This article explains the implications of buffer sizes and latency, including how different sample rates have an effect on them.
When you are recording audio with your interface, you might notice a slight delay in the audio coming back to you. This is because the computer needs time in order to process the audio, and then route it back out of your interface.
The amount of time that this takes is in part controlled by the buffer size, which you can set yourself. You can adjust the buffer size for your current requirements. Using a lower buffer size (and therefore getting less latency) gives your CPU less time to process the audio and therefore the CPU must work much harder to process the audio in that amount of time.
The real objective is to find a buffer size low enough that your CPU can handle before it has to work too hard. You will notice when it's working too hard because there will be problems with the audio, such as crackling, stretching and even dropouts.
To reiterate, when setting your buffer size, what you are doing in effect is giving the CPU of your computer a designated period of time to process audio, before it is passed to and from the interface and the recording software. A higher buffer size will create a longer latency, a lower buffer size will create a shorter latency.
Latency is measured in milliseconds. Depending on the interface, the buffer size will be set in either samples or milliseconds. A buffer size that is set in samples equates to a certain amount of time, but that amount of time is dependant on the sample rate (amount of samples per second). The higher the sample rate, the less amount of time (milliseconds) that the buffer size equates to. Let's say you have set a buffer size of 512 samples. The amount of time (milliseconds) that 512 samples actually equates to, will depend on how long it takes for 512 samples to occur. At higher sample rates, samples happen more quickly (there are more in one second) and therefore 512 samples is a shorter period of time. At lower sample rates, there are fewer samples in a second and therefore 512 samples is a longer period of time. If you set your buffer size in milliseconds and not samples, then you are controlling the overall latency in time, and the buffer is adjusted accordingly for each sample rate so that you always get the latency that you have set in milliseconds.
On a Mac system, the buffer size is set in your DAW - usually in the Audio section of the Preferences page (DAWs may vary).
On a PC system, the buffer size is set either in MixControl, or the Audio Control Panel.
How much latency is too much?
To get an idea of how long one millisecond is, it's best to give some real-world examples that people are familiar with:
Blinking your eye takes between 300 and 400 milliseconds in total, on average.
The time it takes for the sound of a snare drum to reach a drummer's ears, is about 2.1 milliseconds.*
If two people stand at the opposite ends of a double-decker bus (assuming its maximum length of 15 metres), there is a 43-millisecond delay when they speak (or shout) to each other.*
* These examples assume that the speed of sound in air is 343 m/s.
Why can I still hear latency?
The latency that you set using your buffer size (and sample rate) is not the only latency that eventually impacts the sound by the time it reaches your ears. Software can introduce latency. Some plugins cause latency. Some computers have more, or less, latency than others. When connected via a hub, the latency/performance trade-off is less favourable and you may need a higher buffer size.
Generally, the better a computer is, the less latency that it will incur on its own, but this depends on the condition of the computer, and its load at the time. This is another reason to keep your computer in good order. Have a look at our Windows 7 and Windows XP optimisation guides to help keep your computer running well.
Zero/Ultra-Low Latency Tracking and Direct Monitor
Our interfaces offer Zero or Ultra Low Latency Tracking and Direct Monitor options to further help circumvent the issue of latency as much as possible.
Zero/Ultra-Low Latency Tracking is a Routing Preset in Saffire MixControl and Scarlett MixControl. What this does is assign 'Mix 1' to each of the unit's analogue outputs. By default, 'Mix 1' is a combination of all analogue inputs and DAW 1/2, mixed together. The analogue inputs represent a direct feed from the inputs, and so by assigning Mix 1 to the outputs, this essentially routes the inputs straight to the outputs, meaning that you can monitor your recording source (whatever you have plugged into the inputs) without that audio being passed to the computer first. This means that you can hear yourself without the latency incurred from the computer having to process the audio. Mix 1 also contains DAW 1/2, so that you can hear your backing track.
Direct Monitor, which is available on our interfaces that do not have MixControl software, offers the same essential feature as the above. When using Direct Monitor you hear the audio from your recording source before it is passed to the computer, again negating input latency in the same way as above - you will however still hear audio passed from the computer regardless.
When using these features, it is highly recommended that you mute the channel you are recording into in your recording software. If you do not do this, then you may hear a 'doubling effect' or echo. This is because you hear the direct sound (before it is passed to the computer) and then you hear the same audio that has been processed by the computer afterwards. Muting the track you are recording into prevents this, as you will only hear the recording source directly.
Further information on setting up MixControl can be found in our MixControl tutorials.
As always if you have any questions or concerns, you can get in touch with us through our Support Contact Form and we will be happy to help.