New audio interface - need a lower latency
New audio interface - need a lower latency
Hi All
I've been using a steinberg UR22 for a while now. It's been pretty solid but sessions are getting heavier and when it comes to the end of a project I'm having to lower the buffer size to the maximum (2048 samples) in Cubase. That's an input latency of 47 and an output of 51 msec!!
I used to have an EMU1212m and it served me very well. So I'm wondering if it would be prudent to go back to pcie or maybe thunderbolt? Any recommendations?
I recently upgraded to 32gb Ram from 16gb but I've barely noticed a difference.
I've always been a tad confused about audio latency to be honest...
If anyone can shed some light on the subject I'd greatly appreciate it.
I only need 2in 1 out device and within reason, budget is not a problem. I'd rather not go with UAD.
Thanks for any help!
I've been using a steinberg UR22 for a while now. It's been pretty solid but sessions are getting heavier and when it comes to the end of a project I'm having to lower the buffer size to the maximum (2048 samples) in Cubase. That's an input latency of 47 and an output of 51 msec!!
I used to have an EMU1212m and it served me very well. So I'm wondering if it would be prudent to go back to pcie or maybe thunderbolt? Any recommendations?
I recently upgraded to 32gb Ram from 16gb but I've barely noticed a difference.
I've always been a tad confused about audio latency to be honest...
If anyone can shed some light on the subject I'd greatly appreciate it.
I only need 2in 1 out device and within reason, budget is not a problem. I'd rather not go with UAD.
Thanks for any help!
Last edited by Forum Admin on Tue Apr 24, 2018 3:10 pm, edited 1 time in total.
Re: New audio interface - need a lower latency
I am sure a "proper" PC audio person like Martin will be along very soon but here's my 2p'oth.
I have recommended the UR22 many times to people on forums (usually as an alternative to the ubiquitous "2i2"!) and always assumed it had pretty low latency coming as it did from the Steinberg stable? I have certainly never read of any such problems with the AI.
My top recommendation is still the Native Instruments Komplete Audio 6 which I have run at 32 samples on a not particularly swift i3 laptop. You also get a 'spare' copy of a Cubase.
Dave.
I have recommended the UR22 many times to people on forums (usually as an alternative to the ubiquitous "2i2"!) and always assumed it had pretty low latency coming as it did from the Steinberg stable? I have certainly never read of any such problems with the AI.
My top recommendation is still the Native Instruments Komplete Audio 6 which I have run at 32 samples on a not particularly swift i3 laptop. You also get a 'spare' copy of a Cubase.
Dave.
Re: New audio interface - need a lower latency
Hmmm... for recording audio that should be a huge issue, as there's a blend knob on the front of the unit to mix the incoming signal aligned to the playback, whatever the buffer.
So I'm guessing you need to be able to hear either some processing (eg a guitar amp model) in real time, or to use MIDI to trigger soft synths etc in real time?
The thing is, irrespective of whether a new interface would reduce the latency a little, and *any* interface at 2048 samples buffer will exhibit too much latency for that sort of work. Which suggests you need to find a way to work without such large buffers.
That's going to mean either working smarter or tweaking/upgrading your machine so it's capable of running more at lower buffer settings.
Working smarter examples include (1) reassessing your workflow, maybe to compose with less CPU intensive instruments and replace them after you've finished recording, or to record things and bounce the results as audio for a later mixing stage, where the latency is less critical; (2) trying to avoid channel presets that put lots of plug-ins on each channel; (3) making more use of your DAW's Freeze function to free up resources; and (4) examining your signal path in the DAW — find out if there's any long string of plug-ins on channels and busses that are considerably adding to the latency, such as Nebula/Acqua, convolution, and linear-phase plug-ins, which are notorious for this; leave that stuff for later) .
Re upgrading the machine — I doubt if it's the RAM, so probably the CPU. But it might be worth investigating if there are any tweaks to your current machine that could remove obstacles to better performance. Martin and Pete are the guys to ask about that but, for example, it might be that the computer's audio performance is being throttled due to network activity. I had massive issues with a USB wireless adapter once upon a time, and disabling that improved things no end!
Good luck!
So I'm guessing you need to be able to hear either some processing (eg a guitar amp model) in real time, or to use MIDI to trigger soft synths etc in real time?
The thing is, irrespective of whether a new interface would reduce the latency a little, and *any* interface at 2048 samples buffer will exhibit too much latency for that sort of work. Which suggests you need to find a way to work without such large buffers.
That's going to mean either working smarter or tweaking/upgrading your machine so it's capable of running more at lower buffer settings.
