Starise
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Maximizing Audio Interface Performance
I'm trying to squeeze as much out of my Audio interface as possible in terms of minimizing audio latency. I have been playing with buffer settings and resolutions in the setup and I'm getting mixed results. I tried a buffer of 512 at 24/48 on a 20 plus channel interface. The results were pretty bad at around 21ms round trip latency. My best result so far has been a setting of 24/96. This reduced the round trip latency to 10ms. I can go to 256 buffers, but haven't tried that yet. 5ms one way latency is what I get at 24/96. Not too shabby for simply playing with amp sims. Recording them while live is another story though. Are there any other things I could do to improve those numbers aside from getting another audio interface? What if I would eliminate all channels but the ones I'm recording with, would that minimize my latency numbers?
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bitflipper
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Re: Maximizing Audio Interface Performance
2016/04/20 11:00:18
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Another interface isn't likely to lower latency by much, perhaps by as little as half a millisecond or none at all. While interface drivers do vary somewhat in efficiency, most of the latency is attributable to buffering, which has nothing to do with the interface itself, or its driver. Buffers fill at a constant rate determined by the sample rate, so a given buffer size and rate will always yield the same latency. The only way to lower latency is to reduce the buffer size or increase the sample rate. How small a buffer you can get away with depends on your computer's power, the efficiency of its peripherals, the overhead incurred by background processes and hardware interrupt handlers, and the amount of processing done within the DAW. With so many variables, there is no single recipe for making the computer more efficient and thereby allowing you to reduce buffer sizes. All you can do is determine the smallest buffer size that doesn't result in dropouts, and then gradually figure out what changes help in that regard. If you're lucky, you'll identify one major culprit, such as a wireless network adapter or a greedy video card that's sucking up CPU cycles. You may also decide to adopt a two-step method, wherein you do all your tracking at a very low buffer size (e.g. 64 bytes) before adding any effects to the project, and then raising the buffer to a higher value during mixing and mastering. Plugins not only increase demands on the CPU, but may also incur extra latency even with the same audio interface buffers.
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Jim Roseberry
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Re: Maximizing Audio Interface Performance
2016/04/20 11:34:18
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☄ Helpfulby tlw 2016/04/20 12:57:09
The only means of reducing round-trip latency: - Use smaller ASIO buffer size
- Reduce the size of the safety-buffer (most units don't allow this)
- Increasing the sample-rate (doubling the sample-rate will roughly cut RTL in half - at the expense of higher CPU use)
Nothing else will have an effect on round-trip latency (disabling channels on the unit will have no effect).
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Jim Roseberry
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Re: Maximizing Audio Interface Performance
2016/04/20 11:43:54
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If you want to get lower than 5ms round-trip latency at a 64-sample ASIO buffer size, you have two options: - PCIe audio interface (one that allows using a 32-sample or 16-sample ASIO buffer size)
- Thunderbolt audio interface with full "PCIe via Thunderbolt" drivers/support
Win10 now offers "PCIe via Thunderbolt" support for Thunderbolt 3. Have to be running one of the latest generation Z170x or X99p motherboards that supports TB3 via USB-C Microsoft claims that TB3 support should be backward compatible with TB2 and TB1. Right now, all TB audio interfaces are TB2. Thus, you need a USB-C to Thunderbolt adapter. USB-C to TB adapters have been announced... but I've yet to see one actually available.
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JonD
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Re: Maximizing Audio Interface Performance
2016/04/20 12:13:22
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You say you haven't tried a buffer of 256 yet. Why not? Generally, you want to work with the lowest buffer your system will allow and still perform without problems. (This part edited): Jim beat me to it (and said it better).
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tlw
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Re: Maximizing Audio Interface Performance
2016/04/20 12:46:26
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Jim Roseberry If you want to get lower than 5ms round-trip latency at a 64-sample ASIO buffer size, you have two options:- PCIe audio interface (one that allows using a 32-sample or 16-sample ASIO buffer size)
- Thunderbolt audio interface with full "PCIe via Thunderbolt" drivers/support
Or settle for around 6 or 7ms round-trips and get an RME USB/firewire interface. To which has to be added 1ms for every foot between ears and monitors of course. If someone can't cope with playing with their instrument speaker or stage foldback six feet away from them, reducing latency below 6ms is unlikely to help them much.
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Starise
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Re: Maximizing Audio Interface Performance
2016/04/20 14:59:13
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Thank you all for the help!! I'm not close to my DAW and won't be until late next week at the soonest. I'm anxious to try 256 sample rate at 24/96 . I believe if I can get down to around 8ms I'll be happy. I guess there's no point in attempting to disable tracks. tlw I would be overjoyed to get a 6ms round trip latency on usb.
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bapu
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Re: Maximizing Audio Interface Performance
2016/04/21 11:25:08
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On my RME UFX @ 64 buffers @44.1K/24 (connected via UB) SONAR reports 5.1ms RTL. Also if you must track with FXs on other tracks, you can always freeze those other tracks.
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Sonico
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Re: Maximizing Audio Interface Performance
2016/04/21 12:09:30
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I have a Focusrite Scarlett 18i20 and get 9.2ms @64 buffers and 44.1khz The 18i20 has three performance modes (recording, balanced and mixing), each increases the RTL accordingly at the same buffer and sample rate. I'am not sure if that is the safety buffer? Also it allows me to select 32 and even 16 buffers, I have used it with such a small buffer size and It can work for recording, but I' am very happy working at 64 buffers with 9.2ms and I can mix without pops or dropouts. Hope it helps!
