Plugindoctor v1.2.

3 manual
Thanks for using Plugindoctor! While the user interface is hopefully designed in
a way that the basic usage is more or less self-explaining, this manual should
provide you with some more detailed info on what is actually being analyzed in
all the different subsections. It is assumed that the reader has a basic
understanding of digital signal processing; expressions like "phase response"
or "harmonic content" will be used without further definition.

General usage:

Plugindoctor can load VST, VST3 and AU plugins (OSX only for the latter, of
course). Alternatively, Plugindoctor (from v1.0.8 onwards) can also be used to
add external hardware devices, see below for more detail. For Windows, you
were provided separate 32 and 64 bit versions of Plugindoctor, please use the
corresponding program to load 32 or 64 bit plugins. On Mac, a Universal Binary
has been installed which runs in 64 bit mode by default. To analyze 32 bit
plugins, right-click on the application in the OSX finder, click on "Information"
and tick "Run in 32 bit modus".

Plugins can be loaded either directly from disk, or via a plugin browser. The
latter is shown by clicking "Files → Show plugin browser" or by clicking on the
library/book symbol in the upper right corner. Initially, it will be empty. You
can use either the "Scan" button to perform a global scan of all supported
formats (recommended to quickly populate the plugin browser) or make more
specific scans via the "Options" menu, which also allows you to remove
selected plugins from the list. Once you have a list of plugins in your browser,
simply load one by selecting it and clicking the "Load plugin" button, or by
double-clicking on the plugin name in the list.

Alternatively, if you want to load a plugin directly from a location on your

harddisk, use "Files → Load plugin". Either way, once you have selected a
plugin for analysis, its window will pop up, and the analysis signal will be
routed through the plugin's in- and outputs. The plugin window can be closed
by either clicking the close button on the plugin window or the eye symbol in
the upper right corner of Plugindoctor. A second click on the eye symbol will
display a closed plugin window again.Plugindoctor has a range of different
analysis modes which can be accessed via the orange tab buttons. Those
modes are explained in detail below.

The camera symbol lets you take a snapshot of Plugindoctor. A file browser will
pop up, which will let you choose the location and the name of the generated
PNG file. Finally, the settings symbol will open up the "Settings" window (also
explained in more detail below).

You can zoom into all graphs except in "Performance". Zooming can be done in
two ways: first, you can select a rectangle with your mouse, starting in the
upper left corner, will draw the selected rectangle while dragging the mouse in
the lower right direction, and zoom into the corresponding regions upon
releasing the mouse button. Zoom out again by clicking and dragging into the
upper-left direction.
Second, you can also zoom and move the whole graph using your mouse
wheel. Place the mouse cursor to the left of the graph to affect the y-axis, or
to the bottom of the graph to affect the x-axis. Simple mouse wheel scrolling
will move the graph along the desired axis, while holding down the CTRL
button will zoom in or out along the axis. To reset everything, drag a rectangle
from lower left to upper right corner using your mouse again.

Linear analysis

This section features two modes of input signals for the analysis ("Delta" and
"Random"), and two modes of analyzing a plugin's response ("Magnitude" and
"Phase"). "Delta" refers to the input signal being a so-called delta peak, which
means the first sample is equal to 1, all the other samples are zero. This input
signal contains a flat contribution of all admissible frequencies, i.e. if you would
send this signal through a frequency analyzer, you'd get a flat curve when
looking at the magnitudes of the contributing frequencies, and also a flat curve
when looking at the phase response. This means that all deviations from a flat
curve come from the plugin being analyzed.
"Random" will send white noise with a maximum peak level of 0 dB through
the plugin. This signal has a flat frequency response too, but only if you
average it over a sufficiently long time. The phase response, on the other
hand, is flucuating randomly.
Concerning the analysis of the output signal, "Magnitude" will display the
absolute values of the contributing frequency components in the Fourier-
transformed output of the plugin being tested, while "Phase" will display the
corresponding phase response. The default FFT size is 16384 samples. This
size can be increased twice- or fourfold in the "Settings" window (see
discussion further below).

In the linear analysis window, a curve can be stored pressing the "Store"
button and will then continue to be displayed when the loaded plugin's
parameters are being changed or a new plugin is being loaded, for easy
comparison between settings or plugins. "Clear" will remove this stored curve
again.

