meldaproduction MeldaProduction MCompressor [Freeware] MeldaProduction MCompressor Help File

MCompressor

Presets button

Presets button shows a window with all available presets. A preset can be loaded from the preset window by
double-clicking on it, using the arrow buttons or by using a combination of the arrow keys and Enter on your
keyboard. You can also manage the directory structure, store new presets, replace existing ones etc. Presets are
global, so a preset saved from one project, can easily be used in another.

Holding Ctrl while pressing the button loads an existing preset, selected at random.

Presets can be backed up by using either the Export button, or by saving the actual preset files, which are found
in the following directories:
Windows: C:\Users\{username}\AppData\Roaming\MeldaProduction
Mac OS X: ~/Library/Application support/MeldaProduction

Exported preset files can be loaded into the plug-in's preset store using the Import button. Or the preset files
themselves can be copied into the directories named above.

Files are named based on the name of the plugin in this format: "{pluginname}presets.xml", for example:
MAutopanpresets.xml or MDynamicspresets.xml. If the directory cannot be found on your computer for some
reason, you can just search for the particular file.

Left arrow button

Left arrow button loads the previous preset.

Right arrow button

Right arrow button loads the next preset.

Randomize button

Randomize button loads a random preset.

Bypass button

Bypass button (un)bypasses the plugin. If enabled, the plugin doesn't actually process any signal, so it also
saves CPU power. You can use it instead of your host's bypass button. In most cases the outcome will be the
same.

Panic button

Panic button resets the plugin state. You can use it to force the plugin to report latency to the host again and to
avoid any audio problems.
For example, some plugins, having a look-ahead feature, report the size of the look-ahead delay as latency, but
it is inconvenient to do that every time the look-ahead changes as it usually causes the playback to stop. After
you tweak the latency to the correct value, just click this button to sync the track in time with the others,
minimizing phasing artifacts caused by the look-ahead delay mixing with undelayed audio signals in your host. It
may also be necessary to restart playback in your host.
Another example is if some malfunctioning plugin generates extremely high values for the input of this plugin. A
potential filter may start generating very high values as well and as a result the playback will stop. You can just
click this button to reset the plugin and the playback will start again.

Settings button

Settings button shows a menu with additional settings of the plugin. Here is a brief description of the separate
items.

Activate lets you activate the plugin if the drag & drop activation method does not work in your host. In this

case either click this button and browse to the licence file on your computer and select it. Or open the licence
file in any text editor, copy its contents to the system clipboard and click this button. The plugin will then
perform the activation using the data in the clipboard, if possible.

There are 4 groups of settings, each section has its own detailed help information: GUI & Style enables you to
pick the GUI style for the plug-in and the main colours used for the background, the title bars of the windows
and panels, the text and graphs area and the highlighting (used for enabled buttons, sliders, knobs etc).

Advanced settings configures several processing options for the plug-in.

Dry/wet affects determines, for Multiband plug-ins, which multiband parameters are affected by the Global

dry/wet control.

Smart interpolation adjusts the interpolation algorithm used when changing parameter values; the higher the

setting the higher the audio quality and the lower the chance of zippering noise, but more CPU will be used.

WWW button

WWW button shows a menu with additional information about the plugin. You can check for updates, get easy
access to support, MeldaProduction web page, video tutorials, Facebook/Twitter/YouTube channels and more.

Gain

Gain defines power modification applied to the incoming signal. If you set ratio to 1:1 and custom shape is
disabled, then the plugin works simply as a fast gain processor.

Range: -24.00 dB to +24.00 dB, default 0.00 dB

Output gain

Output gain defines the power modification applied to the output signal. If you set ratio to 1:1 and custom
shape is disabled, then the plugin works simply as a fast gain processor.

Range: -24.00 dB to +24.00 dB, default 0.00 dB

Attack

Attack defines the attack time, that is how quickly the level detector increases the measured input level. When
the input peak level is higher than the current level measured by the detector, the detector moves into the
attack mode, in which the level is increased depending on the input signal. The higher the input signal, or the
shorter the attack time, the faster the measured level rises. Once the measured level exceeds the Threshold
then the dynamics processing (compression, limiting, gating) will start.