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Jim Roseberry
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Re: Maximizing Audio Interface Performance
2016/04/26 17:17:17
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tlw
Jim Roseberry If you want to get lower than 5ms round-trip latency at a 64-sample ASIO buffer size, you have two options:
- PCIe audio interface (one that allows using a 32-sample or 16-sample ASIO buffer size)
- Thunderbolt audio interface with full "PCIe via Thunderbolt" drivers/support
Or settle for around 6 or 7ms round-trips and get an RME USB/firewire interface.
To which has to be added 1ms for every foot between ears and monitors of course.
If someone can't cope with playing with their instrument speaker or stage foldback six feet away from them, reducing latency below 6ms is unlikely to help them much.
FWIW, I play out quite a bit. Once you start to venture too far away (without IEMs), you can *really* start to feel the lag (latency). Can you compensate? Yes. Does it affect feel? Absolutely. Lower latency feels tighter. I'm pleased with the RTL of my Fireface UFX. That said, if I can get the RTL lower (which feels even better), I'm all for it. Don't agree that lower latency feels "better"? Load up your favorite piano sample library. Play it at a 256-sample ASIO buffer size. Now drop the ASIO buffer size to 64-samples. Which feels more immediate/responsive? To my ears/sensibilities... it's easy to hear/feel the difference.
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JonD
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Re: Maximizing Audio Interface Performance
2016/04/26 19:41:27
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Guys, speaking of round-trip low-latencies, I've just listed an RME FF400 on Ebay. I've accumulated too many interfaces, and it's just sitting in the closet, so the RME lost the flip of the coin (Actually, I figured that would likely be the easiest to sell). www.ebay.com/itm/272225150783 Because it's not cosmetically perfect, I'm letting it go on the low-ish side. So if you've ever desired an RME interface, but couldn't justify the $1K+ price tag, here's your chance to get a good deal on one. Also, no worries if your PC lacks firewire -- I'm throwing in a SIIG PCIe Firewire card (TI chipset), which has always worked great with all of my FW devices.
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tlw
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Re: Maximizing Audio Interface Performance
2016/04/26 19:42:14
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I've done more than my fair share of gigs where someone's bass amp is 20 feet away from me and vocal/acoustic instruments only in the foldback. Keeping in time becomes as much a matter of watching them as listening to them at that range. This being in a band with no drums, so no-one in the middle of it all to act as a fixed reference point.
I can't handle bass or guitar at much over 10ms latency, my timing just collapses and eveything feels wrong when I feel the fingers hit the string then the note sounds after a distinct gap. Walking out among the audience at the end of a 150' cable might have worked for Albert Collins, but not me.
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Cactus Music
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Re: Maximizing Audio Interface Performance
2016/04/27 23:49:22
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Why I love playing live with in ears now,,, even in my solo act, and why I do most tracking with phones on or sitting less than 2' away from the speakers. And why I track with most efxs disabled and never in a million years monitoring the back end of my interface. My system is about 7 to 9 ms RTL but because I never listen that way it's a none issue.
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Jim Roseberry
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Re: Maximizing Audio Interface Performance
2016/04/28 08:05:06
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I move around a lot when playing... Tried a set of non molded IEMs. Every time I'd move... they'd pop out of place. Without IEMs, on larger stages it's easy to get more than 20' from your monitor. If you roam over to the guitar player and your not in his/her monitor, it can be harder to hear yourself... and then you're dealing with significant latency. The advantage to that much physical space: Far less prone to feedback issues
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Cactus Music
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Re: Maximizing Audio Interface Performance
2016/04/29 23:01:31
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Because I'm always on the mike, I've gone with a hardwired iem system. I removed the right ear bud from my Shure SE215's so I only use one side. This I find gives me the best of both worlds. With both buds in I find I'm too disconnected with the stage sound which after 40 years is just to weird to deal with. In our band this also puts it on the side with the drums and bass so they are attenuated. I can still hear my guitar amp in my right ear the way I've always heard it. I use a Rolls PM 50 which is on my mike stand. My vocal mike can be made as loud as I can stand it, and then I blend in the monitor mix. Perfect. Total control finally. As I say, even in my solo act this tightens up my timing as I'm hearing my backing tracks bang on in my left ear. I can also stay in touch with the audience if they come up and talk to me. So once again,,,, latency is not intuitive to a good performance. In the studio or on stage. Long live IEM and headphones...
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orangesporanges
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Re: Maximizing Audio Interface Performance
2016/05/14 01:32:31
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I like where this thread is going, because some have chimed in with real world examples, like how far you are from your speakers before you lose your "lock" with yourself or your bandmates. If you do some quick math, sound travels about 1 foot per millisecond (700mph x 5280 ÷ 3600 seconds in an hour ÷1000ms=1.02). So ask yourself how far away from an amp can you play before you start to feel the lag? 10ft? 20 ft? That's how many ms of latency you're feeling. When listening to monitors , how far away are you? factor that into your RTL. So if you are , say 8ft and 20ms RTL ,you are really feeling 20 ms RTL + 8. If 20 is the"magic " number where yo really start experiencing it, you should be shooting for 5-10 ms. if you can hit this sweet spot, you're in good shape. Less than 5ms RTL will most likely be imperceptible.
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