Harmonic analysis

The aim of looking at the harmonic response of a plugin is to see the response
of a plugin when it is being fed with a signal of precisely defined frequency
content (as opposed to the delta peak from the linear section).

a: Single frequency mode

Here we have the choice between two different methods of harmonic analysis:
"THD" (total harmonic distortion) and "IMD" (intermodular distortion). In THD
mode, the input signal consists of a pure sine wave (the frequency and
intensity of which can be changed in the "Settings"). The output will typically
consist of a large peak at the input frequency plus, if the plugin generates
harmonic content due to a nonlinear algorithm, one or more peaks, typically at
integer multiples of the input frequency (higher harmonics). When aliasing
occurs, higher harmonics that are theoretically higher than the Nyquist
frequency are being produced, which will be reflected off the Nyquist
frequency's location and lead to additional peaks at non-integer values of the
input frequency.
Apart from visually inspecting the resulting signal, two measures of the
distortion generated by the plugin are being displayed in this mode, which will
be shown in the upper right corner of the graph: THD and THD+N. THD is
defined as the inverse ratio of the magnitude of the input sine signal and the
summed magnitudes of the peaks at integer multiples of the input frequency.
THD+N is the inverse ratio of the magnitude of the input signal and the
magnitude of "everything else", which includes all higher harmonics but also
everything in between.

"IMD" mode uses two input frequencies, a low and a high one, with an
intensity difference of 12 dB, i.e. the lower frequency comes in a 0 dB, the
higher at -12 dB. The lower input frequency is set to 60 Hz, the higher
frequency to 7000 Hz. If the plugin is creating additional harmonic content,
this will now not only show up in higher harmonics of the two input
frequencies, but also in modulated peaks of the higher frequency by the lower
one: you will see a peak at 60 Hz, a peak at 7000 Hz and, if there is
intermodular distortion, several peaks at 7000 +- N*60 Hz. Plugindoctor
calculates a numerical value for IMD by dividing the RMS-summed
contributions of 10 modulated peak lower than 7000 Hz and 10 modulated
peak higher than 7000 Hz (all peaks with a 60 Hz distance between 6400 Hz
and 7600 Hz) by the magnitude of the incoming 7000 Hz peak.

b: sweep mode

In this mode, the reponse of the tested plugin is measured over a whole range
of excitation frequencies: toggle the "sweep" button and Plugindoctor will
continuously sweep from low to high frequencies. You will note that the button
that, before toggling the "sweep" button, was used to switch between THD and
IMD, now lets you choose between THD and fundamental. Set it to THD, and
Plugindoctor will calculate the THD response per frequency. Set it to
fundamental, and you will see the gain or reduction in dB of the currently
applied single frequency wave. For strictly linear plugins, this reponse will be
identical to the results from the "Linear" section; as soon as nonlinearities
come into play, the two measurements are usually different.
The lowest frequency is defined by having to fit 10 oscillations into the audio
buffer that's used for measurement, so it will be influenced by both your
sample rate and your quality settings (the latter is changing the buffer size).
Please note that, for the THD calculation, the results will go to zero as the
sweep frequency approaches ¼ of the sample rate, as there will be no more
higher harmonics left between the applied frequency and the Nyquist
frequency (½ the sample rate).

Oscilloscope: the oscilloscope window displays the audio signal coming out of
the plugin, with a sine wave as input signal. This mode is useful for precisely
looking at possible deformations of a sine wave due to, e.g. a very fast
compressor or a distortion plugin. There are two ways to display the audio
data: when "Time" is selected using the button in the upper right corner, the
x-axis will display the time in seconds, while the y-axis shows the audio signal
coming out of the plugin. When "Wavesh." is selected, the x-axis corresponds
to the audio signal going into the plugin, while the y-axis corresponds to the
outgoing signal again. This mode is useful to see how the plugin shapes the
incoming audio. In waveshaping mode, a slider can be used to adjust the delay
in samples between the in- and outgoing audio data on display. It is set to the
latency that's reported by the plugin per default, but delays can occur even
with a plugin that officially has a latency of zero samples, due to nonlinear
phase effects.

Dynamics looks at a plugin's reaction to varying input levels. Again, the input
signal is a single sine wave whose frequency can be changed in the settings.
There are two modes here: "Ramp" and "Attack/Release". In "Ramp" mode,
the plugin is sequentially fed a sinewave with an input level increasing (by
default) from -100 dB to 0 dB in steps of 1 dB, and then returning to -100 dB
in an endless loop. The level of the sine wave is held constant for 16384
(quality: normal), 2*16384 (quality: higher) or 4*16384 (quality: highest)
samples, with the quality being adjustable in the "Settings" section. The
maximum amplitude of the plugin's output signal is recorded and displayed as
a function of the input signal. Using the respective editable labels in the
Dynamics tab, you can change the default ramp parameters.
In Attack-Release mode, the plugin is being fed a sequence of three sine wave
with three different levels. The lengths for which each of the three waves is
present, as well as their levels, can again be changed in "Settings". Typically,
one would look at a certain time span with an input level below a compressor's
threshold, then a certain time span above the threshold, and again a third time
span below the threshold. By looking at the displayed output signal of the
plugin that is being fed this type of signal, one can see how fast a compressor
reacts when the signal exceeds its threshold ("attack") or when the signal falls
back below the threshold ("release").