Working smarter examples include (1) reassessing your workflow, maybe to compose with less CPU intensive instruments and replace them after you've finished recording, or to record things and bounce the results as audio for a later mixing stage, where the latency is less critical; (2) trying to avoid channel presets that put lots of plug-ins on each channel; (3) making more use of your DAW's Freeze function to free up resources; and (4) examining your signal path in the DAW — find out if there's any long string of plug-ins on channels and busses that are considerably adding to the latency, such as Nebula/Acqua, convolution, and linear-phase plug-ins, which are notorious for this; leave that stuff for later) .
Re upgrading the machine — I doubt if it's the RAM, so probably the CPU. But it might be worth investigating if there are any tweaks to your current machine that could remove obstacles to better performance. Martin and Pete are the guys to ask about that but, for example, it might be that the computer's audio performance is being throttled due to network activity. I had massive issues with a USB wireless adapter once upon a time, and disabling that improved things no end!
Good luck!
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- Matt Houghton
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Re: New audio interface - need a lower latency
Thanks guys. I'm using an ethernet connection for internet so it shouldn't be a problem.
You're right, I didn't specify the latency problems that affect me personally. I'm not worried about audio in latency eg real guitars, bass , vox etc as I can use direct monitoring without fx.
It's just a bit of a nightmare when I come to the end of a project and maybe need to play a midi piano solo or whatever - a small amount of latency is workable but right now it's impossible. I try to bounce down big vst instruments like AD drums, bfd drums or Omnisphere but it's cumbersome. Often I have to do the string, drum and horns parts in a different session. I just can't imagine that the big studios have to perform this workaround.
Maybe I'm wrong however!
I'm seriously thinking of going back to pci-e as I definitely don't want to change my motherboard (no thunderbolt) for at least another 3 years - we'd be talking at least one week off in order re-install everything.
I guess a lot of this comes down to expectancy. Maybe it's just impossible to write and mix in the box with the types of tracks I'm commissioned to write ;(
You're right, I didn't specify the latency problems that affect me personally. I'm not worried about audio in latency eg real guitars, bass , vox etc as I can use direct monitoring without fx.
It's just a bit of a nightmare when I come to the end of a project and maybe need to play a midi piano solo or whatever - a small amount of latency is workable but right now it's impossible. I try to bounce down big vst instruments like AD drums, bfd drums or Omnisphere but it's cumbersome. Often I have to do the string, drum and horns parts in a different session. I just can't imagine that the big studios have to perform this workaround.
Maybe I'm wrong however!
I'm seriously thinking of going back to pci-e as I definitely don't want to change my motherboard (no thunderbolt) for at least another 3 years - we'd be talking at least one week off in order re-install everything.
I guess a lot of this comes down to expectancy. Maybe it's just impossible to write and mix in the box with the types of tracks I'm commissioned to write ;(
Re: New audio interface - need a lower latency
Maybe you could write the midi parts and use a hardware synth for playback until it’s time to mix? A multitimbral machine like the Yamaha MX49 would be perfect for that.
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Re: New audio interface - need a lower latency
I would have to say that this sounds more like a Cubase issue than an interface issue. It seems that Cubase may not be giving the audio interface handling a high enough priority. Have you tried muting individual tracks to see which ones are causing problems?
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Re: New audio interface - need a lower latency
lovesexy wrote:Thanks guys. I'm using an ethernet connection for internet so it shouldn't be a problem.
So was I. But the wireless facility was still enabled in Windows, and it was causing all sorts of issues. My point wasn't that your internet connection might be causing problems. But that something as yet unidentified in your PC configuration might be causing problems. And that might be causing you to use more CPU power. And that in turn might be leading you to increase the buffer. Latency Mon can help you identify such issues... but you'll no doubt need help interpreting the results.
If you're able to achieve suitably low latencies at smaller buffer settings at the start of your projects, then it's not the interface that's the main source of trouble. It's something to do with your computer or the DAW software/plug-ins running on it that is causing the system to underperform/use up its resources, and therefore causing you to increase the buffer accordingly.
Can you tell us more about the computer's specs, and more about what type of and how many FX and instrument plug-ins you're attempting to run when you hit problems?
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- Matt Houghton
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Re: New audio interface - need a lower latency
I can check but it's a gradual thing really so it seems to make sense. I can start off with BFD, Omnisphere, Nectar and a few other VST instruments but once I start mixing the track I have to lower the latency or else I get massive peaks, artifacts etc. I mean we're talking a lot of data here. 64 tracks, 10 groups almost all full with plugins, 3 reverbs, maybe one delay.
