Latency on vst instruments in cubase
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- KVRian
- 833 posts since 18 Jun, 2013
I am trying to use VST instruments on 256 samples buffer size. Sometimes I feel latency while recording VST instruments. What causes this latency?
Below is my computer set up:
AMD Ryzen 9 9950X 4.3 GHz 16-Core Processor
ASRock Phantom Gaming X870E Nova WiFi ATX AM5 Motherboard
Corsair Vengeance 96 GB (2 x 48 GB) DDR5-6000 CL30 Memory
Western Digital WD_Black SN850X 8 TB M.2-2280 PCIe 4.0 X4 NVME Solid State Drive
OS: Windows 11
Audio Interface: Audient ID4
Midi controller: Korg x50 connected to USB-c on computer direct through a printer cable.
Audio Interface: Audient ID4 MKII
Buffer size: 256
Total latency shown on cubase: 15 ms
Is this high? If so, what causes it and how to reduce it?
Below is my computer set up:
AMD Ryzen 9 9950X 4.3 GHz 16-Core Processor
ASRock Phantom Gaming X870E Nova WiFi ATX AM5 Motherboard
Corsair Vengeance 96 GB (2 x 48 GB) DDR5-6000 CL30 Memory
Western Digital WD_Black SN850X 8 TB M.2-2280 PCIe 4.0 X4 NVME Solid State Drive
OS: Windows 11
Audio Interface: Audient ID4
Midi controller: Korg x50 connected to USB-c on computer direct through a printer cable.
Audio Interface: Audient ID4 MKII
Buffer size: 256
Total latency shown on cubase: 15 ms
Is this high? If so, what causes it and how to reduce it?
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- KVRAF
- 5271 posts since 2 Jul, 2005
All the operating systems that cubase runs on are far from real-time. Meaning that you'll be using a buffer for all audio in and out. If you have the hardware you can get this buffer size (cause of latency) down to extremely low levels. 256 samples @ 48khz = 5.333ms. The software will add a bit extra depending on how it works. This is a relatively small latency, but you could get your latency down to 64 samples with a better soundcard. Generally there's a trade-off between lower latency and better performance so as you add more and more to a project you'll have to add more latency to free up system resources. Asio guard does some of this automatically by running two different buffers, one for "live" tracks and one for "playback". The extremely short version of this answer is "there's always gonna be latency". I'd ignore it if it doesn't stick out to you while you work. If it does, then adjusting system settings or getting a better sound card are your only real options. USB on windows can be very iffy for low latency performance so I've mostly stuck with actual "card" based soundcards that bypass the USB system altogether.
Don't F**K with Mr. Zero.
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- KVRAF
- 7825 posts since 20 Jan, 2008
I concur with the above statement. I haven't had a single issue with latency in the year plus since I switched to my current system. I think a great deal of that has to do with soundcard branding With slight exception the more you invest into the sound card the better your timing is.
I stopped using the internal (realtech) card many many years back along with the free asio products. The last ten or so years it's been a Yamaha THR10 Line 6 Pod Go and Zoom L8. The first two while somewhat limited are both Yamaha which is the parent company of Steinberg. Making sure you have a licensed ASIO product rather than a generic ASIO can make a huge difference. The biggest thing to me was the Zoom though I hate the UI l love the fact I can burn audio directly to flash in the unit then transfer over and bounce back.
I don't need that many channels of live audio in. The Zoom is great but as stated I hate the UI. Often think of moving over to a Boss Gigcaster 8 simply because the of the UI
I stopped using the internal (realtech) card many many years back along with the free asio products. The last ten or so years it's been a Yamaha THR10 Line 6 Pod Go and Zoom L8. The first two while somewhat limited are both Yamaha which is the parent company of Steinberg. Making sure you have a licensed ASIO product rather than a generic ASIO can make a huge difference. The biggest thing to me was the Zoom though I hate the UI l love the fact I can burn audio directly to flash in the unit then transfer over and bounce back.
I don't need that many channels of live audio in. The Zoom is great but as stated I hate the UI. Often think of moving over to a Boss Gigcaster 8 simply because the of the UI
Dell Vostro i9 64GB Ram Windows 11 Pro, Cubase, Bitwig, Mixcraft Guitar Pod Go, Linntrument Nektar P1, Novation Launchpad
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- KVRist
- 433 posts since 23 Sep, 2008
Activate the ASIO Latency Compensation button on the track you're recording MIDI to.
What sound do dreams make when they die?
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- KVRian
- Topic Starter
- 833 posts since 18 Jun, 2013
What is ASIO compensation and does it help latency while recording?
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- KVRian
- Topic Starter
- 833 posts since 18 Jun, 2013
Is this an audio interface issue or is it my computer or is it my midi controller (korg x50) which is causing the problem?
- KVRist
- 351 posts since 24 Aug, 2017
Cant make out how you got to 15ms, but you can basically reduce the buffer at the expense of CPU and/or increase the sample rate at the expense of storage and cpu.
Direct monitoring should be engaged when recording through analog inputs which gives you zero latency. You can activate a cubase function called restrain latency compensation when recording virtual instruments, which basically disables all latency incurring plugins when activated.
You can also increase the buffer at the mix stage which will reduce cpu en enable you to run more plugins,
Anything below 10ms is unnoticeable for most people, and anything below 5 would be almost impossible with your gear. You would have to really streamline your rig to disable anything that is causing a lot of real time spikes (WiFi, GPU etc).
Don’t worry about latency unless you experience an issue.
Direct monitoring should be engaged when recording through analog inputs which gives you zero latency. You can activate a cubase function called restrain latency compensation when recording virtual instruments, which basically disables all latency incurring plugins when activated.
You can also increase the buffer at the mix stage which will reduce cpu en enable you to run more plugins,
Anything below 10ms is unnoticeable for most people, and anything below 5 would be almost impossible with your gear. You would have to really streamline your rig to disable anything that is causing a lot of real time spikes (WiFi, GPU etc).
Don’t worry about latency unless you experience an issue.
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- KVRian
- Topic Starter
- 833 posts since 18 Jun, 2013
I am experiencing a lag while recording and I mostly use orchestral vsts for my film scoring work.
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- KVRian
- Topic Starter
- 833 posts since 18 Jun, 2013
I do but not in the chain I'm recording.. I generally add it as sends post recording everythingAh_Dziz wrote: Wed Jun 18, 2025 9:54 pm Are you using any convolution reverb? In your orchestral libraries or in the DAW? That generally adds latency. Midi also has it's own latency.
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- KVRAF
- 1767 posts since 20 Feb, 2003
Firstly, many plugins themselves have latency, in addition to your cards latency. In Cubase you can find out the latency for each plugin by going to Studio - VST Plug-in Manager, selecting a plugin, and it will list its latency in the Plug-in info panel at the bottom. Whilst many are zero, or relatively low, some can be very high, and these figures can add up, particularly if you use FX chains with several plugins.
Secondly, a MIDI input device has latency. It has to scan every keyboard key to see if you’re pressing it, then it has to convert that to a signal and send it over MIDI or USB. This time varies by device (and there’s no good information available about this topic AFAIK.) Count on it adding at least a few milliseconds (meaning ~5ms) with the number being higher for older digital devices. EG I recall the original Korg Wavestation keyboard being said to require about 10ms. How the system processes this timing information also matters (aka jitter etc).
Thirdly, there’s the time for sound to travel in air, which is (roughly - things like temperature and humidity change the speed) 1.1ms per 12 inches. So, if you’re 7ft from any speakers that’s adding nearly 8ms of latency.
On a sound card, I’ve found the magic figure to stay below, for real-time VST instrument usage, is 10ms. The perception, of any delay, varies by individual. EG a latency change between 3 and 8ms will go unnoticed in typical instrument usage. But change that from 10 to 15ms and, even though the increased amount is the same, it will cross a threshold where some people definitely notice a delay. Generally, almost everyone will notice once it gets to 30ms, which is surprisingly low if you think about it (1/33rd of a second).
