What are CPU "C-states" and how to disable them if needed?
https://gist.github.com/wmealing/2dd2b543c4d3cff6cab7

 

I cannot recommend LatencyMon enough. Should be a must when dealing with audio issues.

 

Friendly suggestion: call someone [a pro] who is experienced in making music with a DAW *and* computer hardware to check your computer, and to show you how you can make your projects without too much frustrations. There is so much about it, what can be wrong with your computer, OS, or your workflow, that I could write a book about it. 200-300 pages minimum.

Have you disabled Speedstep, C-states, all CPU power management, and unnecessary devices in the BIOS? Put your power mode in OS to *highest performance*? Turn on background tasks priority in advanced system settings? Make page file 1024MB? Turn off disk file access time checking, Superfetch and Prefetch? Turn off AV, OS, and other unnecessary software in the systray updates? Stop unnecessary software from running at startup and in the systray [run msconfig and go to "startup" tab]? These are just some of the 100 things you can do to help your DAW work smoother. Turning down unnecessary polyphony in VSTis helps greatly. Bass, lead, most drums can be monophonic. Unison mode doesn't help if you're not pitch-shifting the voices etc. ASIO buffer size 128-256 samples while tracking, 256-512 when mixing and mastering. These are just a few suggestions of the top of my head.

Use 1-2 reverbs and possibly even delay as sends on a separate track, not as inserts on every channel, or insert them on buses. Use bus compression instead of track compression etc. While at it, I could give you this helpful link: https://www.soundonsound.com/mixing-production You can learn a lot there. They also have a very informative Youtube channel. SonicState is also good, SpectreSoundStudios, too.

Good luck!

 

Last edited: May 16, 2021

192 buffer size? Never heard of such value. Are you sure that is not a samplerate value? In any case, if you are not able to change buffer size (and samplerate, if 192 is the samplerate is very high indeed and you will hear clicks easily with a small buffer size, especially on Ableton) it means that you have to change it before on your external audio card dedicated software or you don't have an external audio card with asio at all, do you have an external asio audio card? Btw from my experience i can say than even with my new pc Ableton makes some clicks with some more plugins, it seems that the more they update it the heavier on cpu it gets, if i use Reaper i can load twice the plugins with the same buffer size and samplerate, no clicks at all, nowadays if you wanna use a ton of plugins on Ableton you need a very big PC and even with that you probably end up resampling /freezing or set a higher buffer sooner or later anyway, remember that lower buffer size are needed mainly while recording, tracking, playing vst instruments and doing editing to avoid latency, when you move over to the mix you can increase buffer size and clicks will be gone for good, if you don't have tons of kontakt instances opened of course, once you finished with your midi tracks you will render/resample/freeze anyway, clicks also depends on the samplerate you set up, the higher samplerate is the more your pc will struggle with small buffer size selected, but again, 192 seems more a samplerate value rather than a buffer size value to me, never heard of it, common buffer size values are 48, 64, 96, 128, 256, 512, 1024, 2048.

 

Last edited: May 16, 2021

If you enable the "High Performance" power profile in Windows, it also won't let your CPU underclock. It will keep your CPU at max clock speed.
I have two shortcuts on my taskbar that let me switch between the "High Performance" and "Balanced" power profiles. I prefer using this solution over disabling C-states altogether.

 

Oh, 2 selectable shortcuts - a pretty smart solution.

 

That can only mean that you MUST SET your Latency while live/drivers are off!
Turn Live or any daw off, then change the latency in the driver settings.
Turn on Daw done

If that doesnt work get yourself some DSP/AudioGridder or adjust your workflow

 

Well, it all looks like settings on the PC. I would start in the BIOS and turn off everything that we don't need. These are virtualization, Turbo Boost, if available, and all power-saving functions. Then in the OS itself energy plan maximum performance.

And look through all the entries again under advanced settings and switch off all energy-saving functions. Likewise in the Device Manager, especially with the USB hubs. What about simultaneous processes?

Which software is running at the same time or do you have guitar rig open at the same time? Set the buffer size higher for the Asio Driver "ASIO Control Panel" - not too high so that the latency is not uncomfortable - not too low so that it does not crack.

 

sync clock speed of all cores is also worth a try

 

First thing you should do: shut down your browser and all other applications you don't need while making music. I often got CPU spikes when my PC is online with the browser opened in the background. You can also try deactivating your LAN-device in the device manager, since those often cause glitches.