It's true that my way of writing poses problems because I'm essentially mixing as I'm composing. I guess I'm asking alot of a USB device.
It's true that my way of writing poses problems because I'm essentially mixing as I'm composing. I guess I'm asking alot of a USB device.
Re: New audio interface - need a lower latency
Matt Houghton wrote:lovesexy wrote:Thanks guys. I'm using an ethernet connection for internet so it shouldn't be a problem.
So was I. But the wireless facility was still enabled in Windows, and it was causing all sorts of issues. My point wasn't that your internet connection might be causing problems. But that something as yet unidentified in your PC configuration might be causing problems. And that might be causing you to use more CPU power. And that in turn might be leading you to increase the buffer. Latency Mon can help you identify such issues... but you'll no doubt need help interpreting the results.
If you're able to achieve suitably low latencies at smaller buffer settings at the start of your projects, then it's not the interface that's the main source of trouble. It's something to do with your computer or the DAW software/plug-ins running on it that is causing the system to underperform/use up its resources, and therefore causing you to increase the buffer accordingly.
Can you tell us more about the computer's specs, and more about what type of and how many FX and instrument plug-ins you're attempting to run when you hit problems?
Do you know where I can turnoff the wifi facility?
Specs:
intel i7-4790 cpu 4.00ghz
32gb ram
Windows 7 ultimate 64bit
Re: New audio interface - need a lower latency
I'm confused.
You say that you have to raise the audio buffer to prevent dropouts / cpu overloads with your project. This is normal for cpu-heavy projects - we all do that and have done for years. It's the audio buffer which is causing the latency for your VSTi overdubs - not the audio interface. Typically, the AI only adds on a very small amount. I think I measured mine to be about 32samples...
Unless I'm missing something here, it sounds like you need to start making more use of something like the track freezing to lower your cpu load. If you keep the resources down to a minimum, you can keep the audio buffer low and ensure your latency is acceptable.
No?
You say that you have to raise the audio buffer to prevent dropouts / cpu overloads with your project. This is normal for cpu-heavy projects - we all do that and have done for years. It's the audio buffer which is causing the latency for your VSTi overdubs - not the audio interface. Typically, the AI only adds on a very small amount. I think I measured mine to be about 32samples...
Unless I'm missing something here, it sounds like you need to start making more use of something like the track freezing to lower your cpu load. If you keep the resources down to a minimum, you can keep the audio buffer low and ensure your latency is acceptable.
No?
Veni, Vidi, Aesculi (I came, I saw, I conkered)
Re: New audio interface - need a lower latency
lovesexy wrote:Matt Houghton wrote:lovesexy wrote:Thanks guys. I'm using an ethernet connection for internet so it shouldn't be a problem.
So was I. But the wireless facility was still enabled in Windows, and it was causing all sorts of issues. My point wasn't that your internet connection might be causing problems. But that something as yet unidentified in your PC configuration might be causing problems. And that might be causing you to use more CPU power. And that in turn might be leading you to increase the buffer. Latency Mon can help you identify such issues... but you'll no doubt need help interpreting the results.
If you're able to achieve suitably low latencies at smaller buffer settings at the start of your projects, then it's not the interface that's the main source of trouble. It's something to do with your computer or the DAW software/plug-ins running on it that is causing the system to underperform/use up its resources, and therefore causing you to increase the buffer accordingly.
Can you tell us more about the computer's specs, and more about what type of and how many FX and instrument plug-ins you're attempting to run when you hit problems?
Do you know where I can turnoff the wifi facility?
Specs:
intel i7-4790 cpu 4.00ghz
32gb ram
Windows 7 ultimate 64bit
I don't think you're taking in what I'm saying. Don't just randomly start turning off things willy nilly. I was offering my problem as an example of the sort of things that can possibly lurk behind the scenes.
I'll try to simplify:
1. You're turning up the buffer because you're maxing our your system at low buffer settings, right?
2. Maxing out your system might be due to hardware problems, or it might be due to the number of CPU-hogging plug-ins you're attempting to run simultaneously. If you get low latencies at smaller buffer settings, there's no problem with your audio interface, so don't spend money on a new one hoping this will change things.
3. You need to figure out which of (2) is the case, and then figure out the best way of resolving it: ie. (a) identifying the specific problem and fixing your machine if there are problems; (b) running fewer CPU-hogging plug-ins simultaneously; or (c) upgrading your machine to cope with the CPU-hogging plug-ins.
