![]() On Windows 10, the Duo-Capture driver installs automatically (no separate discs or downloads from Roland required), however I experienced a lot of driver instability on my first round of latency tests with this interface after the drivers installed (some tests straight-up failed while others took significantly longer or shorter to complete than expected).Also, for all control panel settings other than buffer size, I just left them at their factory defaults. I had to choose the nearest approximate buffer size for the comparative tests. The Roland drivers are kind of weird in that they do not use powers-of-two buffer sizes, and their configuration control panel also has a number of non-standard options and metrics.I don't actively use my Duo-Capture these days, but I keep it around for testing and ad-hoc stuff. The Duo-Capture EX is one of my favorite entry-level interfaces, just because it packs a lot of features into a small, reliable and affordable package. Since this is the only interface I currently own that supports two different data buses, I tested it both as a Firewire and as a USB device.In order to perform the latency tests at different buffer sizes I had to use the RME control panel to choose the new buffer size then "reload" the driver in the test tool before performing each round of tests. RME's drivers are a little bit odd in that they don't appear to advertise more than one sample buffer size at a time like most others do.In recent years, RME has released an updated model, called the Fireface UFX+, which includes USB 3 and Thunderbolt support, and the Fireface UFX II, which is USB 2 only.I typically use it as a Firewire device, just to avoid possible USB contention in my studio. I own the first generation UFX, which has both USB 2 and Firewire support.This is my primary interface for writing and recording. The Fireface UFX is a prosumer legend, offering tons of analog and digital I/O with top-tier performance, reliability, and flexibility in a single rack space form factor. These are the ones that I still own and used for these tests. ![]() I have bought and sold many interfaces over the years. With this tool, you patch your audio interface's outputs to its own inputs, forming an audio loop, and measure the time it takes for a full output->input round trip at a given sample rate/buffer size. A driver written for a specific device with efficiency and optimization in mind can significantly outperform a less optimized driver on similar hardware.įor my tests, I used a free tool called RTL Utility, by Oblique Audio. There are many factors that contribute to interface performance, but the most important appears to be driver quality. When shopping for audio interfaces, it's good to know which devices offer you the lowest reliable round-trip latencies at given sample rates and buffer sizes. The trade-off (and the reason we're studying this at all) is that too big of a buffer at a given sample rate can result in such a great delay that it can become difficult or impossible for a musician to keep in time with the rest of the music while attempting to sing or record. The buffer provides protection against glitches (pops and drop-outs in the audio stream), and the harder your computer is working, generally the bigger the buffer you need for glitch-free audio. When you record at high sample rates, your computer processes more audio data, which usually requires larger sample buffers in order to handle audio as reliably as at lower sample rates. The sample rate is the number of samples per second the audio stream is encoded at, and the buffer size is the number of individual samples included in each streaming buffer. RTL is the metric I tested for: What is the total amount of time an interface takes to send and receive audio given certain settings?Ī latency measurement is only meaningful if you know two other values: Sample Rate and Buffer Size. There is some amount of latency in both an interface's input and output audio path, and round-trip latency (RTL) is the combination of both of those times. This buffering introduces some amount of latency that is, the fact that the audio data is buffered means that there is a small amount of built-in delay between when the audio data is first transmitted by one device (an audio interface) and received by another (your computer). ![]() Audio interfaces are audio streaming devices, and on modern operating systems all streaming is "buffered" or "packeted." Rather than truly sending a constant binary stream of audio data, your computer bundles up tiny chunks of audio into separate data buffers that are reassembled at the destination end of the stream.
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