There must be a reasonable balance between attack and release times. If the attack is too long compared to
the release, the detector will tend to keep the measured level low, because the release would cause that level to
fall too quickly. In most cases you may expect the attack time to be shorter than the release time.

To understand the working of a level detector, it is best to cover the typical cases:

In a compressor the attack time controls how quickly the measured level moves above the threshold and the
processor begins compressing. As a result, a very short attack time will compress even the beginning transient
of a snare drum for example, hence it would remove the punch. With a very long attack time the measured level
may not even reach the threshold, so the compressor may not do anything.

In a limiter the attack becomes a very sensitive control, defining how much of the signal is limited and how
much of it becomes saturated/clipped. If the attack time is very short, limiting starts very quickly and the limiter
catches most peaks itself and reduces them, providing lower distortion, but can cause pumping. On the other
hand, a higher attack setting (typically above 1ms) will let most peaks through the limiter to the subsequent in­built clipper or saturator, which causes more distortion of the initial transient, but less pumping.

In a gate the situation is similar to a compressor - the attack time controls how quickly the measured level can
rise above the threshold at which point the gate opens. In this case you will usually need very low attack times,
so that the gate reacts quickly enough. The inevitable distortion can then be avoided using look-ahead and hold
parameters.

In a modulator, the detector is driving other parameters, a filter cut-off frequency for example, and the situation
really depends on the target. If you want the detector to react quickly on the input level rising, use a shorter
attack time; if you want it to follow the flow of the input signal slowly, use longer attack and release times.

Range: 0 ms to 1000 ms, default 10 ms

Release

Release defines the release time, that is how quickly the level detector decreases the measured input level. The
shorter the release time, the faster the response is. Once the attack stage has been completed, when the input
peak level is lower than the current level measured by the detector, the detector moves into the release mode,
in which the measured level is decreased depending on the input signal. The lower the input signal, or the
shorter the release time, the faster the measured level drops. Once the measured level falls under the

Threshold then the dynamics processing (compression, limiting, gating) will stop.

There must be a reasonable balance between attack and release times. If the attack is too long compared to
release, the detector would tend to keep the level low, because release would cause the level to fall too quickly.
Hence in most cases you may expect the attack time to be shorter than the release time.

To understand the working of a level detector, it is best to cover the typical cases:

In a compressor the release time controls how quickly the measured level falls below the threshold and the
compression stops. As a result a very short release time makes the compressor stop quickly, for example,
leaving the sustain of a snare drum intact. On the other hand, a very long release keeps the compression
working longer, hence it is useful to stabilize the levels.

In a limiter the release time keeps the measured level above the limiter threshold causing the gain reduction.
Having a very long release time in this case doesn't make sense as the limiter would be working continuously
and the effect would be more or less the same as simply decreasing the input gain manually. However too short
a release time lets the limiter stop too quickly, which usually causes distortion as the peaks through the limiter
to the subsequent in-built clipper or saturator. Hence release time is used to avoid distortion at the expense of
decreasing the output level.

In a gate the situation is similar to a compressor - the release time controls how quickly the measured level can
fall below the threshold at which point the gate closes. Having a longer release time in a gate is a perfectly
acceptable option. The release time will basically control how much of the sound's sustain will pass.

In a modulator, the detector is driving other parameters, a filter cut-off frequency for example, and the situation
really depends on the target. If you want the detector to react quickly on the input level falling, use a shorter
release time; if you want it to follow the flow of the input signal slowly, use longer attack and release times.

Range: 0 ms to 1000 ms, default 50 ms

RMS length

RMS length smoothes out the values of the input levels (not the input itself), such that the level detector
receives the pre-processed signal without so many fluctuations. When set to its minimum value the detector
becomes a so-called "peak detector", otherwise it is an "RMS detector".

