Once you plug a cabinet into the THRU output:
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:
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.
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:
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.
http://www.soundonsound.com/sos/feb13/articles/pt-0213.htm
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