And of course you need an ASIO-capable audio interface, ideally one that has good driver performance such as the RME interfaces!
https://gearspace.com/board/music-c...erface-low-latency-performance-data-base.html

Optimize PC for audio production:
https://support.focusrite.com/hc/en-gb/articles/207355205-Optimising-your-PC-for-Audio-on-Windows-10

 

Last edited: May 16, 2021

I bought a cheaper Raedon graphics card because I thought it had nothing to do with performance. I also dont think its a total piece of sh** though. Yes in some of the projects with more instances of kontakt its glitching out more. If i just am doing something with synths its way less likely to be messed up. I just dont understand why some days its manageable to deal with and other days I cant work on anything practically. I run task manager to look to see whats eating CPU and its all background processes running at very low cpu usage. (I have a komplete audio 6 interface.) What interface you have can affect the cpu performance?

 

My PC is already overclocked. Disabling C-states?

 

I have a raedon graphics card it wasnt the best or the worst. This effects performance? I have a komplete audio 6. Your interface effects performance?

 

Hi Nick Bellagio look here:
How to enable and disable C-States on Intel platforms
https://itigic.com/how-to-disable-c-states-on-intel-platforms/

Have you already called up your "Control Panel"?
The KOMPLETE AUDIO 6 driver controls communication between the audio interface and your computer.
• The control panel software is used to set various parameters related to performance, synchronization and preamplification in Windows.

Spoiler: Now you can select the Komplete Audio ASIO Driver in your DAW.

Now you can select the Komplete Audio ASIO Driver in your DAW.

Spoiler: Turbo Mode

Turbo Mode
On some laptops and systems the Turbo Mode of the CPU causes glitches and instability when running low latency audio. Turbo Mode can be turned off and you may find running your system at the stock CPU speed gives you a glitch-free and more consistently reliable playback. If you are having problems then this is something you should try.

Go to the Advanced Power Settings and go down to where you set the Processor Power Management. Set the Minimum and Maximum power to 99% – this disables Turbo Mode. See if that sorts out your problem.

Spoiler: Call the Task Manager on which shows you the GPU

Call the Task Manager on which shows you the GPU (Graphical Processor Unit), the graphics card relieves your PC considerably.

Spoiler: Use ASIO Drivers

Spoiler: Latency settings

Latency settings

The smaller the audio buffer is set, the more the computer has to do. If your computer is not able to complete all the work steps (see above) within the set buffer, audible acoustic disturbances such as clicks and dropouts in the playback can occur. Accordingly, you have to increase the audio buffer size so that the computer can cope with the resulting processes within the allotted time. Start with a buffer size of 512 samples and test whether interference-free playback is possible with this setting.

If not, increase the buffer size in small steps and test again. On some modern multicore CPUs, it is possible that you will achieve better results with low buffer settings than with high settings (e.g. cracks may occur with a buffer size of 512 samples, but with a setting of 256 samples Not). You should therefore definitely test the buffer settings of 256 and 512 samples. Please also research the manual for your audio interface to find out more about its driver and the recommended latency settings.

Spoiler: Native Instruments - Windows optimization tips for audio processing

Native Instruments - Windows optimization tips for audio processing
https://support.native-instruments....-Optimierungs-Tipps-für-die-Audioverarbeitung
Use the " Google Translator" https://translate.google.com/?hl=de&tab=TT

Spoiler: Optimizing the audio settings for KOMPLETE AUDIO 1/2/6

Optimizing the audio settings for KOMPLETE AUDIO 1/2/6
https://support.native-instruments....-Audio-Einstellungen-für-KOMPLETE-AUDIO-1-2-6

 

Last edited: May 16, 2021

@Nick Bellagio

If latency and network traffic is not the issue...

While pops and clicks can be decreased on exporting/rendering audio and increasing the buffer size to much higher fixes some things, this sounds like something else. This is usually done for realtime exporting when you have used a load of plugins and are getting a CPU overload, or a disk overload where you do this plus increase the space prior to the first note by around 5-8 seconds so the disk can start caching.

