What is the sample rate?

In digital audio, the sample rate is literally how fast samples are taken. Sampling rate or sampling frequency defines the number of samples per second taken from a continuous signal to make a discrete or digital signal. For time-domain signals like the waveforms for sound (the ones you can see in your DAW), frequencies are measured in hertz (Hz) or cycles per second.

There is an interesting thing about sample rates by using Neural DSP plug-ins:

Over 88.2Khz the plug-in won't upsample anymore. So, running on 44.1Khz with high oversampling is the same as running at 88.2 with normal oversampling (since high oversampling doesn't work anymore). 

When you are in 96KHz with high oversampling, you would be running at 384KHz internally, which consumes a lot of CPU and the sound doesn't improve anymore.

You can control the oversampling in any Neural DSP plug-in by clicking on the QUALITY SWITCH. It changes the quality with which the plugin will process the signal, based on different levels of oversampling (2x Normal, and 4x High). The higher the quality, the more processing CPU power needed.

So, basically we recommend using our plugins in 44.1Khz or 48Khz audio sessions with HIGH oversampling settings to get the best quality possible.


How does the buffer size works?

While we all want as little latency as possible, the buffer size is dependent on a number of things such as how many plug-ins are loaded on a track, and the computer’s processing power.

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, however, there will be quite a bit of latency which can be frustrating.

Here are a few tips when working with the buffer size:

Low buffer Size:

Set the buffer size to a lower amount to reduce the amount of latency for more accurate monitoring. The downside to lowering the buffer size is that it puts more pressure on your computer’s processors and forces them to work harder.

Use as few plug-ins as possible during the tracking phase so that your computer’s processing bandwidth is uninhibited. If you start to choke your processors with other tasks, you will experience clicks and pops or errors which will make tracking your project a nightmare. Some DAWs like Pro Tools or Logic Pro X features "Low Latency Mode", that reduces the latency in high buffer size settings.

You can usually raise the buffer size up to 128 or 256 samples without being able to detect much latency in the signal.

High buffer size:

In the mixing phase, you will be monitoring playback only, so it's safe to raise the buffer size to a higher setting since you are no longer monitoring live signals. This allows you to use more plug-ins before encountering clicks and pops or errors, depending on your computer’s resources and limitations.

You may notice a slight delay when you start playback in your DAW with the buffer turned all the way up, but this is normal and is not a sign that your DAW is choking.