4. An alternative is just to rethink your workflow to use the tools at your disposal — work out your arrangements and record them. Mix them on a separate project. Make efficient use of reverbs/delays as sends instead of inserts. Use less CPU-hogging plug-ins while you're playing parts in and replace them with more demanding ones later. Etc.
Last edited by Forum Admin on Tue Apr 24, 2018 3:07 pm, edited 1 time in total.
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Re: New audio interface - need a lower latency
Dave B wrote:I'm confused.
You say that you have to raise the audio buffer to prevent dropouts / cpu overloads with your project. This is normal for cpu-heavy projects - we all do that and have done for years. It's the audio buffer which is causing the latency for your VSTi overdubs - not the audio interface. Typically, the AI only adds on a very small amount. I think I measured mine to be about 32samples...
Unless I'm missing something here, it sounds like you need to start making more use of something like the track freezing to lower your cpu load. If you keep the resources down to a minimum, you can keep the audio buffer low and ensure your latency is acceptable.
No?
I think you're right. it's just that I'm a little disappointed given that my system is half decent and I was wondering if changing to PCIe might give me a little more room for manoeuvre. Maybe I need to do some monitoring of CPU usage when playing large projects in order to get an idea of what's going on.
Re: New audio interface - need a lower latency
Here's a screenshot of the same project at 192 samples and 2048.
It's not like the CPU or RAM is being maxed at 192 but Cubase is suffering like a dog.
https:*?*?*/nyKYjH
https:*?*?*/eV1Prx
It's not like the CPU or RAM is being maxed at 192 but Cubase is suffering like a dog.
https:*?*?*/nyKYjH
https:*?*?*/eV1Prx
Re: New audio interface - need a lower latency
What you are describing is just the classic issue of using a DAW. Any DAW.
The solution is simple. Monitor in the analog domain when recording. That's how it's done. The other way is to bounce a cue mix of your complex session, and let the performer track to that using the DAW as a VI with a low buffer setting, if you only have 1 computer.
'Real' studios aren't going to let you use their tracking and mixing PC to play your VI. You have to show up with your own PC to run the VI so it is just like any other instrument and can be monitored in the analog domain.
You can't combine a complex session that needs a large buffer with real-time monitoring of a VI at the same time on a single computer. This is the basics of digital audio.
So just bounce a cue mix, bring it into a new session with a small buffer. Play your VI along with the cue mix. Then bring that audio back into your mix-in-progress session. That is effectively the same thing as doing a 'track freeze' on all your tracks and how we've always managed to get along with only one PC.
The solution is simple. Monitor in the analog domain when recording. That's how it's done. The other way is to bounce a cue mix of your complex session, and let the performer track to that using the DAW as a VI with a low buffer setting, if you only have 1 computer.
'Real' studios aren't going to let you use their tracking and mixing PC to play your VI. You have to show up with your own PC to run the VI so it is just like any other instrument and can be monitored in the analog domain.
You can't combine a complex session that needs a large buffer with real-time monitoring of a VI at the same time on a single computer. This is the basics of digital audio.
So just bounce a cue mix, bring it into a new session with a small buffer. Play your VI along with the cue mix. Then bring that audio back into your mix-in-progress session. That is effectively the same thing as doing a 'track freeze' on all your tracks and how we've always managed to get along with only one PC.
Last edited by DC-Choppah on Tue Apr 24, 2018 6:21 pm, edited 2 times in total.
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Re: New audio interface - need a lower latency
Disclaimer: I'm not familiar with cubase.
My first thought is, why a buffer size of 192? Most computery stuff likes powers of 2, so 128, 256, 512 etc. give greatest buffer shifting efficiency.
Next, what is your sample rate? Is it possibly higher than you really need?
Processor load figures have to be taken with (at least) a pinch of salt, and are usually higher than they seem. Only the CPU itself knows how hard it is working. Having any kind of network (or other background) process running can invisibly soak up some cycles.
Independent per channel effects are notoriously greedy - especially reverbs. Do you have any built in that could be on a common bus?
Just my 2d (old money)
My first thought is, why a buffer size of 192? Most computery stuff likes powers of 2, so 128, 256, 512 etc. give greatest buffer shifting efficiency.
Next, what is your sample rate? Is it possibly higher than you really need?
Processor load figures have to be taken with (at least) a pinch of salt, and are usually higher than they seem. Only the CPU itself knows how hard it is working. Having any kind of network (or other background) process running can invisibly soak up some cycles.