Windows 10 Fiber Local Storage (FLS) basically removed the amount of plugins a DAW could load. So theoretically, in Win 10, your computer would never blue screen or crash by a person going crazy loading 10 inserts per track on a 128 track mix (theoretically only - nobody sane would do that).
There's an article here https://devblogs.microsoft.com/wind...-the-fls-slot-allocation-limit-in-windows-10/

Unfortunately, what nobody said when they implemented this is that two REALLY important things also have to be taken into consideration. There are three in your case using Kontakt. The first is the amount of RAM and the second is the virtual disk. While increasing the paging file sizes in Windows can help it won't fix the problem in your case. RAM however works in tandem with the paging file. This is where unused data not presently in memory is theoretically placed and cached as needed(loaded but not used at that point in time). The third is the problem by the sound of it being too many Kontakt instruments - If you load a 100GB library, then a 5, then a 20, then a 40, then a 2.... do you have 1TB of RAM? - Nobody does - it caches to and from the disk after loading. Someone said freezing the audio. This works.

You might need to consider changing some Kontakt settings while auditioning and writing if you intend to use many instruments (e.g. ten or more) and turn it back on when exporting. Increasing the buffer size as suggested earlier will facilitate the voices not dropping or pops and clicks on exporting. There is a workaround (simple fix below) - I know this works. I upgraded my RAM and it fixed most issues.


The simplest solution:
I am confident the sheer amount of data size is the problem if it is not the audio module being interfered with by latency or network traffic (try turning network off while recording) too).
1. Save your tune/song/composition as a version #2 (you have the original always then).
2. When you have an instrument you like and are happy, solo it, export it as a high quality wav file and delete the instrument. Do this with what you are happy with until you are working with more audio files than you are with vsti or vst plugins. If you want to go back to the original, you've already saved it at #1.

As a personal choice, when I get to mix, the only vst's I want to see are audio inserts (e.g. EQ, compressor etc etc).

P.S - Microsoft didn't tell the whole truth with FLS either. There is a ceiling - read the above enclosed article link.

 

Last edited by a moderator: May 16, 2021

I've yet to see a single computer that had superfetch enabled and wasn't running at 100% Disk Use constantly.

 

Damn is this something most of you generally do when you produce? I have never really messed with the buffer size after getting it down to something manageable.

But now I'm realizing that maybe some of you are using that buffer fader like a rubber band on various tracks. I think I'm about to try doing this too and see if that helps.

It would make sense since some tracks seem to do fine and others seem to just have too much load per project, which heavily has to do with how many plugins you run simultaneously and how much ram you have to manage them.

But still each project does seem to demand a different load too. Buffer size never dawned on me.

 

Changing the buffer size? Sure. As less plugins and smallest buffer size as possible while recording and a bigger one for mixing and mastering. Common approach.

 

Crazy, I've always had it at a fixed value after setting it up as best as I could. But all that talk about C-states was doing my head in when I just searched it.

 

To start. Not knowing exactly the what & how of your system, I can only recommend some areas of opportunity to address.
The audio device should be connected directly to the PC (not through a USB hub). Some boards USB ports are actually hubs.
Also use a quality (not expensive) USB cable capable of full bi-directional speeds.
----------------------------------------------------------------------------------------------------
From the Taskbar, Right-Click the Volume icon and select Open Volume Mixer.
Double-Click System Sounds.
On the Playback tab, is your Audio Interface set as the Default Device?
How is your Windows System Sounds setup?
Have you disabled devices for the built in audio (playback & recording) if not being used?
On the Enhancements tab, is Disable all enhancements checked?
On the Advanced tab, is the Default Format set to the highest value for your audio device?
Example: 24 bit, 192000 Hz, (Studio Quality)
Also on the same tab under Exclusive Mode, are both boxes checked?
After making any changes it is good practice to Restart your system. This ensures the new
settings are committed & buffers are cleared.
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You mentioned gaming systems for audio. Not all motherboards give the same performance. This is especially true for prebuilt systems where the goal is cost savings on components to maximize profit (not performance). Consumer grade boards fall short compared to Workstation (Creator) grade boards in most cases.

Professional prebuilt Workstations for Video Production usually work well for Audio Production at the entry to Mid level pricing.

You also mentioned that your system is over-clocked. In general this is not good for audio (a separate major topic - latency, instability).

Most gaming boards push up to or, past the Intel specs in the BIOS settings.
Memory should only be over-clocked using the XMP Profile settings letting the board train to the installed RAM.
The RAM should also not be mixed types and, should be on the boards QVL (qualified vendors list).
Modules not on the list may or may not work/perform correctly.

Just some random thoughts that hopefully bare fruit.

Happy tracking!