0:00 What can it do? 2:38 Control Centre Software 6:00 MIDI Control 6:43 Sound Test vs UAD 7:41 Round Trip Latency 8:45 Zero Latency 10:12 Some Nice Extras 11:33 AudioFuse or Neve and UAD?
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I just realized using my MPC with this badboy I can map any and every liitle thing to it still 
That would actually work perfectly with the mixer layout of the Arturia AudioFuse 16 Rig
I wonder if that would be a nice combo 
…
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Exactly the setup I’m thinking about. I have been contemplating upgrading from my Presonus StudioLive 12, and now its Control Room outputs have died it’s def going to happen. I had been contemplating going down the Audient Evo16/SP8 route, but the DC coupled outputs, USB hub, and MIDI ports make the 16Rig look like a winner.
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Anyone tried getting 2 of these? I have two studio areas with synths and was considering daisy chaining one to the other through adat being the second one just slave. Or I can just add two 18i20s as slaves but will take 2U.
I’m not sure anyone on here has got 1, let alone 2, or at least they haven’t been excited enough to post about it.
It could be a cool setup. I was thinking of doing similar with the Evo interfaces, but I don’t know much about ADAT in terms of cable length supported or robustness of connections (other than they seem delicate and have a tendency to pop out and the slightest bump).
The thing I don’t love about the EVOs is not having all inputs on the back (just for my setup anyway).
So has anybody here gotten one of these? I’m likely going to upgrade from my 18i20s to this + one 18i20 soon and I’d love to hear the experiences of users here if they’ve got one.
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I’d like to know also. YouTube isn’t a reliable source of real-world info for new things at all, and there are very few product reviews on retailers either (not that they’re much better than YouTube, but the low-star ones can be useful).
You can get 120 US off at a few retailers, but it’s still over a grand and that’s a lot for a ‘hope it works as advertised’ piece of gear
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I am thinking about taking the risk on it, but I’ve also been hoping to hear from someone who has tried it already. I did find a Sound on Sound review. Unfortunately, the article is mostly paywalled, but I was able to read a copy of the article through my local library’s online magazine access. It’s pretty positive in general. The main complaints were:
- Uses class compliant OS drivers instead of custom ones, so latency isn’t spectacular (my main concern)
- Wall wart PSU with a too-short cable (sounds annoying but not a deal breaker)
- Lose channel count with high sample rates (I work at 48, and isn’t this inherently true of ADAT?)
- Introduced noise when re-amping in his testing (somewhat concerning but possibly a one off? not something I need anyway)
- No talkback functionality (irrelevant to me and I would imagine most of the intended audience)
That said, the review doesn’t seem particularly in depth or scientific. Mostly a review of the features as advertised.
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Thanks! I wonder whether this is related to my only concern, the lack of pre-amps (Arturia AudioFuse 16 Rig - #73 by icaria36).
There aren’t a lot of details about how he tested, but if he was actually re-amping I wouldn’t have expected it to need another preamp afterwards. I had the impression that the problem was with the front panel outputs.
According to ableton, I currently have 12.4ms input and output latency with 256 samples at 44.1 in ableton with 2 focusrite 18i20s. As long as the latency is as good or better than that, I’ll be happy. But I’d love to see numbers (I think one of the video reviews has some but I don’t remember which one).
Yeah, I’m kind of in the same boat, coming from a Scarlett myself. I think Julian Krause’s review tested latency. The Sound on Sound article mentioned measured latency of 7.5ms at 44.1 kHz and 32 sample buffer size. When I set my Scarlett to that, Ableton reports 8.59ms overall latency. At this point I think we’re splitting hairs, but I don’t know whether running at that buffer size will work out in practice. I usually run at 48/128, which my Scarlett reports as 11.6ms. Supposedly RME has less latency, but I think the workflow and I/O count of the 16Rig wins out for me. To be honest, it hasn’t been a huge problem for me as it is, and I usually use direct monitoring when tracking, which the 16Rig supports well.
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I’m rewatching the Julian Krause video. Here are the latencies that he saw:
| Sample Rate (Hz) | Buffer Size (samples) | RTL (Ms) |
|---|---|---|
| 48000 | 8 | 6.3 |
| 48000 | 16 | 6.9 |
| 48000 | 32 | 7.1 |
| 48000 | 64 | 8.1 |
| 48000 | 128 | 11.0 |
| 48000 | 256 | 16.7 |
Not outstanding, but I think I can live with it. He also tested at 192 kHz and the numbers are similar, but not at 44.1. I think he tested on Windows, which might differ slightly from Mac.
