> Has anyone managed to use buffer 32 without pops
yes
> and if so, what are the windows or driver tweaks i could do to achieve this?
- only some settings can be generalized / recommended for every system
- the rest is fully customized
- especially drivers and different versions of the same driver can have severe impact in terms of DPC latencies
- tweaking can take time and in some cases the design/BIOS of a mainboard can be bad, so that even the best tweaking can have limitations that results in a difference of around 200+ microseconds latency (or not), sounds less, but on system level this makes a difference how agile the system can react on a concrete workload
It is very different from system to system, best is to contact a company which is specialized for audio PCs, then you can expect least hassles.
If you want to measure the agility of your system and whether your system is suited for audio, then you can use LatencyMon, you can find a lot of threads here in the forum. It needs some experience to use this tool. All that I can say is to use it on an IDLE system as this tool itself creates already a DAW load on its own. Let it run for 10 minutes, then the values are more accurate and its more likely to catch a spike issues by background processes or scheduled tasks.
For a final validation let it run for several hours.
In terms of the Eventide and your goal
My overall impression is that your expectations of a PC recording system are too high overall if you expect to be able to achieve the same low latency values of a hardware-only solution with a PC.
With an Eventide, everything is built discretely and you will achieve the latency values reliably and independend on any "load" situation, that on a PC can easily occurr, as it has also to process many other things and drivers and settings might not be optimum.
Now with a PC or even Apple computer, several things come together:
1. Both operating systems are not real-time operating systems by design. The various drivers run on highest execution priority and may not be interrupted by a process scheduler for e.g. an important audio related task.
Instead of this its pure programming conventions when a driver detaches from a CPU ... Some drivers are written good, but not all. If the driver occupies a CPU core for too long where also an audio related task is waiting for execution on this CPU, then you can get problems and audio loss occurrs if audio can not be processed in time.
This problem also depends on other things like system load or also how the DAW has been coded and is able to distribute the DAW load efficiently across many cores.
2. the data transfer to a recording interface is done via a serial bus system (PCIe, USB, Firewire, Thunderbolt), which takes time. More time compared to the pure converter latency for AD and DA alone.
3. depending on the selected sample rate, the amount of audio data that has to be transferred in one time interval changes. The higher the sample rate, the higher the stress on the computer. More data has to be transferred in a shorter time interval.
4. Depending on how well the "overall system PC" is tuned by the BIOS/Windows settings and the drivers, depending on the sample rate and complexity of the DAW project and depending on the CPU consumption / requirements of the VST/VSTi used, you have more or less air to be able to work with the lowest ASIO buffersizes crackle-free.
The rule of thumb is actually to choose the highest possible ASIO buffersize for microphone recordings, because stability and reliability are more important here, instead of achieving the most latency-free monitoring possible.
5. The desire to get something like near real-time processing with a PC-based recording solution must be viewed in a more differentiated way.
For the audio communication within the routing matrix in the recording interface smth like near real-time is easier possible, here you only have to deal with the converter latency for A/D and D/A. As soon as you send something over the serial bus (PCIe, USB, FW, ...) then you always have to reckon with the full RTL. And this depends mainly on the ASIO buffersize.
And the objective is to choose the ASIO buffersize in a way that it fits to the project and the capabilities of the PC and that the chosen value is high enough to ensure stable and crackle-free operation.
5. Most important
Just because a minimum ASIO buffersize is selectable, it does not automatically mean that you can use it in every project and in combination with every computer.
It's like driving a car .. there you can also only drive at full speed where it is possible, but definitively not in curves.
There are many threads about how to optimize a PC with the goal to reduce DPC latency (PC internal latencies) to a minimum. But also this is no guarantee
- that it gives you the same optimum results like on other PCs
- that you can run all DAW projects with lowest ASIO buffer sizes
If you need to ensure latency free effects without audio loss .. I would have chosen a HW device like the Eventide
or I would have involved a company whether they have such a tested validated PC ...
At the end such PCs are more expensive, you might have ended up at the same price compareable to the Eventide.
BR Ramses - UFX III, 12Mic, XTC, ADI-2 Pro FS R BE, RayDAT, X10SRi-F, E5-1680v4, Win10Pro22H2, Cub14