You don’t really want to play with those settings under normal circumstances. You can see from this illustration that latency is not bad at all. In fact, you should never need to care about latency. The computer takes care of everything behind the scenes for you.

It should be noted that these principles hold no matter what your latency is. The DAW can correct for your latency just as well at 7ms as it can at 700ms. You should never notice the difference in your recorded tracks. The only thing you might notice at extremely high latency settings is a general feeling of sluggishness in the controls of your DAW. This should not concern you. It doesn’t matter if your song doesn’t start playing until a quarter second after you hit play. It doesn’t feel snappy but it will sound just as good.

Is high latency bad?

If 7ms and 700ms are no different in terms of sound, is one better than the other? That is a complex question but the general answer is no. For the majority of home recording enthusiasts there is no practical difference between higher or lower latency. Here is a comparison.

Low latency:

  • Pros: DAW is generally more responsive. Softsynths perform better (more on that in a bit). Bragging rights (useless).
  • Cons: Harder on your processor. Can’t run as many plugin effects live. Lower potential track count.

High latency:

  • Pros: Easier on your processor. More plugins can be used. Higher potential track count. Softsynths perform better (more on that in a bit).
  • Cons: DAW can feel a touch sluggish. “Experts” will make fun of you (clueless).

The higher your latency, the more time your computer has to batch up data in large chunks and process it. This is primarily important when reading data from the hard drive because physical drives are notoriously slow (in comparison to computations done in memory). If you ever notice your song’s playback starting to experience dropouts or stuttering, this is a sure sign you are running out of horsepower. Increasing your latency can help you out. Higher latency translates to more live effects at mixdown and a higher potential track count.

Who does care about latency?

Well, someone must care about low latency or everyone wouldn’t be making such a big deal about it. There are two situations where low latency matters. Latency is a big issue if you plan on using software monitoring or virtual instruments.

I truly believe software monitoring is one of the worst things to happen in the home studio space during the computer DAW revolution. Other than being a tool to keep you from playing with feel and enjoying your home studio experience, what is software monitoring? Software monitoring is when you use your DAW’s internal mixer to create your monitor mix. When do people feel the need to do this? If you were recording with something like a direct guitar box, then you have no way to hear the guitar signal because the outputs are plugged into the computer for recording. DAW makers started allowing you to use the software mixer to echo this signal back out for monitoring purposes. There is a big problem with this approach. All signals have to make a round trip through the soundcard drivers before you can hear them. Consider our previous examples of recording with 31ms latency. If you wanted to hear that guitar track through software monitoring, it first has to go through the 31ms latency on the way in to the computer, then go through another 31ms latency on the way back out of the computer for you to hear it. You are trying to play in time with the song but all your notes coming back out the speakers are horribly lagging behind your fingers. If you want to experience this first hand, set up a delay effect on your sound with 60ms or so of delay time and 100% wet. Try to play anything with a constant time. It will mess with your head. Avoid direct software monitoring like the plague unless you enjoy spending a lot of money on the latest computer hardware and daydream about configuring every aspect of your computer for barebones, stripped down performance. For as little as $30 you can have a perfect, latency free, hardware monitoring solution (the techniques are outlined in this article).

Virtual instruments are synthesizers that run on your computer with no external host software. You remember that Casio keyboard your sister had in the 9th grade? Virtual synthesizers are a lot like that, except without the $20 of plastic and they sound a ton better. Synthesizer is actually too limited a term for the virtual instrument. Some virtual instruments are drum machines, guitar amp modelers, analog synth modelers, and even artificial band members.Virtual instruments are actually quite useful in the modern home recording studio. The only problem arises when you want to play them live with an instrument in your hands. This can take the form of a midi keyboard playing through a virtual synthesizer or trying to play your guitar through a virtual amplifier. The same situation applies that we saw previously while discussing software monitoring. First your computer has to get the signal of your playing in to the computer, then it has to be played back to you. This essentially doubles your latency and makes everything you play come back to you with a lag.

You can set your rig up to successfully run live virtual instruments but it won’t come cheap. If your sense of rhythm and feel is highly developed, it is unlikely you will be able to tolerate anything higher than 3-4ms of latency.

I hope this explanation of latency has been both informative and enlightening.

This article has been fueled by my love of gourmet ramen. Shout out to WorldRamen.net!

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30 Responses to “What is latency? (and why you don’t care)”

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  1. Matia says:

    Greetins from Serbia,i am thriled with this article,wish there was more like this,im sick of those who write just anything on net,know it all guys and when you need advice badly its good to know there is stil people who want to share knowledge and write stuff to help us mortals,ciao!

  2. bvesco says:

    Thanks for the nice comment! I am doing everything I can to keep this site running and build it up with more articles.

  3. binkdog says:

    ben – question on this. what do you suggest for those of us who are using software plugins for our amp/cab/effects. I record straight into my mac dry through a firewire m-audio device and the only way to hear the sound I’ve selected through amplitube (2) is to have monitoring turned on. I don’t have monitors – just headphones and no mixer – straight in with the firewire box, using garageband with the amplitube 2 setup. Thankfully I have a fast mac and the firebox, so I can monitor live. My experience on this is that it’s pretty much an on/off option – you either get to hear the sound as you play it (from the software), or just hear what you’re playing dry and then apply the effects. The problem with the later is that it’s so hard to really “feel” the music when you’re supposed to have a rockin distorted tone and all you hear is the twanging on dry strings :-)

    thanks – and great site !

