Mixing a real cabinet miking and a Torpedo Loadbox

On stage or in the studio, you may need to feel the sound coming out of a cabinet. You can simply connect a cabinet to the THRU output on your Torpedo Digital Loadbox. You will be able to hear the sound coming from the amp as usual, without any simulation or effect from the Torpedo.

Once you plug a cabinet into the THRU output:

  1. The amplifier is directly connected to the cabinet,
  2. The internal loadbox of the Torpedo is disconnected.


On this figure above, we have added an optional microphone to take the sound directly from the cabinet. That way you can get at the same time, in one take:

  1. A classic track with a real cabinet and microphone;
  2. A simulated track using the Torpedo library of speakers and microphones;
  3. If needed, a dry digital track with no simulation, for later re-miking.

This gives you a wide range of sonic possibilities, for you to make sure you get the tone you are looking for!

When you plug a cabinet into the THRU output, the impedance of that cabinet is the one that your amplifiers sees. In other words, with a 4 Ohms cabinet, you need to connect to the 4 Ohm output of your amplifier. In this configuration, the loadbox inside the Torpedo will be disconnected.

"Why am I hearing phase cancellation effects?" or fixing the latency-related issues when using the analog I/Os on the Torpedo

The Torpedo loadboxes embedding a digital signal processor are digital products. Digital means that the analog signal at the input will be converted into digital, processed, the converted back to analog.

The whole process takes a certain amount of time, usually called latency. We can describe it as the time needed by the signal to go from one input to one output. In the Torpedo Studio or VB-101, you can choose between two different latency modes (normal - 5.063ms or Low - 3.064ms). Please note that when switching to Low, the Overload parameter is no longer accessible, to save some CPU processing.

Mixing the output signal of the Torpedo Studio and a real cabinet miking will cause some issues and possibly a comb filter effect if you do not compensate for the latency.

To fix that, you will need to add a delay of the latency value on the microphone track (as the microphone sound will arrive "earlier" than the Torpedo sound), in the DAW or on the mixer, either in (milli)seconds or in samples.

In the following table we give you the rounded number of samples corresponding to the Normal or Low latency, in function of the sampling frequency of your project or digital mixer:

For alternative frequencies, please use the formula:

where nsample is the number of samples and fs is the sampling frequency. A sampling frequency of 44.1KHz means that you are recording (or playing) 44100 samples per second.

As the number of samples is usually an integer (some plugin may let you fix subsample delay, use as mutch precision as you can with the formula), you will never exactly match the exact latency value in milliseconds. Here is how we recommend you proceed to minimize the phase cancellation effect:

  1. Set a time/number of samples as close as possible from the Torpedo latency value (with the help of the table),
  2. mix the microphone and the Torpedo signals in mono,
  3. play around the value you set to find the maximum energy, the loudest signal.
  4. An alternative way to find the best value is to inverse the phase of one of the signals and search for the maximum attenuation. When you find it, put the track back on phase and control that you actually found the best value.

 "What if I am using the digital I/Os on my Torpedo?" or how to figure out the latency of the full digital system


It is not possible to determine a priori the exact latency of your system as soon as you use the digital I/Os of the Torpedo. Why is that? Because the digital product sending and/or receiving the signal to/from the Torpedo have a latency at its I/Os too. It is not an issue, per se, the issue is that the processing time of the analog to digital converted on your soundcard will never be the same as the processing time of its digital inputs.

In other words, the processing time on the audio interface will introduce another delay between the signal coming form the analog and digital inputs, and we do not know that extra delay.

Is that a lost war? NO.

  1. The most common way to compensate for the latency when you don't know its value is to zoom on the waveforms and move the Torpedo record so it will be time-aligned with the microphone record. To improve the precision, you can just "slap" your guitar or bass to produce a kind of a square waveform. The shorter the slap (in time) and stronger, the more precise you will be and it will be easier to find where you should focus your attention for the alignement. If you can't be as precise as you would want (because the DAW won't allow sub-sample positioning), just add a delay on the microphone track and play around to find the right value.
  2. Some DAWs embed a system able to measure and compensate for the outboard latency. For example, Pro Tools can offer that option, please refer to your DAW's user's manual to find out if it provides that kind of option. There are several tutorials online, here is one published by Sound on Sound: