What is the buffer size calculate rate ( = speed at which one buffer is calculated to be processed at ) based upon ?


Is it the resonance of some crystal component, or what gives the different buffer sizes


What if i want to have buffer size of two ( 2 ), that is calculated 0.5 x/ twice the speed of regular buffer calculation — how would i speed up my system to do that ?

 

The buffer can hold samples. Let's say you set your soundards sample rate at 256 samples/second (just hypotetically). A 256 buffer will hold one second of audio. Now let's say you set your soundards sample rate at 512 samples/second. A 256 buffer will hold HALF of a second of audio. If you set your soundcard sample rate at 1024 samples/second, a 256 buffer will hold a QUARTER of a second of audio. The bigger your sample rate is, for the same buffer size, the smaller the latency.

You can only speed up the system in 2 ways: a very very fast CPU and a soundcard capable of a very small buffer and very high sample rates. I am not sure you could use that computer (because almost all CPU processing will go towards processing the buffer) but that's the only way to get smaller latency

 

Thank you

I would like to speed up my system thinking one second is faster than it really is

 

First, get yourself a great audio interface with fast drivers; measure round-trip latency.
Then try with the lower buffer size (32 samples).
Finally, use plugins that do not add additional latency (I stumbled upon some Zero-latency plugins - list)

 

How to beat ( all ) latency ?

your DAW latency compensation algorithm will do it for you even if you have 10 seconds latency you can play and record your music perfectly in time, no need to have a quantum computer 1Km under the ground to do that.
Relax and make music.

 

Not if you need live monitoring for ex recording VSTi with MIDI keyboard or guitar into Ampsim etc...

 

Latency compensation for external hardware only works the way he seems to think it does, if midi data is played correctly on time within the DAW and audio being returned to the daw is recorded. the incoming/recorded audio can then be corrected via latency compensation. the adjustment is the time it takes for a midi note-on to occur in the DAW until the resulting incoming audio is captured by the audio interface.

to get around the vsti scenario, you have to play and record the midi data to the daw. if the monitoring delay causes a timing issue for the player, you can just quantize the data after it is recorded. you can get this to work the way he thinks then, but at the cost of doing a second recording pass. One is plugin delay compensation (pdc), and one is latency compensation. different things.

 

Last edited: Nov 6, 2023