When you look at a typical waveform in any editor, you can see that the signal is constantly changing and
contains various transient bursts and separate peaks. This is especially noticeable with rhythmical signals, such
as drums. Trying to imagine how a typical attack/release detector works with such a wild signal may be
complex, at least. RMS essentially takes the surrounding samples and averages them. The result is a much
smoother signal with fewer individual peaks and short noise bursts.

RMS length controls how many samples are taken to calculate the average. It stabilizes the levels, but it also
causes a slower response time. As such it is great for mastering, when you want to lower the dynamic range in
a very subtle way without any instabilities. However, it is not really desirable for processing drums, for example,
where the transient bursts may actually be individual drum hits, hence it is usually recommended to use peak
detectors for percussive instruments.

Note that the RMS detector has 2 modes - a simplified approximation is used by default, and a true RMS is

processor can be enabled from the advanced settings (if provided). Both respond differently, neither of them is
better than the other, they are simply different.

Range: Peak to 100 ms, default 1.0 ms

Threshold

Threshold determines the minimal signal power at which the effect starts to apply.

Range: -80.0 dB to 0.00 dB, default -12.0 dB

Ratio

Ratio defines the compression ratio of the input signal above the threshold.

Range: 1.00 : 1 to 20.00 : 1, default 1.80 : 1

Knee size

Knee size defines size of the knee if not hard.

Range: 0.00% to 100.0%, default 25.0%

Knee type

Knee type defines the type of knee separating the compressed part from the uncompressed one.

Link channels

Link channels defines if the power is defined by all channels together instead of compression based on separate
channel signal powers.

Maximize to 0dB

Maximize to 0dB defines if the resulting signal power should be maximized to approach 0dB if possible.

Custom shape

Custom shape is useful to specify your own dynamics processing shape, whether for compression, gating or
some more extraordinary effects.

Side-chain panel

Side-chain panel lets you do additional filtering of the level detector input. Please note that its name does not
mean it is related to the processor's secondary input (if it has one). Its main purpose is to filter the signal fed to
the level detector. This is useful when, for example, you want to remove high volume peaks of a particular
frequency.

For example, de-essing typically reduces "s" sound contained at about 2.5kHz and above, which is often far
louder than the rest of the recorded voice signal. As another example, you may be processing a drumset, where
the hi-hat is too prominent. You would like to compress the hi-hat, but keep the rest of the drumset intact. So
you filter out everything except for the hi-hat in the level detector's side-chain and the compressor will listen to
the hi-hat only. It is rarely possible to filter out everything except for the requested signal, so compromises need
to be made.

Note that the filtered signal is typically lower in amplitude, but volume maximization is not recommended, since
it may increase the volume to dangerous levels.

Enable button

Enable button enables or disables the side-chain filter.

Audition button

The Audition button toggles playback of the filtered signal instead of the actual effect output. When enabled,
you will hear the actual filtered level detector signal. This may be processed in various ways, but in most cases
you will be interested in setting up the side-chain filter.

Side-chain input

Side-chain input switch activates the secondary plugin input, which lets you process the main signal, but follow
the side-chain level instead. This is useful for all sorts of creative effects such as pumping the whole mix by bass
drum.

Signal graph

Signal graph defines the compression envelope. X axis represents input level, Y contains the output level. You
can edit it if you enable Custom shape.

Collapse button

Collapse button minimizes or enlarges the panel to release space for other editors.

Plugin toolbar

Plugin toolbar provides some global features, A-H presets and more.

Upsampling

Upsampling can potentially improve sound quality by processing at a higher sample rate. Processors such as
compressors, saturators, distortions etc., which employ nonlinear processing generate higher harmonics of the
existing frequencies. If these frequencies exceed the Nyquist rate, which equals half of the sampling rate, they
get mirrored back under the Nyquist rate. This is known as aliasing and is almost always considered an artifact.
This is because the mirrored frequencies are no longer harmonic and sound as digital noise as this effect does
not physically occur in nature. Upsampling (or oversampling) reduces the problem by temporarily increasing the
sampling rate. This moves the Nyquist frequency which in turn, diminishes the level of the aliased harmonics.
Note that the point of upsampling is not to remove harmonics, we usually add them intentionally to make the
signal richer, but to reduce or attenuate the harmonics with frequencies so high, that they just cannot be
represented within the sampling rate.

