The sample rate is the frequency in which the original audio signal is sampled to be converted into digital audio. A sample rate of 44100 Hz means that the signal is being sampled 44100 times per second. You can increase the sample rate to obtain a better resolution, but as a rule of thumb, to accurately reproduce the original signal, we need a sample rate twice as high as the highest frequency we want to reproduce. Most humans can’t hear beyond 16 - 18kHz, which means that 44.1kHz or 48kHz is more than enough to accurately reproduce the entire audible spectrum.
The buffer size indicates the number of samples per “batch” of audio that’s processed by your system. Lower buffer sizes lead to lower latency at the cost of more CPU usage. At 128 or 64 samples, the latency is negligible. The sample rate also dictates the “duration” of each sample, which means that a buffer size of 128 at 48kHz will be shorter than a buffer size of 128 at 44.1kHz.
As mentioned above, the optimal settings should be around 44.1kHz and 128 samples. Our Plugins also include oversampling, which solves any potential aliasing caused when processing the signal internally sampling the audio at a higher sample rate. You can learn more about this here.