Two Notes Torpedo Studio User Manual page 20

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2017/11/08
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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 the rounded number of samples
corresponding to the Normal or Low latency, in function of the sampling frequency of your project or
digital mixer:
Latency / Delay value (ms) 32KHz 44.1KHz 48KHz 88.2KHz 96KHz 176.4KHz 192KHz
Normal - 5,063
Low - 3,064
For alternative frequencies, please use the following formula: nsample = latency × fs, 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 an Integer, you will never match the exact latency value in milliseconds.
Here is how we recommend you proceed to minimize the phase cancellation effect:
Set a time length/number of samples as close as possible from the Torpedo Studio latency value
(with the help of the table)
mix the microphone and the Torpedo signals in mono
play around the value you set to find the maximum energy, the loudest signal
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 in phase and confirm that
you actually found the best value.
4.6.2 "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 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 really the problem however. 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.
Can anything be done to remedy this? YES
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 (such as Pro Tools) are able to measure and compensate for the outboard latency.
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Printed on 2017/11/10 22:17