To understand aliasing, try this experiment: Set the sampling rate in your host to 44100 Hz. Open MOscillator
and select a "rectangle" or "full saw" waveform. These simple waveforms have lots of harmonics and without
upsampling even they become highly aliased. Now select 16x upsampling and listen to the difference. If you
again select 1x upsampling, you can hear that the audio signal gets extensively "dirty". If you use an analyzer
(MAnalyzer or MEqualizer for example), you will clearly see how, without upsampling, the plugin generates lots
of inharmonic frequencies, some of them which are even below the fundamental frequency. Here is another,
very extreme example to demonstrate the result of aliasing. Choose a "sine" shape and activate 16x
upsampling. Now use a distortion or some saturation to process the signal. It is very probable that you will be
able to hear (or at least see in the analyzer) the aliased frequencies.

The plugin implements a high-quality upsampling algorithm, which essentially works like this: First the audio
material is upsampled to a higher sampling rate using a very complicated filter. It is then processed by the
plugin. Further filtering is performed in order to remove any frequencies above the Nyquist rate to prevent

aliasing from occurring, and then the audio gets downsampled to the original sampling rate.

Upsampling also has several disadvantages of which you should be aware before you start using it.
Firstly, upsampled processing induces latency (at least in high-quality mode, although you can select low-quality
mode in the plugin settings), which is not very usable in real time applications. Secondly, upsampling also takes
much more CPU power, due to both the processing being performed at a higher sampling rate (for 16x
upsampling at 44100 Hz, this equates to 706 kHz!), and the complex filtering. Finally, and most importantly,
upsampling creates some artifacts of its own and for some algorithms processing at higher sampling rates can
actually lower the audio quality, or at least change the sound character. Your ears should always be the final
judge.

As always, use this feature ONLY if you can actually hear the difference. It is a common misconception that
upsampling is a miraculous cure all that makes your audio sound better. That is absolutely not the case. Ideally,
you should work in a higher sampling rate (96kHz is almost always enough), while limiting the use of
upsampling to some heavily distorting processors.

Channel mode button

Channel mode button shows the current processing channel mode, e.g. Left+Right (L+R) indicates the
processing of left and right channels. This is the default mode for mono and stereo audio material and
effectively processes the incoming signal as expected. However the plugin also provides additional modes, of
which you may take advantage as described below. Mastering this feature will give you unbelievable options for
controlling the stereo field.

Note that this is not relevant for mono audio tracks, because the host supplies only one input and output
channel.

Left (L) mode and Right (R) mode allow the plugin to process just one channel, only the left or only the

right. This feature has a number of simple uses. Equalizing only one channel allows you to fix spectral
inconsistencies, when mids are lower in one channel for example. A kind of stereo expander can be produced by
equalizing each side differently. Stereo expansion could also be produced by using a modulation effect, such as
a vibrato or flanger, on one of these channels. Note however that the results would not be fully mono
compatible.

Left and right channels can be processed separately with different settings, by creating two instances of the
plugin in series, one set to 'L' mode and the other to 'R' mode. The instance in 'L' mode will not touch the right
channel and vice versa. This approach is perfectly safe and is even advantageous, as both sides can be
configured completely independently with both settings visible next to each other.

Mid (M) mode allows the plugin to process the so-called mid (or mono) signal. Any stereo signal can be

transformed from left and right, to mid and side, and back again, with minimal CPU usage and no loss of audio
quality. The mid channel contains the mono sum (or centre), which is the signal present in both left and right
channels (in phase). The side channel contains the difference between the left and right channels, which is the
"stereo" part. In 'M mode' the plugin performs the conversion into mid and side channels, processes mid, leaves
side intact and converts the results back into the left and right channels expected by the host.