Independent per channel effects are notoriously greedy - especially reverbs. Do you have any built in that could be on a common bus?
Just my 2d (old money)
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Re: New audio interface - need a lower latency
The "playing MIDI along with a track" comment gave me pause (if I understand you correctly? NOT a given!)
Several years ago my son (he musical and plays, Me not, solder and pays) had a problem with playing piano parts along with existing guitar parts using a MIDI keyboard controller.
He said the latency just playing the music from the Cubase instrument was fine, it was just keeping 'in time' when following an existing tune that was the problem.
This was using a single core, 3.2G PC on XP and a 2496 card. After some fettling we found we could use down to 128 samples and no glitching. 64 samples is as low as the 2496 would go and was great for playing but stuttered a bit, 128 he said he could just about live with if he concentrated! Note though that the computer was doing practically nothing else and XP was a pretty simple OS. Note too that ONLY the 2496 and Cubase (Ess 6) would run this low latency.
I then bought another desktop PC(V cheap PCW Jan sales jobby) and linked the two with S/PDIF but ran the soundcards into a small mixer. Thus he could copy tracks over and play them out on one PC and play MIDI in on tother.
There is usually always SOME kludge you can pull!
I later found the KA6 would hiss all over 128 samples for MIDI and stayed clean for 64 samples. Grot started at 32 samples but a faster PC would probably cope?
Dave.
Several years ago my son (he musical and plays, Me not, solder and pays) had a problem with playing piano parts along with existing guitar parts using a MIDI keyboard controller.
He said the latency just playing the music from the Cubase instrument was fine, it was just keeping 'in time' when following an existing tune that was the problem.
This was using a single core, 3.2G PC on XP and a 2496 card. After some fettling we found we could use down to 128 samples and no glitching. 64 samples is as low as the 2496 would go and was great for playing but stuttered a bit, 128 he said he could just about live with if he concentrated! Note though that the computer was doing practically nothing else and XP was a pretty simple OS. Note too that ONLY the 2496 and Cubase (Ess 6) would run this low latency.
I then bought another desktop PC(V cheap PCW Jan sales jobby) and linked the two with S/PDIF but ran the soundcards into a small mixer. Thus he could copy tracks over and play them out on one PC and play MIDI in on tother.
There is usually always SOME kludge you can pull!
I later found the KA6 would hiss all over 128 samples for MIDI and stayed clean for 64 samples. Grot started at 32 samples but a faster PC would probably cope?
Dave.
Re: New audio interface - need a lower latency
Hi lovesexy,
My current machine seems vaguely similar in spec to yours (a quad-core running at ~ 4GHz), but I don't have any of the stuttering problems that you seem to be suffering from.
With this amount of computing power at my disposal I can run far more plug-in effects than I have yet needed, and it only tends to be playing softsynths with large polyphony and high quality settings that put any noticeable strain on my system. However, given that my projects currently involve as many recorded audio tracks of electric and acoustic instruments as there are virtual ones being created by software 'on the hoof', I mostly seem to stay well within its computing limitations. As a ballpark, I typically run around 30 audio/synth tracks and 50 plug-in effects per project.
However, like plenty of other musicians I'm quite happy running my projects at a sample rate of 44.1kHz, and I'm also happy working with a fairly modest audio interface buffer size of 128 samples, which at 44.1kHz gives me a fairly snappy 5mS latency when performing MIDI-based softsynths in 'real time' (a fixed latency of 1mS or so for the MIDI input, 3mS latency for the audio interface playback buffers, plus a further 1mS for the D/A converters in the interface).
If you're suffering from stuttering with such settings then as Matt suggests, you could have a PC device occasionally taking more than its fair share of Windows resources. Having said that, your two Task Manager screengrabs show a fairly smooth allocation of resources across your available cores.
I think we need more information before we can offer more specific advice
Martin
My current machine seems vaguely similar in spec to yours (a quad-core running at ~ 4GHz), but I don't have any of the stuttering problems that you seem to be suffering from.
With this amount of computing power at my disposal I can run far more plug-in effects than I have yet needed, and it only tends to be playing softsynths with large polyphony and high quality settings that put any noticeable strain on my system. However, given that my projects currently involve as many recorded audio tracks of electric and acoustic instruments as there are virtual ones being created by software 'on the hoof', I mostly seem to stay well within its computing limitations. As a ballpark, I typically run around 30 audio/synth tracks and 50 plug-in effects per project.