15ms seems high for a 256 buffer, and I assume what you're actually quoting is the round-trip figure (in and back out - Edit: Your screenshot confirms this). Unless it's acting on an audio input an instrument only accounts for the "out" portion of that. So its likely your "actual" source of latency is an effect plugin, as some can be very high - to the point of making things unusable for real-time playing.
Secondly, a MIDI input device has latency. It has to scan every keyboard key to see if you’re pressing it, then it has to convert that to a signal and send it over MIDI or USB. This time varies by device (and there’s no good information available about this topic AFAIK.) Count on it adding at least a few milliseconds (meaning ~5ms) with the number being higher for older digital devices. EG I recall the original Korg Wavestation keyboard being said to require about 10ms. How the system processes this timing information also matters (aka jitter etc).
Thirdly, there’s the time for sound to travel in air, which is (roughly - things like temperature and humidity change the speed) 1.1ms per 12 inches. So, if you’re 7ft from any speakers that’s adding nearly 8ms of latency.
On a sound card, I’ve found the magic figure to stay below, for real-time VST instrument usage, is 10ms. The perception, of any delay, varies by individual. EG a latency change between 3 and 8ms will go unnoticed in typical instrument usage. But change that from 10 to 15ms and, even though the increased amount is the same, it will cross a threshold where some people definitely notice a delay. Generally, almost everyone will notice once it gets to 30ms, which is surprisingly low if you think about it (1/33rd of a second).
15ms seems high for a 256 buffer, and I assume what you're actually quoting is the round-trip figure (in and back out - Edit: Your screenshot confirms this). Unless it's acting on an audio input an instrument only accounts for the "out" portion of that. So its likely your "actual" source of latency is an effect plugin, as some can be very high - to the point of making things unusable for real-time playing.
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- KVRian
- Topic Starter
- 833 posts since 18 Jun, 2013
When I enabled CDC on my cubase, the latency is workable. It isn't much. Does it mean a plugin in my chain is the problem?
I have few send fx but don't use them.
On my chain, I only have a PRO-L and Sound ID reference plugin.
Latency value for Sound Id reference plugin: 7
Latency value for PRO-L: 3141
I have few send fx but don't use them.
On my chain, I only have a PRO-L and Sound ID reference plugin.
Latency value for Sound Id reference plugin: 7
Latency value for PRO-L: 3141
- KVRAF
- 16797 posts since 8 Mar, 2005 from Utrecht, Holland
You should remove FabFilter Pro-L from the chain while recording. When it "looks ahead" to gracefully limit a peak coming up in the future, that's where latency is added.
FabFilter wrote:Note that some algorithms (especially the last four new algorithms introduced in Pro-L version 2) need more latency than others. Do you need to minimize processing latency? Then choose one of the first four algorithms (e.g. Transparent) and disable True Peak Limiting and oversampling.
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- KVRist
- 42 posts since 7 Apr, 2023
Why buffer size so high? You have a powerful cpu. I have Amd 5900x and my buffer size is 64. Cubase/Reaper/Bitwig. Audio interface is Audient mk2 14. No issues.
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- KVRian
- Topic Starter
- 833 posts since 18 Jun, 2013
Is there an other low latency limiter I can use?BertKoor wrote: Wed Jun 25, 2025 6:40 am You should remove FabFilter Pro-L from the chain while recording. When it "looks ahead" to gracefully limit a peak coming up in the future, that's where latency is added.FabFilter wrote:Note that some algorithms (especially the last four new algorithms introduced in Pro-L version 2) need more latency than others. Do you need to minimize processing latency? Then choose one of the first four algorithms (e.g. Transparent) and disable True Peak Limiting and oversampling.
I use Yamaha hs5s and the speaker volume isn't much and soundid reference also does a -6db for the volume it increases to compensate frequencies