To understand what a mid signal is, consider using a simple gain feature, available in many plugins. Setting the
plugin to M mode and decreasing gain, will actually lower or attenuate the mono content and the signal will
appear "wider". There must be some stereo content present, this will not work for monophonic audio material
placed in stereo tracks of course. Similarly amplifying the mono content by increasing the gain, will make the
mono content dominant and the stereo image will become "narrower".

As well as a simple gain control there are various creative uses for this channel mode.
Using a compressor on the mid channel can widen the stereo image, because in louder parts the mid part gets
attenuated and the stereo becomes more prominent. This is a good trick to make the listener focus on an
instrument whenever it is louder, because a wider stereo image makes the listener feel that the origin of the
sound is closer to, or even around them.
A reverb on the mid part makes the room appear thin and distant. It is a good way to make the track wide due
to the existing stereo content, yet spacey and centered at the same time. Note that since this effect does not

occur naturally, the result may sound artificial on its own, however it may help you fit a dominant track into a
mix.
An equalizer gives many possibilities - for example, the removal of frequencies that are colliding with those on
another track. By processing only the mid channel you can keep the problematic frequencies in the stereo
channel. This way it is possible to actually fit both tracks into the same part of the spectrum - one occupying the
mid (centre) part of the signal, physically appearing further away from the listener, the other occupying the side
part of the signal, appearing closer to the listener.
Using various modulation effects can vary the mid signal, to make the stereo signal less correlated. This
creates a wider stereo image and makes the audio appear closer to the listener.

Side (S) mode is complementary to M mode, and allows processing of only the side (stereo) part of the signal

leaving the mid intact. The same techniques as described for M mode can also be applied here, giving the
opposite results.
Using a gain control with positive gain will increase the width of the stereo image.
A compressor can attenuate the side part in louder sections making it more monophonic and centered, placing
the origin a little further away and in front of the listener.
A reverb may extend the stereo width and provide some natural space without affecting the mid content. This
creates an interesting side-effect - the reverb gets completely cancelled out when played on a monophonic
device (on a mono radio for example). With stereo processing you have much more space to place different
sounds in the mix. However when the audio is played on a monophonic system it becomes too crowded,
because what was originally in two channels is now in just one and mono has a very limited capability for 2D
placement. Therefore getting rid of the reverb in mono may be advantageous, because it frees some space for
other instruments.
An equalizer can amplify some frequencies in the stereo content making them more apparent and since they
psycho acoustically become closer to the listener, the listener will be focused on them. Conversely, frequencies
can be removed to free space for other instruments in stereo.
A saturator / exciter may make the stereo richer and more appealing by creating higher harmonics without
affecting the mid channel, which could otherwise become crowded.
Modulation effects can achieve the same results as in mid mode, but this will vary a lot depending on the
effect and the audio material. It can be used in a wide variety of creative ways.

Mid+Side (M+S) lets the plugin process both mid and side channels together using the same settings. In

many cases there is no difference to L+R mode, but there are exceptions.
A reverb applied in M+S mode will result in minimal changes to the width of the stereo field (unless it is true-
stereo, in which case mid will affect side and vice versa), it can be used therefore, to add depth without altering
the width.
A compressor in M+S mode can be a little harder to understand. It basically stabilizes the levels of the mid and
side channels. When channel linking is disabled in the compressor, you can expect some variations in the sound
field, because the compressor will attenuate the louder channel (usually the mid), changing the stereo width
depending on the audio level. When channel linking is enabled, a compressor will usually react similarly to the
L+R channel mode.
Exciters or saturators are both nonlinear processors, their outputs depend on the level of the input, so the
dominant channel (usually mid) will be saturated more. This will usually make the stereo image slightly thinner
and can be used as a creative effect.

How to modify mid and side with different settings? The answer is the same as for the L and R channels.
Use two instances of the plugin one after another, one in M mode, the other in S mode. The instance in M mode
will not change the side channel and vice versa.