However, like plenty of other musicians I'm quite happy running my projects at a sample rate of 44.1kHz, and I'm also happy working with a fairly modest audio interface buffer size of 128 samples, which at 44.1kHz gives me a fairly snappy 5mS latency when performing MIDI-based softsynths in 'real time' (a fixed latency of 1mS or so for the MIDI input, 3mS latency for the audio interface playback buffers, plus a further 1mS for the D/A converters in the interface).
If you're suffering from stuttering with such settings then as Matt suggests, you could have a PC device occasionally taking more than its fair share of Windows resources. Having said that, your two Task Manager screengrabs show a fairly smooth allocation of resources across your available cores.
I think we need more information before we can offer more specific advice
Martin
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Re: New audio interface - need a lower latency
lovesexy wrote:Here's a screenshot of the same project at 192 samples and 2048.
It's not like the CPU or RAM is being maxed at 192 but Cubase is suffering like a dog.
https:*?*?*/nyKYjH
https:*?*?*/eV1Prx
That's because real-time throughput is limited by the weakest link in the chain, not the strongest.
People focus on the CPU (and the smart ones on RAM) but it's like as with cars: if you have a V6 engine but your tires are bicycle size, replacing the engine with a more powerful V8 would not help much at all
Something in the architecture or configuration of your system is preventing it to make use of all the available grunt it has. It can be anything from the USB controller to the network controller to RAM speed to something as simple as a persistent power management agent or mouse driver (!).
The way to resolve this is to be quite systematic: disable everything you can disable (starting from the BIOS, stuff like C States and CPU throttling) and the in the Device Manager. You want to start with a bare-bone machine with nothing more enabled that the CPU, the chipset, the graphic card, the hard disk controllers, the keyboard and mouse and USB devices. Basically, the essentials. For all USB (and everywhere you can) disable device power management.
Then add hardware one piece at time and see what's the impact. Make sure you have no background software and keep an eye on the Task Manager for activation time of housekeeping tasks. Look at the Event Viewer for possible error message, very often the information is clearly there but people don't know where too look.
Good luck!
Last edited by CS70 on Tue Apr 24, 2018 9:13 pm, edited 1 time in total.
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Re: New audio interface - need a lower latency
Those are the buffer sizes the interface gives me; I only have these to chose from:
64, 96,128,192,256,384,512,768,1024,1536,2048
It's important to note that I don't take any sessions into other studios. I mostly work alone for TV and Film. All the companies I work with use 48khz at 24 bit - so I .
I generally have 2 reverbs and one delay and use sends. However, I often use nectar for lead vocals and that uses up a lot of power.
64, 96,128,192,256,384,512,768,1024,1536,2048
It's important to note that I don't take any sessions into other studios. I mostly work alone for TV and Film. All the companies I work with use 48khz at 24 bit - so I .
I generally have 2 reverbs and one delay and use sends. However, I often use nectar for lead vocals and that uses up a lot of power.
Re: New audio interface - need a lower latency
I have been down this road and over time, got the latencies down. In the past I have used MOTU, Liquid Saffire 56/RME. Currently I also have a UAD PCIe card installed (yes I see you don't want to buy into UAD) The UAD PCIe card doesn't seem to have had much impact on reducing latency where I have effects/dynamics plug ins
Using the MOTU/Liquid 56/RME I have not seen a great deal of difference between them regarding latency, on average I work at 64 samples, but drum software seems to whack that up to 128 samples
The one that caused most problems was NI Studio Drummer, it had my i7 creaking at the knees, with only a few other tracks running. Then I read that when Studio Drummer starts, it loads whatever kit with all the effects/dynamics turned on, and it was suggested turning them off - that reduced the latency by a country mile! It was more than apparent by consulting the internet that I wasn't the only one unhappy with NI SD and its whack on the CPU and stuttering performance.
I came across my copy of BFD1 the other day and thought I would re-install it, see how it compared against Steinberg Groove Agent 4, and NI SD, and to my surprise the BFD1 hardly touches the CPU/RAM, the performance figures (F12 in Cubase) hardly register, but as softwares become more complex it seems they place more and ore demands on the PC (no surprises there then) but some are over complex.
But turning the effects off in Studio Drummer helped a lot in my case.
SOme basic changes - My Computer/Properties/Advanced/Settings(under performance tab)/Visual Effects and I set it for 'Adjust for better performance)
Under the Advanced Tab I have read conflicting advice regarding the Programs vs Background Services debate.
I have read conflicting advice regarding the network interface. Some say turn off the NIC others say turn off the wireless, so do either/both and see which if any has any profound impact. Go to network services and see adapter properties. You can simply right click then enable/disable.
