How To Reduce Download Latency

If you want to measure latency that is specifically a result of FMOD, then recording a profiler session via live update, observing when a specific API command is processed in the API calls view, and comparing it to when the audible effect of that command occurs is a better option.

Audio interfaces always will have the direct monitoring feature. This is a feature that allows you to monitor your input source directly before it passes through the A/D convertors so therefore the sound is direct analog and has zero latency. This is mixed in your monitors/ headphones with the playback from your computer via the ASIO audio / USB / D/A convertors which your DAW will automatically adjust for latency so that your overdubbed tracks are perfectly in snyc with the already recorded material in your DAW.

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You can also have issues with latency if you try and overdub tracks while listening the the sound of the output of the DAW instead of the input. The output of any DAW will have some amount of latency depending on the quality of the system.

For piano, I find a buffer size of 64 at 44.1K is the maximum I can tolerate with my Focusrite Scarlett (4.6 msec latency, 9.3msec roundtrip). Any higher, and I notice the latency.

Increasing the sampling rate to 96K will more than half these figures, giving a input/output latency of 2.2 msec (4.4 msec roundtrip)... but this also increases the CPU usage.

For a CPU intensive project, that might mean temporarily bypassing FX or freezing tracks to avoid drop-outs.

A more extreme version of this would be to mix down the whole project to a stereo track, and temporarily archive all the other tracks. In some ways this can be more convenient as there's very little "thought" about it when switching - i.e. you select all, unselect the mixdown track, and archive everything else.

Another alternative is to use an external MIDI sound module. I used to use my old Yamaha MU10 for this purpose, however depending on the part I was recording, the differences in response (e.g. velocity / feel) may or may not work as well when the part is played on a different piano sound.

Which method I choose depends largely on where I am in the project. In the early stages, I'll use FX bypassing or freezing. The mixdown/archive approach is more useful for last minute overdubs.

In all honesty though, I rarely need to any of this as I tend to treat tracking / mixing / mastering as separate tasks - i.e. I avoid trying to mix as I go along, and only use effects that contribute to the sound during the tracking stage.

To do this: Go to preferences /hit "P" on the keyboard. Under the Sync and Caching TAB you will find the default size of 256KB by Cakewalk. Set your ASIO driver accordingly and find the setting that works best for you.

The "nuclear option" is a Thunderbolt interface or a PCIe card interface (e.g., RME, ESI), either of which can have lower latency than USB. For USB, MOTU claims 2.5 ms round-trip latency for their M2 interface at 96 kHz with 32 sample buffers.

My bad. I wrongly assumed that in order to change the buffer size you needed to enable caching. Clearly you can change the buffer size without that.

I noticed that playback was double (512 KB) the size of the record (256 KB) buffer size. Is that standard for CbB? Do you know why making them both the same size has a positive effect on latency or is it just by trial and error that you found it to be better?

My bad. I wrongly assumed that in order to change the buffer size you needed to enable caching. Clearly you can change the buffer size without that.

If the buffer size is too low, you may encounter errors during playback or may hear clicks and pops. If the buffer size is set too high while recording, there will be quite a bit of latency which can be irritable.

Cakewalk sets both Playback and Record I/O to 256KB by default, so set it back to 256KB. There might be a clock sync issue with your system? Try to reset your configurations in preferences and update your drivers.

Your ASIO drivers needs to be sync with that of your hardware - that's why it is important to update to the latest Asio drivers, I think it's v4.65 for focusrite -- nowhere near the studio right now to check.

If you use direct monitoring it doesn't matter what buffer size etc you use. You will hear your input synced perfectly with the output. DO NOT TURN ON INPUT ECHO on new audio tracks your about to record. Use direct monitoring.

RTL only matters if you are trying to listen to what you are inputting AFTER it has passed through your system. There will always be latency if you do this. And ya,, with a top notch system and interface you can get that way down to something like 2.5 ms but there will always be latency. Most run of the mill systems ( Computers/ audio interfaces) run at more like 8ms to as high as 30 ms.

