Digital Sound & Music: Concepts, Applications, & Science, Chapter 5, last updated 6/25/2013
Figure 5.24 Mix knob on audio interface
In general, the way to reduce latency caused by buffer size is to use the most efficient
driver available for your system. In Windows systems, ASIO drivers are a good choice. ASIO
drivers cut down on latency by allowing your audio application program to speak directly to the
sound card, without having to go through the operating system. Once you have the best driver in
place, you can check the interface to see if the driver gives you any control over the buffer size.
If you're allowed to adjust the size, you can find the optimum size mostly by trial and error. If
the buffer is too large, the latency will be bothersome. If it's too small, you'll hear breaks in the
audio because the CPU may not be able to return quickly enough to empty the buffer, and thus
audio samples are dropped out.
With dedicated hardware systems (digital audio equipment as opposed to a DAW based
on your desktop or laptop computer) you don’t usually have the ability to change the buffer size
because those buffers have been fixed at the factory to match perfectly the performance of the
specific components inside that device. In this situation, you can reduce the latency of the
hardware by increasing your internal sampling rate. This may seem counterintuitive at first
because a higher sampling rate means that you’re processing more data per second. This is true,
but remember that the buffer sizes have been specifically set to match the performance
capabilities of that hardware, so if the hardware gives you an option to run at a higher sampling
rate, you can be confident that the system is capable of handling that speed without errors or
dropouts. For a buffer of 1024 samples, a sampling rate of 192 kHz has a latency of about 5.3
ms, as shown below.
If you can increase your sampling rate, you won’t necessarily get a better sound from your
system, but the sound is delivered with less latency.
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