The audio interface is your studio systems link to the outside world, which allows the sound to be both captured and played back through your speakers and headphones. There are a number of factors to consider when choosing a suitable audio interface depending on how much external hardware you wish to connect and how much overall performance you require.
The first consideration is to what connectivity options you have available. For most people a standard USB interface will be more than adequate and easy to get started with, but for more demanding users they may find recording nirvana through other means.
This is easily the most common type of audio interface available today. Great value and simple to use, you will often be able to set it up and get to work quickly with many different types of system, including possibly your mobile phone in some cases!
The USB standard is slower than the other options here although can still easily handle dozens of high quality channels simultaneously. However, due to the typically slow throughput it is also the standard most prone monitoring latency whilst playing through it. If you wish to play, monitor and process an instrument all in real-time, you can often still manage this at the lowest ASIO buffer settings, but often this will be offset by a decreased track count as you put the system under more load.
The newer USB-C standard is quickly appearing on the latest interfaces and whilst this can offer more bandwidth overall as well as better power delivery to help with your connected powered mics, it does not inherently cut down on the recording latency.
That is not to say that USB experience cannot be fine-tuned and some of the more premium firms like RME spend a lot of time developing and optimizing their hardware from the ground up, allowing their USB devices often perform closely to their other models. Although the trade-off is that R&D cost do tend to reflect in the product price and these tend to be more commonly seen in professional broadcast suites, recording studios and touring shows.
TThe classic internal audio sound card solution is still often the highest performing choice. Popularity has dropped over the years as internal cards often lack the plethora of I/O connectivity found on the other options making internal cards less of a one-stop solution for most people.
The other side of the coin however is that these cards often feature ADAT or other high track count digital connections, so they can still allow you to connect-up your own external converters of choice. This means you can mix and match the best tools for the job and whilst this can prove to be a costly option, you will not find a better way to customize the setup to meet your exact requirements.
Also with the cards having a direct connection via the PCIe lanes to the CPU, this means that the latency and overall performance of an internal card is the best we tend to see out of any of these options. Their internal nature also makes them a great fit for any scenario where can be installed and relied upon to keep running 24/7, making them popular for broadcast and commercial production studios.
This buffer control setting is found on all ASIO based interfaces and gives the artist control over the system performance to a degree. ASIO is the driver standard originally developed by Steinberg but adopted by everyone that bypasses the Windows WDM audio engine and gives the interface a direct line to the CPU and makes low latency, high quality recording possible on a PC.
The buffer captures data in real-time and then feeds it to the CPU to process and send back out. The lower the buffer setting, the more often it empties to the CPU and the quicker the response. However if it tries to access the CPU too often and another driver is hogging the processor this can result in drop outs of the audio and is one of the factors that the system tweaks already mentioned will try to resolve. Another side effect of this, is that by lowering the buffer you make the CPU work harder and speed up likelihood of it being overloaded.
The newest standard here and the best of both worlds. Thunderbolt interfaces will talk to the system via a dedicated Thunderbolt connection in much the same fashion as the PCIe card, meaning that a great Thunderbolt implementation can almost be as fast as fitting an internal soundcard.
Of course as these interfaces are also external like the USB options, it makes it possible to build interfaces with both high I/O counts and top of the line performance, with plenty of great interfaces having appeared over recent years.
The downside currently is that the Thunderbolt options tend to be the most costly as the standard is still new enough to have its more premium price tag and all the extra features tend to position these solutions in the more premium end of the market. Over the recent years however, Thunderbolt costs have been slowly coming down as more and more models continue to be released we are now seeing some firms such as Presonus offering superb Thunderbolt interfaces as very reasonable prices.
So, connectivity aside what other features should we be considering?
The easier of these choices and one that can tend to guide the interface selection for many people, especially for those recording bands as well as other live audio sources, is how capable it is in capturing a large number of external sound sources. Whilst smaller home project interfaces will range from relatively simple 2 in / 2 out solutions where you only look to record a single instrument at a time, we see models right up to the broadcast level MADI based units which can handle track counts close to 400 channels of streaming audio in use.