I put a shortcut to both the NIC and the Wireless adapters on the Desktop. In my case it was the NIC that had more impact on the latency.
I am on WIn10/64, earlier versions of WIndows allowed a setting called Hardware Profiles, where at boot time you could select from several user profiles, I used one where any non-music hardware was turned off - but anyway, Microsoft removed the utility
I downloaded a utility called 'Ultimate Windows Tweaker' this allows you to switch off a lot of the cludge that Windows loads at StartUp. I would imagine the same job can be done within WIn10 settings anyway, but some settings they seem to make accessible by way of a convoluted route such that they don't want you to turn the thing off.
I run disk clean up (including clean system files), along with System Ninja and cCleaner each week or so, and whilst using ccleaner check the StartUp, a lot can be disabled, if disabling results in a negative impact, you can simply enable it. I think in StartUp I have about 80% of the stuff disabled with no unwelcomed apparent effect at all.
Of course, if going about doing an overhaul, make drive images, always have an escape route - this is true 'forward planning' you can return to year zero if things go wrong
Within Cubase, some suggest Devices/Device SetUp/VST Audio System and Activate/DeActivate/Experiment with 'ASIO-Guard'
If you are using products like Kontakt (I use Kontakt 5 and Halion 5) have you got Cubase set such that Cubase handles the multi-processing as opposed to the VST?
And as other have suggested, load VSTi/EFX one by one or load a troublesome project, one that gulps up the CPU and working through the tracks see if any has a disproportionate impact, then go on the Google and look for issues specific to that VST as opposed to a more general 'my computer runs slow' query
Judging by your setup, I would thin you have plenty of horsepower and RAM.
I also replaced my HDD with SSDs, with a separate drive for OS/Apps and one for Data/Samples etc and although there might be a 'perceived' improvement in performance, I am convinced there is an improvement
Using the MOTU/Liquid 56/RME I have not seen a great deal of difference between them regarding latency, on average I work at 64 samples, but drum software seems to whack that up to 128 samples
The one that caused most problems was NI Studio Drummer, it had my i7 creaking at the knees, with only a few other tracks running. Then I read that when Studio Drummer starts, it loads whatever kit with all the effects/dynamics turned on, and it was suggested turning them off - that reduced the latency by a country mile! It was more than apparent by consulting the internet that I wasn't the only one unhappy with NI SD and its whack on the CPU and stuttering performance.
I came across my copy of BFD1 the other day and thought I would re-install it, see how it compared against Steinberg Groove Agent 4, and NI SD, and to my surprise the BFD1 hardly touches the CPU/RAM, the performance figures (F12 in Cubase) hardly register, but as softwares become more complex it seems they place more and ore demands on the PC (no surprises there then) but some are over complex.
But turning the effects off in Studio Drummer helped a lot in my case.
SOme basic changes - My Computer/Properties/Advanced/Settings(under performance tab)/Visual Effects and I set it for 'Adjust for better performance)
Under the Advanced Tab I have read conflicting advice regarding the Programs vs Background Services debate.
I have read conflicting advice regarding the network interface. Some say turn off the NIC others say turn off the wireless, so do either/both and see which if any has any profound impact. Go to network services and see adapter properties. You can simply right click then enable/disable.
I put a shortcut to both the NIC and the Wireless adapters on the Desktop. In my case it was the NIC that had more impact on the latency.
I am on WIn10/64, earlier versions of WIndows allowed a setting called Hardware Profiles, where at boot time you could select from several user profiles, I used one where any non-music hardware was turned off - but anyway, Microsoft removed the utility
I downloaded a utility called 'Ultimate Windows Tweaker' this allows you to switch off a lot of the cludge that Windows loads at StartUp. I would imagine the same job can be done within WIn10 settings anyway, but some settings they seem to make accessible by way of a convoluted route such that they don't want you to turn the thing off.
I run disk clean up (including clean system files), along with System Ninja and cCleaner each week or so, and whilst using ccleaner check the StartUp, a lot can be disabled, if disabling results in a negative impact, you can simply enable it. I think in StartUp I have about 80% of the stuff disabled with no unwelcomed apparent effect at all.
Of course, if going about doing an overhaul, make drive images, always have an escape route - this is true 'forward planning' you can return to year zero if things go wrong
Within Cubase, some suggest Devices/Device SetUp/VST Audio System and Activate/DeActivate/Experiment with 'ASIO-Guard'
If you are using products like Kontakt (I use Kontakt 5 and Halion 5) have you got Cubase set such that Cubase handles the multi-processing as opposed to the VST?
