☰ Menu+  instagram  ringing ear 

Welcome to my home page. I live in Brooklyn, NY and Oak Bluffs, MA; I'm dad to Hazel. I work as a programmer; I have a dog, and she has a web site too.

Thatcher's rants and musings

rants index: current | 2016 archive | 2015 archive | 2014 archive | 2013 archive | 2011 archive | 2009 archive | 2008 archive | 2007 archive | 2006 archive | 2005 archive | 2004 archive | 2003 archive | 2002 archive

14 Sep 2016

Tascam US-16x08

I got a Tascam US-16x08 USB audio interface in anticipation of recording a band. Up til now in Tuffy I've just been using a 2-input Focusrite Scarlett 2i2 which has been fine since we use programmed drums and overdub everything one track at a time. But for drum kit and live band recording I need more mic preamps and A/D, and the Tascam US-16x08 seems like a good value.

A couple of things weren't clear in the manual and weren't Googleable so I'll put them here.

Latency -- for recording into the computer and monitoring out of the computer, the default settings seem fine. However, going round-trip through the computer and DAW adds latency. How much? It depends on the DAW buffer size, but there is some built-in overhead to the converters and USB even aside from the buffering setting. I did some recording tests, running a mic signal into the 16x08, into the computer, back to one of the line outs in the 16x08, back into an analog in, etc, and multi-track recording all the inputs simultaneously. By looking at the time gaps in a sharp sound like drum-stick clicks, I could measure the round-trip latency from the waveforms. My measurements basically showed that the latency value reported in Ableton Live's audio preferences dialog is exactly right. For the 16x08 at 24 bit / 44100KHz through Ableton Live 9.6 on a MacBook Pro running OSX 10.11.6:

buffer (samples)latency (milliseconds)

For minimum latency, 32 samples at 96KHz reports 5.27ms. That's not enough of a difference for me to change sample rates. 6 or 7 millis is low enough latency for doing live effects like practicing guitar through an amp sim (I tested latency through Amplitube's fake Marshall and it was fine). But for recording, I don't trust my computer enough to use small buffer sizes, so I crank the buffer up to 2048 and monitor directly from the interface.

Which brings us to live monitoring while recording. The 16x08 has an internal digital mixer that is controlled from a custom app on the host computer. When the mixer app is not open, the 16x08 does no live monitoring by default. The analog inputs and outputs just get mapped to independent digital channels via USB, and all routing etc would have to be through the DAW. You get full flexibility, but you also get latency.

Alternatively, you can open the Tascam mixer app, which has Phase switch, EQ, Compression, Solo, Mute, Pan and Fader for each analog input channel, plus a master Mute and Fader.

The manual and the internet are not clear at all on how the mixer effects are routed in relation to recording, so I played around to find out. Basically, the analog signal comes in, is trimmed and pre-amped, with gain controlled by the analog gain pots on the front panel (for channels 1-10), and then is converted to digital. It appears that clipping occurs in analog just before, or as part of, A/D conversion. In any case, it sounds like clipping. Then, the digital signal is affected by the Phase, EQ and Compression controls in the mixer, and then passed to the USB input channels in the computer. Then, the mute, solo, pan and fader controls in the mixer affect the contribution of the channel to the 16x08's internal Stereo bus. By default, this Stereo bus is sent to the headphone jack and Line Out 1 & 2, which you can use for live monitoring.

I measured the latency for live monitoring through this mixer and it was around 1ms from input to headphone and line out, perfectly usable.

I had thought about using the compressor in limiter mode to protect myself from clipping, but I gave it a try and it seems useless. The problem is that the digital compressor sees the signal after it has already been clipped, which is too late. Seems reasonable in hindsight; anything else would require analog limiters or compromised A/D resolution.

The mixer also seems dangerous for sweetening the live signals for headphone monitoring, because the effects will also end up in the recording, and may not be what you want. Seems like it would be more useful if the effects were applied only to the internal bus (the way fader and pan are), and not sent to the USB input channels.

Another observation: the 16x08's mixer settings are lost the moment you unplug the 16x08 from the USB host, so you have to have the computer plugged in and the mixer app open in order to do live monitoring.

Is the mixer good for anything? Yes, it's still useful to configure low-latency live monitoring levels, though there is only one stereo pair output, not enough for a different mix for several players.

It would be nice if there were controls for sending different amounts of each input channel to any of the 8 line outputs, but that doesn't seem to be possible.

So I will keep all this in mind for tracking.

Re noise floor -- with nothing plugged into input 1 and the input gain pot below 50%, the input noise through most of the frequency range is around -119dB (around 20 bits equivalent). Turning the gain from 50% up to 100% gradually raises the noise to -95dB (around 16 bits equivalent). Turning on the phantom power makes a big click, but then things settle back down and the noise is about the same. No complaints here.

rants index: current | 2016 archive | 2015 archive | 2014 archive | 2013 archive | 2011 archive | 2009 archive | 2008 archive | 2007 archive | 2006 archive | 2005 archive | 2004 archive | 2003 archive | 2002 archive

tu@tulrich.com |