We tend to find that these I/o differences within a given interface range can often be a major point of differentiation between the units. With models such as Focusrites popular Scarlett series being largely the same core hardware within a range, but with more and more I/O options being added as you progress across the models, making the top end units much more suited to larger multi-track recording sessions.
For users recording live acts the signal path of the interface will also be of prime concern, with the selection and quality of the included preamps normally being a well-advertised part of the feature set and one, which many manufacturers place a focus on when designing an interface. On lower end interfaces not every input on an interface will necessary have a preamp, rather choosing to often mix and match with regular line inputs. If recording is the prime requirement such as guitars and traditional instrument recording via microphones, then making the choice to go with an interface with more preamps as standard can often prove beneficial.
As well as the signal chain heading in to the interface, the conversion setup of the device is also something to be considered. While even the cheapest DAC (Digital to Analogue Converter that digitises your audio) found in modern interfaces is more than up to the job for regular recording duties, as you move up through the ranges the sound quality often tends to improve to mastering grade. If you have a well-tuned listening and editing environment with great reference speakers, a high quality DAC is the icing on the cake, providing improved stereo imaging and crisper, tighter reproduction that makes mixing and editing an easier task thanks to having the extra level of detailed feedback.
Other I/O that can be found on an interface include digital connections such as S/PDIF, digital coaxial and ADAT, which can all vastly expand upon the basic input options and offers the capability of directly connecting large track count mixing desks or rack-able external preamps to further expand the control or I/O options.
Performance in regards to audio interfaces is down to how well the driver is able to integrate the device into Windows, and the better performing drivers tend to offer lower latency, which translates in a number of ways.
Firstly, there is the latency you can hear, which is the time taken for the audio to be recorded into or generated by the system and then outputted from the interface. This referred to as the RTL or Round Trip Latency.
As an example for artists recording themselves this can translate into the time taken to get a guitar signal into the system, be processed by the CPU and back out to the monitors so they can hear any audio processing that may have taken place.
Here in Scan Pro Audio we have found that as an rough guideline to real world terms drummers tend to require the best RTL of around 10μs (10 microseconds), guitarists 12μs and vocalists 15μs for it not to be noticeable and lag free in use. The good news is that more or less every interface currently on the market can achieve this with at least one setting (the lowest buffer setting) so in those regards the really is no wrong choice , although there are certainly benefits of going with the top end audio devices as not all interfaces are created equal.
Some audio interfaces may perform with a far higher track count or prove to be more responsive than another running at the same ASIO buffer setting. This is often the case as you move up to the more premium brands who choose to focus on improving driver performance when refining their hardware. In order to reduce this latency whilst recording, we can look to bring it down by reducing our audio interfaces ASIO buffer setting, but the trade-off is a higher processing load placed upon the CPU often resulting in a reduced overall track count.
The buffer captures data in real-time and then feeds it to the CPU to process and send it back out. This means that the lower the buffer setting, the more often it passes data to the CPU and the quicker, more responsive the system will become for recording. However, if it tries to access the CPU too often and another driver is placing high demands on the processor as well this could result in dropouts of the audio as it runs out of processing time.
On the other side of the equation you can raise the buffer, relax the CPU workload and get rid of any audible glitches although the real world audio latency will then start to become noticeable as it can begins to make it harder to monitor and process the audio in real-time during recording. This will also exhibit with other connected controllers and MIDI instruments like keyboards, which will also may to feel more lagged as you play. For anyone working solely in the box this is less of a concern, as the sound does not have to travel into the system, effectively halving the latency from the CPU to your ears as it has a much shorter journey.
This means that you will sometimes have to balance all these factors whilst recording. Many users will adopt the method of reducing the driver buffer to capture the artist performance with processing in real time, then to raise it again during the mixing part of the process in order to get more performance out of the machine and power additional plugins.
Scan Pro Audio conducts extensive testing of driver performance along with critical comparisons of the other features found on within out interface range. If you find yourself looking into a new interface to use alongside with your 3XS system, it is always worth contacting us for a discussion about all the latest options.