Left+Right(neg) (L+R-) mode is the same as L+R mode, but the the right channel's phase will be inverted.

This may come in handy if the L and R channels seem out of phase. When used on a normal track, it will force
the channels out of phase. This may sound like an extreme stereo expansion, but is usually extremely fatiguing
on the ears. It is also not mono compatible - on a mono device the track will probably become almost silent.
Therefore be advised to use this only if the channels are actually out of phase or if you have some creative
intent.

There are also 4 subsidiary modes: Left & zero Right (L(R0)), Right & zero Left (R(L0)), Mid & zero

Side (M(S0)) and Side & zero Mid (S(M0)). Each of these processes one channel and silences the other.

Surround mode is not related to stereo processing but lets the plugin process as many channels as the host

supplies (up to 8). To use it, you have to first activate surround processing, by selecting the menu item. This is
a global switch for all MeldaProduction plugins, which configures them to report 8in-8out capabilities to the host,
on loading. It is disabled by default, because some hosts have trouble dealing with such plugins. After
activation, restart your host to start using the surround capabilities of the plugins. Deactivation is done in the
same way. Please note that the sidechain inputs will be multi-channel too

First place them on a surround track - a track that has more than 2 channels. Then select Surround from the
plug-in's Channel Mode menu. The plugins will regard this mode as a natural extension of 2 channel processing.
For example, a compressor will process each channel separately or measure the level by combining the levels of
all of the inputs provided. Further surround processing properties, to enable /disable each channel or adjust its
level, can be accessed via the Surround settings in the menu.

AGC button

AGC button enables or disables the automatic gain control - the automatic adjustment of the output volume
such that it matches the input volume. Human hearing is very adaptable. In fact differences in loudness, for
example when loading a preset, may go unnoticed and instead be perceived by the listener as "better
sounding", leading to a misjudgement. This feature should prevent this effect, thus allowing the listener to focus
on the sonic qualities only.

AGC works by measuring input and output loudness, and then compensating for the difference while also taking
into account any induced latency. The loudness measurement follows the ITU and EBU specifications with an
RMS of 400ms, meaning that the reaction time is 400ms. This is very important, as you should be aware that
AGC needs time to properly adjust after any change of settings. Also note that this is a nonlinear operation. It
may cause some distortion due to the long measurement time. It should be negligible though.

AGC makes sense in most applications including reverberation and equalization for example. However, in some
cases it can work against the plugin. A simple example of this is a tremolo, where the plugin manipulates output
volume. If the tremolo rate is slow enough, say 1Hz, it makes the period longer than the actual AGC
measurement time. So whenever the tremolo changes audio level, the AGC starts compensating for it. This can
of course be used creatively, since AGC will always be a little "late", but it is definitely not a desired outcome in
normal use.

Another example of this is compression. When used with short attack and release times, AGC can effectively
compensate for the attenuation of the compressor. However when the attack and release times are higher than
100ms, the compressor's reaction time becomes too slow, and in conjunction with AGC, severe pumping can
occur.

As a general rule of thumb as for all audio processing tasks, use it only if you know you need it. AGC is a
powerful tool that can make your workflow easier, but it can also be damaging.

Set button

Set button uses the AGC (automatic gain compensation) processor to calculate the ideal output gain to ensure
that the output audio loudness is equal to the input level. To use it, simply enable playback in your host and
click the button. The plugin's output gain will be adjusted to match the input and output levels as closely as
possible.
If the AGC is already enabled, the change will be instant and you can disable the AGC afterwards. Typically you
will browse presets, generate random settings etc. During the entire time you will have AGC enabled to prevent
you from experiencing different output loudness levels. When you find a sonically ideal setup, you simply click
the Set button to set the output gain automatically and disable the AGC as you won't need it anymore.
If the AGC is not already enabled, clicking the Set button displays a window with progress bar for a few
seconds, while the plugin temporarily enables AGC and analyses input and output of the plugin. After that the
AGC is disabled again.