Hammerstein: this tab provides a more detailed look at the nonlinear

behaviour of a plugin or a piece of hardware. Briefly speaking, the
Hammerstein model tries to fit the system under investigation with a power
series, with each component of the series multiplied by its own linear filter.
You can select the order of the power series, with a maximum of 7 allowed.
That way you can see not only how much a certain higher harmonic (say, x^3)
is contributing to the signal, but also how its response is different across
different frequencies.
Technically, the Hammerstein model consists of the diagonal terms of the
corresponding Volterra kernel. This manual is not the right place to provide a
more detailed explanation, I'd kindly refer you to https://ant-
novak.com/pages/sss/ for a more detailed explanation of the underlying
analysis method.

Performance: this module measures the time spent in a plugin's processing

callback as the number of samples the plugin is processing. There will be some
fluctuation due to the varying CPU load of your computer, but generally it will
give you a good idea whether your plugin compares favourably with other
plugins doing comparable processing tasks, or whether there are specific
sample sizes which take more time than they should (maybe due to
reallocations).

The Settings page (which you can reach via the button with the standard
settings symbol in the upper right corner of Plugindoctor) lets you control a
number of values that are being used in the analysis and when displaying the
results:

• the first row lets you select the overall sample rate
• in the next row, "quality" refers to both the global buffer size, and the
FFT size being used in "Linear" and "Harmonic". "Normal" corresponds to
16384 samples, with "Higher" and "Highest" increasing this number by a
factor of 2 each.
• Linear range is the maximum vertical dB scale in the "Linear" analysis
display.
• "Test frequency" lets you select the input frequency for "Harmonic
->THD", "Oscilloscope" and "Dynamics". The input frequency is not freely
selectable, but you are given a choice of possible values such that the
input sine signal has an integer number of oscillations in the input buffer,
which leads to the most precise calculation when analysing the output
signal via FFT.
• "Harm. sig. amp. dB" is the input intensity of the sine wave in the
"Harmonics → THD" mode.
• Finally, the three remaining rows let you specify the lengths and
intensities of the three sine signals in "Dynamics → Att./Rel.". Just click
on the numbers to edit their values.

If you want to analyze external audio hardware, please click the "Show audio
hardware settings" button to properly route your signal to and from the
external device. Finally, use the "Save as default" button to set all current
settings as default.

Hardware analysis

To analyzer external audio hardware like an outboard EQ or compressor, you

need to attach the hardware gear to your audio interface, with one output
going from the interface to the hardware's input (can be mono or stereo) and
the output from the hardware going back to an input channel of your audio
interface. In Plugindoctor's settings, use the "Show audio hardware settings"
dialog to select the in- and output of your interface that you have used to
connect the piece of hardware. On Windows, you can also choose which driver
you want to use: DirectSound, Windows Audio or ASIO (if your device supports
the latter). Please make sure to use the same sample rate in the hardware
settings dialog that you've also chosen in Plugindoctor's settings, otherwise
you'll get incorrect results.
Once everything is set up, simply toggle the "Use hardware" button in the
upper right corner of Plugindoctor and all analysis signals will be routed
through your hardware chain. From here on, all analysis will be performed in
exactly the same way as if you were analysing an audio plugin. To stop
analyzing your hardware and go back to plugin analysis, just toggle the "Use
hardware" button again and select a plugin to analyze as usual.
Your hardware loop will inevitably have a non-zero latency, which will have an
effect on the calculated phase response in the linear analysis section. This
phase response is real, in the sense that the delay is actually taking place
when you are integrating your outboard gear. Nevertheless, you might be
interested in the latency-corrected phase response. For this purpose, click the
"Adjust HW latency" button, wait one or two second, and the latency of your
whole hardware chain will be taken into account. This needs to be redone when
changing the sampling frequency.

Should any questions remain, or problems pop up during the usage of

Plugindoctor, please do not hesitate to contact [email protected]!