    Bink

  4. bvesco says:

    IF your computer hardware is fast and well maintined, and IF you enjoy keeping it in that state, and IF you are getting great results and are totally happy with your current setup, I don’t suggest anything. You’re doing great, stick with what works. If there is a latency problem you want to deal with then I do have a suggestion. I recommend not using software monitoring and using a mixer like this:
    http://benvesco.com/tonemonster/tracking/2008/small-mixer-connectivity/
    As for monitoring without the lame, dry sound of a guitar, you can use an A/B or direct box to split the guitar. Send a dry signal to the computer and a signal to a regular amp or other device for monitoring. Another idea is to use something like POD X3 (a bit spendy) or TonePort GX (under $100 typically) which has the capability to split the signal in hardware or software to send a dry signal to the computer while allowing you to headphone monitor the “amped” signal for inspiration while tracking.

  5. Darryn says:

    Hi. Thanks for the great article. This has really helped me understand latency better.
    I recently picked up a Behringer FCA202 and have started recording some riffs with Kristal and Amplitube. I set the latency really high because my computer is old, but I’m getting a decent tone with the Amplitube VST at around 48kHz.
    I was wondering if you (or anyone reading) could answer me two quick questions?

    1) I started recording at 96kHz and found my tracks were slowing down! To hazard a guess, would this be more a PC/Processor issue or the FCA202 Audio Interface?

    2) Because I cannot hear the signal “on the fly” (I literally play to a recorded click and the twang of my strings hehe), can I use some sort of “line splitter” before the FCA202. Thus sending one signal to the FCA202 and the other to my practice amp at home? If so, do you know of a decent piece of hardware to handle this?

    Thanks again for all the information.

  6. bvesco says:

    @Darryn

    1) Could be a combination, hard to say.

    2) Try a A/B/Y box such as those made by Morley. Use it in the Y configuration sending one signal to your amp and the other to your DAW.

  7. Rene Dwight says:

    Latency is an issue that I think many musicians are not aware of when they first get into using sequencers such as cubase, I think your articles are bound to help a few unfortunates who come up against the problem. Wish I had found them when I first started out.

    Well done for publishing them.

  8. adanata says:

    i cannot find the words to thank you….as a person who really appreciates people who know what they are talking about and feels sorry for people that “think” they know what they are talking about, i have to say that your article has to be sent in every recording forum in the Universe

    You CANNOT imagine(actually i am sure you can) what things are written and said about latency, either cos of ignorance or merely promotional reasons
    thanks again

    • bvesco says:

      > You CANNOT imagine(actually i am sure you can)

      You’re right, I can imagine! Please do spread the word about my site to anyone who can benefit from it.

  9. Nice article.. In my case, I am trying to trigger soft synths in Logic using a drumKat.. Doesn’t work.. I have tried going thru my digi 002 and using the computer output.. Too much latency for me.. I’m using a mac pro with 8 3gh processors so I’ve got plenty of speed and power.. Is my only choice to use an external midi module to monitor with and then transfer the midi info to the Logic plugin?.. thanks for your help.. jamal

    • bvesco says:

      Yeah, you will either have to buy a new soundcard that has lower latency or do what you are suggesting. The approach of recording the midi but monitoring via hardware is exactly the approach I take when recording midi. In my case I have a Windows machine with a SoundBlaster Live which has an onboard hardware synth. I like to make a soundfont sample set of whatever I want to play and use that for monitoring.

  10. Rob says:

    Man thanks for this article this has cleared up a big issue that I was dreading having to deal with, I used to use cubase sx and have software monitoring on and hated the lag that I got when recording guitar from a solid state amp output. I’m now building a new home studio and I was really worried about this when it comes to recording but seeing as I’m micing up a cab now anyway it eliminates the need for any monitoring!

    thanks man

  11. Cole says:

    Hey I’m not too sure if you are still on this site as this post is quite old but I came across it.

    I read your article and it shed a lot of light on latency, thanks. One question I have – My latency correction seems to be leaving about 5ms latency (even with my device set at lowest latency). How can I fix this? Could it be due to my computer speed?

    Thanks

  12. bvesco says:

    It depends on how you are using the term latency and where you’re experiencing it. In general you cannot completely eliminate driver latency from a system. It is not possible. 5ms is very respectable for driver latency.

  13. Jason Rivera says:

    being a computer programmer myself makes me very proud of my job.~;

  14. Yoel says:

    Hey, I’m using Live 8.0.1 and I’ve been told to keep my Overal Latency under 12 ms. I’m using the audio driver of my soundcard (Alesis IO2), simply because I always endure cracks, pops and even interruptions in my recordings when using Asio4All…

    However, in order not to make my guitarsound go buzzing and become distorted in Live I have to adjust the buffer size to about 400(..). It says my overall latency is now 55,8 ms. The Input latency is around 5ms and the output latency is 50ms (!).

    Should I worry about the ‘output’ latency? And how could I adjust this amount?

    A bit of long story, but I hope you understand my problem (at least, I guess it’s a problem).

    Greetings from Holland,
    Yoel

  15. bvesco says:

    Who told you to keep your latency that low and why? If you’re not using soft-synths then your latency does NOT matter at all. 55ms is fine for audio recording and mixing without soft-synths.