I also looked at his RME Babyface Pro FS video for comparison. Unfortunately, he didn’t test at all of the same buffer sizes, but for comparison, 64 samples is 3.8 ms, 128 is 6.5, 256 is 11.8.
Here’s what the SoS review wrote about the re-amping:
My own experiments with the re-amping were marred by hiss and other noise pollution, which seemed to originate either from the PSU or the USB connection to my Mac, or both. I never managed to get to the bottom of this during the review period.
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I’ve got a bit of a strange hybrid setup in my hobby room, as it contains a studio desk (via a TX-6), a DJ desk (with a Xone:96) and a rack with a consoles plus home theater gear (with a Denon AV amp). I’ve got it set up with 3 monitor speakers (2 fronts & 1 center) and a sub, and will get two more (identical) monitors tomorrow to complete a 5.1 setup.
I’m thinking of getting a 16Rig to manage my speaker setup and the 3 kinds of sources:
- DJ: stereo out from the Xone -> 2 inputs on the 16Rig
- Synths: mixed stereo out from the TX-6 (or plugging in synths directly) -> 2 inputs on the 16Rig (more if synths plugged into 16Rig direct)
- Movies/games: Decoded Dolby/DTS 5.1 analog out -> 6 inputs on the 16Rig
On the 16Rig outs, I’d hook up the 5 monitor speakers, plus the sub. Then in the matrix mixer of the 16Rig I’d make presets for the various setups:
- Stereo monitoring, both for DJ and studio use)
- Stereo to 5.1 (summing to mono for the center and sub channels), so DJ or studio audio goes into party/club mode
- 5.1 passhtrough from the AV processor for surround movies/games etc.
I should be able to use it like this, right?
And with 10 of the 16 inputs and 6 of the 10 outputs used, I could maybe have separate drum / melody bus and/or hook up a stereo FX unit. Or maybe get an ADAT expansion and directly connect all my synths/drum computers to it directly instead of through the TX-6. I’m curious to hear more about real world ADAT uses…
That’s not great but not awful, I guess? I honestly don’t know how much of a difference a handful of ms even makes in practice. I agree we’re likely splitting hairs, I just don’t want to spend this much money and then in my ignorance make things painful for myself when recording or when monitoring while jamming.
If those numbers above are accurate for the AudioFuse 16 you’d probably want to stick to a buffer size of 64. You might notice a delay at 128 samples, and you’d almost surely notice the delay at 256 samples. Latency of ~11-12 ms is often cited as the point where people start to notice the latency.
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Update: So I may have misspoken earlier because I didn’t understand what I was talking about in my previous comments. My current Overall Latency (which I believe represents Roundtrip Latency as others have phrased it, please correct me if I’m wrong) is 24ms (yikes!). The 16 Rig, according to the numbers above and in a couple other videos, should be a fairly large improvement over what I’m currently dealing with given the same settings, unless I grossly misunderstanding something, and assuming the drivers work just as efficiently on Windows. I can’t know for sure until I try it, but is this understanding correct?
My current latency:
Bonus question: is there a downside to dropping the buffer size to the minimum I can use without the audio popping/crackling? I can drop as low s 128 with no issues according to Ableton’s built-in latency test. This is an area where my knowledge is pretty lacking. I’ve finished a few full albums like this, but this might explain why I have to spend some time lining things up on the grid occasionally.
I’m not aware of any downsides. If you can go lower, go lower. But yeah, the latency on this thing is a bit disappointing.
The smaller the buffer, the lower the latency, but also the more trips required to ferry data between the audio interface and host computer. If your machine has cycles to spare, this shouldn’t be a problem. If your system is marginal, smaller buffers may overwhelm it. If it works, it works. But if plugins struggle, review your buffer size again.
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And i move things deliberatly away to get some microtiming groove ^^ - for sure i know this obsession. I come to the conclusion that live isnt optimal for recording, and bought a external midi clock - it works, but not with projects having 100 tracks, and compression etc. Direct monitoring can help, but my interface dosent have that.
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