To get the best results, you should feed the plugin with some "universal" signal. If you are processing a specific

instrument, play a typical part, a chorus in case of vocals for example. If you are creating presets designed for
general use, white/pink noise may be the best signal to use.

Limiter button

Limiter button enables or disables the safety limiter. Its purpose is to protect you from peaks above 0dB, which
can have damaging effects to your processing chain, your monitors and even your hearing.

It is generally advised to keep your audio below 0dB at all times in all stages of your processing chain. However,
several plugins may cause high level outputs with certain settings, often due to unprevented resonances with
specific audio materials. The safety limiter prevents that.

Note that it is NOT wise to enable this "just in case". As with any processing, the limiter requires additional
processing power and modifies the output signal. It is a transparent single-band brickwall limiter, but you still
need to be careful when using it.

Diff button

Diff button lets you audition the difference between input and output. This is especially useful for dynamic
processors, such as compressors, where you can simply listen to the parts being modified. The output may give
you insight about which parts of the signal are being processed and how.

A-H presets selector

A-H presets selector controls the current A-H preset. This allows the plugin to store up to 8 sets of settings,
including those parameters that cannot be automated or modulated. However it does not include channel mode,
upsampling and potentially some other global controls available from the Settings/Settings menu.

For example, this feature can be used to keep multiple settings, when you are not sure about the ideal
configuration When you change any parameter, only the currently selected preset is modified.

The four buttons below enable you to switch between the last 2 selected sets using the A/B button, morph
between the first 4 sets using the morphing button and copy & paste settings from one preset to another (via
the clipboard).

It is also possible to switch between the presets using MIDI program change messages sent from your host. The
set selected depends on the Program Change number: 0 selects A, 7 selects H, 8 selects A, 15 selects H and so
on.

A/B button

A/B button switches between the active and previously active A-H preset (not necessarily the A and B presets
themselves). To compare any 2 of the A-H presets, select one and then the other. Clicking this button will then
switch between these two. You can do the same thing by clicking on the particular presets, but this makes it
easier, letting you close your eyes and just listen.

Morph button

Morph button lets you morph between the A, B, C and D settings. Morphing only affects those parameters that
can be automated or modulated; that does include most of the parameters however. When you click this button,
an X/Y graph is shown allowing you to drag the position indicator to any position between the letters A, B, C and
D. The closer you drag the indicator to one of the letters, the closer the actual settings are to that preset.

Please note that this will overwrite and change the preset that is currently selected, so it is best to
select a new preset e.g. 'E', then use the morphing method. This way you will define the settings
for A, B,C and D, morph between them, and store the result in 'E' without any modification of the
original A, B, C and D presets.

Please note that the ABCD morphing itself cannot be automated and that, while morphing, the changes to the
underlying parameters are not notified to the host (there may be hundreds of change events).

Copy button

Copy button copies the current settings to the system clipboard. Other presets, upsampling, channel mode and
other global settings are not copied.
Hold Ctrl to save the settings as a file instead. That may be necessary for complex settings, which may be too
long for system clipboard to handle. It may also be advantageous when you want to send the settings via email.
You can load the settings by drag & dropping them to a plugin or holding Ctrl and clicking Paste.

Paste button

Paste button pastes settings from the system clipboard into the current preset. Hold Ctrl to load the settings
from a file instead. Hold Shift to paste the settings to all of the A-H slots at once.

Undo button

Undo button reverts the last change. Only changes to automatable or modulatable parameters and global
settings (load/randomize) are stored.

Redo button

Redo button reverts the last undo operation.

WAV button

WAV button lets you process a file using the plugin with current settings. You can either click the button and
select a file, or drag & drop the file (or multiple files) onto the button. If you let the plugin process WAV files,
these will be saved with the original settings. If you use a different file type (such as MP3), the plugin will create
WAV files with 32-bit bits-per-sample floating point.

Please note that the files will be overwritten, so make a copy first if you want to keep the original.

