We've already covered why musicians should practice in headphones with Realphones. Now for the technical side: how to connect your instrument, where latency comes from, and how to make sure it does not get in the way.

The first step is to choose how the audio will get into your computer. There are several scenarios here.

If you have a multi-effects unit or guitar processor - first check whether it can work as a USB audio interface. Most modern units can. If it can, use that and do not connect the processor to a separate interface via its analog outputs.

An instrument with a pickup (electric guitar, bass, violin with a piezo pickup) - you will need an audio interface with an instrument input (Hi-Z). This is a standard input that will pass the signal without loss. It is important to set the input level correctly: at your most aggressive attack, the peak level should not exceed -6 dBFS. This gives you headroom before clipping and protects against distortion.

A synthesizer or keyboard instrument with a line output - connect it to a line input (Line In). The main rule stays the same: make sure your peaks do not go above -6 dBFS.

How to Check the Level

In Cubase: open MixConsole (F3), find the input channel - it is usually named after the input (for example, Stereo/Mono In 1). The peak value is shown as a number to the right of the level meter.

In Reaper: create an empty track and select as input the audio interface channel your instrument is connected to, then click the red Record Arm button. The level will appear on the track meter.

In both cases, the logic is the same and will not differ in any fundamental way in any other DAW: play at full force and watch the peak value. If it is above -6 dBFS, turn down the Gain on the interface and reset the peak by clicking the number. Repeat until you get the right result.

Latency is the time between the moment you produce a sound and the moment you hear it in your headphones. It arises during digital conversion and signal processing.

Latency consists of several parts:

●      Input latency - the time it takes for the interface to digitize the signal;
●      The buffer is a "holding area" for samples that the computer processes. The smaller the buffer, the lower the latency, but the heavier the load on the CPU;
●      Output latency - the time required for the reverse digital-to-analog conversion;
●      Plugin latency - some plugins add tens or even hundreds of milliseconds to the total latency on their own.

Buffer-related latency is easy to calculate:
Latency (ms) = Buffer Size / Sample Rate (kHz). At a sample rate of 48,000 Hz and a 64-sample buffer, we get: 64 / 48 = 1.33 ms for one pass alone. For perspective: sound travels through air at ~343 meters per second, so 1 ms is about 34 centimeters. The real total latency of the signal path with a 64-sample buffer will be 2.5-4 ms, which is comparable to a distance of about 1.4 meters from the sound source. In practice, this amount of latency is not perceptible.

A 256-sample buffer at 48 kHz is already 5.3 ms just for buffering, and once you factor in the interface's input and output latency, the total latency of the signal path becomes noticeably higher. Most musicians can already feel it at those settings. A 1024-sample buffer is beyond the comfort zone for live playing. If the processor does not support operating as an audio interface and is connected to a separate interface via its analog output, that means two extra signal conversions, each of which adds its own latency to the total. Check the documentation and see how many milliseconds the processor itself adds - the final latency may turn out to be quite noticeable.

How to Check Plugin Latency in Reaper: 
The total latency of all plugins on the track is shown as a number at the bottom of the FX window. For live playing, aim for plugins with latency up to 128-256 samples. Plugins with latency in the thousands of samples are not suitable for monitoring while playing - they add so much delay that you pluck the string and hear the sound noticeably later. At that point, playing becomes almost impossible.

Different instruments call for different signal chains:
 
●      Electric guitar/bass: virtual amp + cabinet (IR). Add reverb and delay carefully, keeping an eye on the overall latency of the chain;
●      Acoustic instruments with piezo pickups: light EQ as a starting point.
 
Since Realphones does not add any latency of its own, you can safely use it for home practice and recording.

If you do not need plugins and already have a finished sound from your processor or synthesizer, you do not need to open a DAW at all. Realphones System-Wide lets you monitor the signal directly from your audio interface inputs, with headphone correction and space emulation.
 
How to set it up:
 
1. Open the Realphones System-Wide settings (the gear icon in the bottom-right corner).
 
2. Find the Use Realphones System-Wide line input checkbox (it is enabled by default). In this mode, Realphones processes all system audio sent to the Realphones System-Wide virtual audio device. We need a different mode, so disable this checkbox. You can now select specific inputs of your audio interface directly.

3. Open Audio-MIDI Settings.

4. Disable the Feedback Loop checkbox and choose the input your instrument or processor is connected to in the Active input channels section.

5. Set the buffer size to the lowest stable value - start with 64 samples and increase it if you hear clicks or dropouts.
 
6. Make sure the sample rate matches both in your audio interface settings and in the Realphones System-Wide virtual device. A sample rate mismatch is the most common cause of unstable operation.
 
7. Restart Realphones System-Wide. You will now hear the instrument or voice with the selected space emulation and headphone correction.

Once the gear is set up and latency is reduced to a minimum, the most important stage begins: creating the right acoustic atmosphere. In Realphones, space is not just a reverb effect, but a full acoustic environment emulation that determines how you will work on sound production and dynamics.

We covered how to use the Acoustic Collection expansion for detailed work on material in this article. It already breaks down scenarios that will help you adapt home practice to specific creative tasks. 

In addition, Realphones Ultimate gives you access to a wide range of other professional venues:


●      Concert Hall Medium and Concert Hall Large are your primary tools for preparing for large stages and festivals. Rehearsing in the emulation of a huge hall teaches you to play with the space. Here it is important to watch how quickly the reverb tails die away and whether your playing turns into muddy blur. A great choice for feeling the scale of the sound and learning to leave pauses where the room acoustics require them.

 ●      Public Spaces is a set of emulations that imitate small venues: cafes and intimate clubs. In such conditions there are no long reflections to forgive performance flaws. If you are preparing for a show in smaller rooms, choose these spaces. They will force you to pay closer attention to nuances of sound production and to the dynamic balance between the instrument and the arrangement.

●      Night Club is a must-have for DJs and electronic musicians. This mode recreates the specific low-frequency response and the very sound pressure you will encounter on the dance floor. Rehearsing in Night Club lets you understand in advance how your synthesizers and drum machines will physically feel in the mix, and whether you have gone too far with the sub-bass at home.

And most importantly: if you are performing live through the same processor or plugin chain you rehearse through, dial in your sound directly in the emulation of the target venue. What sounds good in headphones may behave very differently on stage.
 
Practicing in the right environment is the best insurance against surprises at soundcheck!