I find it's important to use a good midi driver that comes with most of the better controllers. I had terrible latency all of sudden a while ago and it turned out the issue was caused by W10 update overwriting my Roland midi driver with a generic midi driver. I re installed the Roland midi driver and the latency was gone.

The other times I will experience midi latency is when I have certain effects like the LP multi band active. The latency goes away the minute I bypass all my effects. So I'm in the habit of making sure I by pass all my effects before I try and overdub midi parts with my midi keyboard or my midi drums.

As far as I can tell midi latency has little to do with audio settings. There is for sure still some latency which will be caused by the system. I can hear just a bit of echo if I listen to my midi drums brain mixed with a VST drum output. So once again I use direct monitoring when playing drums and only monitor the brain while tracking. On a top notch system it might not be as noticeable but on mine it is. Someday I'll purchase a better interface like RME but for now I get by fine with a Focusrite or my Tascam.

Just like buffer size "Direct Monitor" also has an affect on certain machines with latency. I had the problem with a laptop I was using on the road. It was an i5 4th gen if i'm correct - if not older, with a standard 500GB HDD, 8GB ram with Windows 10 (Can't remember which build,) My Solo gave me latency with direct monitoring on.

Yes, you're spot on on few things you've mentioned, but I think it was time wasted. Until we know the specs of the system and hardware used by Konskoo, there's no way to claim an answer. Right now, you have to assume that he is using "Asio4all." Different setups, totally different problems and approach on latency.

Before the interface and a upgrade on the laptop (with a group that hired me everytime the went on tour,) I ran a successful on-the-road setup for demo projects, with only a USB midi, USB Mic, with the laptops onboard recording interface to record guitars, a pair of Alesis 3 monitors; using Cakewalk Sonar LE and only ASIO4ALL, with the buffer size set to 1024msec and Playback buffer at 256KB and guess what? NO LATENCY.

(So you think that both Playback and Record I/O to 256KB is better?)

Later on I found out that it was "thread scheduling" value 3 that gave problems. This however seems to have been fixed in the recent CbB releases.

Using a simple project (some EQ, compression and two normal reverbs + one convolution) and a few audio tracks + many inactive tracks and inactive busses with preloaded FX from my template) I measured it today against against "thread scheduling" 2 which now shows the following during playing:

thread scheduling 3: Audio Processing 6% (Max. 16%, jumps to 21% during pause and jumps to 16 again during play???), 35% (Max. 55%, jumps to 103 during a pause???), Engine Load, 0 Late Buffers

thread scheduling 2: Audio Processing 7,5% (Max. 17%, jumps to 22% during pause) Audio Processing, 38% (Max. 54% jumps to 110 during pause), Engine Load, 0 Late Buffers

(numbers are rounded averages since they change continuously).

So at first sight some small performance improvements. Strangely enough when I stop the project the Max. values jump up and when hitting play again the Max. values drop significantly?

I've got the latest RME driver for my AOI PCIe card, so that shouldn't be a problem. I make sure to always have the latest driver and software updates for anything on the PC (has become a bit of an obsession?).

Things go reasonably well right now. I expected my 16-core Threadripper 1950X with 32 GB RAM and fast PCIe SSD drives to run smooth all the time, even under heavy workload. Still, if you don't pay attention to the many different Windows settings (sleep timers, GPU audio, power settings, wireless connections etc.) and other background processes, such a system can be disappointing. After tweaking it at many sides it's finally becoming nice to work with. Because of the many uncertainties introduced by other software I now use a program (ProcessKO) to disable any background process that is not needed for the system when I run CbB. This also seems to help a bit.

System is now running 96kHz, 24bit + 64bit double precision engine (I wonder how useful the latter is and if it uses more resources) with latency of 2,7 ms at 256 (both Playback and Record I/O, according to your advice) samples buffer size.

What I don't get is why its still possible to get pops and cracks (which happens rarely at the moment, touch wood...) when you see on the performance monitor that Audio processing and Engine Load are way below 100%? I've never seen one of the cores max out during an audio glitch.

And why in some cases late buffers seem to be building up when a song project has been opened but is not playing??? 152ee80cbc

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