And as other have suggested, load VSTi/EFX one by one or load a troublesome project, one that gulps up the CPU and working through the tracks see if any has a disproportionate impact, then go on the Google and look for issues specific to that VST as opposed to a more general 'my computer runs slow' query
Judging by your setup, I would thin you have plenty of horsepower and RAM.
I also replaced my HDD with SSDs, with a separate drive for OS/Apps and one for Data/Samples etc and although there might be a 'perceived' improvement in performance, I am convinced there is an improvement
Last edited by OneWorld on Wed Apr 25, 2018 6:04 pm, edited 3 times in total.
Re: New audio interface - need a lower latency
You are not using some really high sample rate on your projects are you? That will send your CPU to the edge really quickly with heavy plugins.
Also its probably not a popular thing to say around here but Asio runs much smoother on Win8 & 10 compared to Win7. Not only that but WASAPI is even a usable option now, they rebuild the audio stack in Win10.
https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/windows-threshold--what-s-new-for-audio
Its one of the rare good things about Win10
Maybe give Win10 a try. Take an image of your computer so you can go back if you have issues.
It might not completely solve your problem but I would expect it to be better. You gotta think Steinberg will not be developing their software to run optimized for Win7 anymore, they will be developing for Win10. You are not getting the best out of that processor running Win7.
MS have pulled back on the spying now, Win10 is much less annoying than it was when it first came out. And they don't force updates, you can defer them for months.
Also its probably not a popular thing to say around here but Asio runs much smoother on Win8 & 10 compared to Win7. Not only that but WASAPI is even a usable option now, they rebuild the audio stack in Win10.
https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/windows-threshold--what-s-new-for-audio
Its one of the rare good things about Win10
Maybe give Win10 a try. Take an image of your computer so you can go back if you have issues.
It might not completely solve your problem but I would expect it to be better. You gotta think Steinberg will not be developing their software to run optimized for Win7 anymore, they will be developing for Win10. You are not getting the best out of that processor running Win7.
MS have pulled back on the spying now, Win10 is much less annoying than it was when it first came out. And they don't force updates, you can defer them for months.
Last edited by craigtumps on Thu Apr 26, 2018 9:43 am, edited 2 times in total.
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- craigtumps
Poster - Posts: 53 Joined: Tue Jan 24, 2017 5:49 pm
Re: New audio interface - need a lower latency
craigtumps wrote:You are not using some really high sample rate on your projects are you? That will send your CPU to the edge really quickly with heavy plugins.
Also its probably not a popular thing to say around here but Asio runs much smoother on Win8 & 10 compared to Win7. Not only that but WASAPI is even a usable option now, they rebuild the audio stack in Win10.
https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/windows-threshold--what-s-new-for-audio
Its one of the rare good things about Win10MS have pulled back on the spying now, Win10 is much less annoying than it was when it first came out. They don't force updates, you can defer them for months.
Maybe give Win10 a try. Take an image of your computer so you can go back if you have issues.
you gotta think Steinberg will not be developing their software to run optimized for Win7 anymore, they will be developing for Win10. You are not getting the best out of that processor running Win7.
+1 for that, I was a very reluctant WinTenner, but the various updates seemed to have trimmed off a lot of the fat and I too feel ASIO runs better on Win10 than it did with Win7
Re: New audio interface - need a lower latency
Re "trying W10". I did this and when I went back to W7 things were NOT the same!
Very likely my fault but since hard drives are cheap I suggest you load W10 on a fresh one (physically remove the W7 drive?) to give it a go.
One of the few good things about dongle protection is that you can have Cubase on as many machines as you like and no arguments about 're-activation'.
Dave.
Very likely my fault but since hard drives are cheap I suggest you load W10 on a fresh one (physically remove the W7 drive?) to give it a go.
One of the few good things about dongle protection is that you can have Cubase on as many machines as you like and no arguments about 're-activation'.
Dave.
Re: New audio interface - need a lower latency
yeah I was a dedicated Win10 hater when it first came out.
But they changed some of their annoying practices and the latest builds won me over. I wouldn't go back to Win7 now, no chance.
One really important thing to add. I hope you have an SSD drive as your system drive? Because Win10 runs quite badly without one.
But they changed some of their annoying practices and the latest builds won me over. I wouldn't go back to Win7 now, no chance.
One really important thing to add. I hope you have an SSD drive as your system drive? Because Win10 runs quite badly without one.
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- craigtumps
Poster - Posts: 53 Joined: Tue Jan 24, 2017 5:49 pm