This is Part 1 of a 3 part article on latency.
Intro – PART 1
What is latency and how can I minimize it?
Latency is the time delay for an analog signal (such as a guitar or microphone) to be converted to digital signal, processed by a computer, then converted back to analog to be delivered to speakers.
This delay can get annoying, especially in live scenarios, or when recording in the studio when you need to hear yourself play check out for More at orangecoupons.org.
What can you do to reduce this time? Ultimately, latency is a balance and a compromise of quality and speed, measured in milliseconds. There are a ton of variables in the production process that can cause or affect latency, and we’ll explore all of them in this series.
Why is something on latency this lengthy? Isn’t latency pretty simple? Change your buffer? Yes it’s lengthy, but you’re getting years of experience with audio, condensed into a few short articles.
There’s a certain amount of tech savviness that is needed in music production to fully utilize your tools because most everything is ran on a computer. A good analogy is that most people aren’t knowledgeable about cars. They just know how to drive from point A to point B. Likewise, most people know just enough about computers to get their work done. However because of their line of work, a racecar driver knows significantly more about how a car works than the average person. They’ve learned how their tools work to get that competitive edge and better use them to their advantage.
1. How is latency created?
To get better results with latency, we need to understand latency is a balance of quality and speed. Again, there are a ton of variables in the production process that can cause latency. So first let’s explain the different types of latency, so we know how to tackle the problem.
Input Latency: When recording an instrument such as guitar, your computer has to convert this audio from analog to digital.
Processing Latency: This audio is processed with fx on it (processing latency), which is a HUGE variable depending on the amount of fx layered (since each one can delay your signal chain), especially if it’s on your master.
Output Latency: The time it takes for your computer to convert digital signal back to analog out to your speakers.
These 3 processes combined is called round trip audio latency. This can often time take the most time because of the signal path. We’re going in and back out. And no, longer wires don’t cause latency, don’t even think about it… We can see the delay time of the input and output latency listed in milliseconds in our Audio Preferences in Ableton.
If you’re playing a midi controller, there is no latency to transfer the data of notes played to your computer, because it’s already digital. Midi controllers have no input latency.
Latency can occur in the process of generating sound of the midi instrument such as a bass wobble in the soft synth Serum. There’s an additional amount of processing latency in the fx after the bass such as distortion or reverb.
And finally, the output of the midi instrument will have a certain amount of latency. The time it takes for your computer to convert digital signal back to analog out to your speakers. But often midi instruments have less latency than audio because input audio latency is not part of the equation when using midi instruments.
So with all that said, HOW DO WE REDUCE LATENCY?
As I said before, latency is a balance of quality and speed. We could just say, let’s not use FX… and then we wouldn’t have any latency right? Not quite because you’ll still have the input and output latency to deal with. So what can we do?
If you’re just starting out, Get an audio interface.
This piece of gear is much better designed to get high performance audio in and out of your computer. A small affordable USB audio interface costs around $100.
How much do you think the digital/analog converter built into your laptop costs to make? Most likely, just a few dollars. You get what you pay for.
But what kind of interface should you get? USB, firewire, thunderbolt? What’s your budget? USB interfaces are the most affordable, but the speed of USB is limiting. You can still get decent low latency results with USB interfaces. This interface is two channels in and two channels out.
But when you want to do multitrack recording, such as tracking drums or a full band, you need a heavy duty interface such as a thunderbolt interface. This is because the more channels that are recorded at the same time, the more data is being sent to your computer. Because of this, we don’t see any 16 channel standard USB interfaces out there (since USB is slower) but 16 channels is totally doable with a thunderbolt interface.
2. When to pick the right settings, Part 1
2a. Monitoring From Your Interface.
The simplest thing to do to get zero latency (perfect for recording) is to switch on monitoring mode on your interface. When in monitoring mode, audio is still sent to your DAW to record, but you can mute the output of your DAW and only hear the direct output of your interface. The audio that you hear bypasses the computer.
(PC Only) If you’re an on the go producer, have a Windows laptop, and don’t want to lug around an interface, There is a sound driver called asio4all which can give you better latency with the built in sound card on your laptop.
In Live’s preferences, pick Asio as the driver type, then the Audio Device as ASIO4ALL v2
But you’ll get better results with an audio interface since it’s designed to do what we’re wanting. (ie a screw driver, vs a cordless drill.)
2b Set Your Buffer.
When producing a song and you have a ton of midi and audio tracks with a ton of fx or plugins, you computer may struggle to keep up and so you may hear dropouts or pops and clicks. You can increase your buffer to a higher setting. This will work your computer less, but will increase your latency. To do this, go to Ableton Live’s Preferences, Click on the Audio Tab.
If on a PC using ASIO4ALL or the Asio Driver for your interface, to change your buffer size, click Hardware Setup
Once the ASIO4ALL panel is open, move the buffer slider to the right to increase buffer size and to the left to decrease.
On a Mac, to change your buffer, simply go to your audio preferences and change your buffer size from the pull down menu of buffer size options.
You can reduce latency by decreasing your buffer size, but that will work your computer harder. To reduce the load on your computer, you can freeze or flatten audio tracks
that take a significant amount of CPU percentage. Remember, Muting a track will not disable it, but you can select all devices on a track and group them in an audio effect track, then disable that rack.
In a live performance scenario, it’s a balance of midi instruments and backing tracks that you flattened. The more tracks you freeze that are just audio tracks, the less you workload your computer has, thus the lower you can set your buffer. However you still want some midi instruments that have quality sound to play live. It’s a balancing act.