Used controls

Here we discuss the general properties of all application controls. As a most important rule you should note, that
you can always use any question mark button or F1 (or Ctrl+F1 or Ctrl+H) key with the mouse cursor over a
specified control to get detailed information about what it does and how to use it.

Zoomer

Zoomer provides a simple way to zoom and move in an enlargeable view.



Plus button zooms-in.



Minus button zooms-out.



Zoom default button zooms to the default ratio, which typically means full zoom-out.



Lock button locks the zoom ratio.

Graph editor

Graph editor will show and edit one or more graphs.



Zoomers below and on the right control the zoom amount and position of the view.



Mouse wheel zooms in or out. Alternatively you can zoom in using Alt + right button double click

and out using Alt + left button double click. You can also use keyboard numbers 0 to 9 to quickly
set the zoom level.



Drag a rectangle using the left mouse button while holding Alt zooms into the selected

rectangle if possible.



Drag using the left mouse button while holding Alt and Ctrl to scroll the view. This is not

possible when zoomed all the way out as there is nothing to scroll.

Knob

Knob is an alternative to a tracker, which simulates physical knobs.



Click and drag using the left mouse button to change the value.



Right mouse button selects the default value.



Mouse wheel, arrow keys and vertical drag using middle mouse button or using left mouse
button while holding Ctrl modifies the value more precisely.



Home key configures the minimal possible value, conversely end key setups the maximal one.



Esc or Backspace keys restore the original value when either one is pressed during dragging.



Shift + left mouse button or double-click using left mouse button lets you edit the value as text.

You can use the virtual keyboard or type on your computer keyboard. In some cases this shows a menu
with all possible values instead.



Alt + press then release measures the time between the press and the release and applies it as

time/frequency tap. Usable only for certain values of course.

Switcher

Switcher is an alternative to a tracker or knob control, but it has a limited set of values.



Left mouse button shows a menu with list of all possible values. This function might be unavailable in

certain cases when the number of possible values is too high.



Right mouse button selects the default value.



Up and Down arrow keys, buttons in the control and mouse-wheel increase or decrease the value.

Installation, activation,

introduction to audio plugins

Installation

All MeldaProduction plugins are currently available for Windows and Mac OS X operating systems, both 32-bit
and 64-bit versions. You can download all software directly from our website. Since the installation procedures
for the two operating systems are quite different, we will cover each one separately.

The download files for the effects include all the effects plug-ins and MPowerSynth. During the installation
process you can select which plug-ins or bundles to install. If you have not licensed all of the plugins in a bundle
then you just need to activate each plugin separately.

If you have multiple user accounts on your computer, always install the software under your own account! If
you install it under one account and run it under a different one, it may not have access to all the resources
(presets for example) or may not even be able to start.

Installation on Windows

All plugins are available for VST, VST3 and AAX interfaces. The installer automatically installs both the 32-bit and
64-bit versions of the plugins.

Note: Always use 32-bit plugins in 32-bit hosts, or 64-bit plugins in 64-bit hosts. 64-bit plugins
cannot work in 32-bit hosts even if the operating system is 64-bit. Conversely, never use 32-bit
plugins in 64-bit hosts. Otherwise they would have to be 'bridged' and, in some hosts, can become
highly unstable.

You can select the destination VST plugins paths on your system. The installer will try to detect your path,
however you should check that the correct path has been selected and change it if necessary. In all cases it is
highly recommended to use the current standard paths to avoid any installation issues:
32-bit Windows:
C:\Program files\VstPlugins

64-bit Windows:
C:\Program files (x86)\VstPlugins
C:\Program files\VstPlugins

If your host provides both VST and VST3 interfaces, VST3 is usually preferable. If a plugin cannot be opened in
your host, ensure the plugin file exists in your VST plugin path and that if your host is 32-bit, the plugin is also
32-bit, and vice versa. If you experience any issues, contact our support via [e-mail]

(for 32-bit plugins)

(for 64-bit plugins)

Installation on Mac OS X

All plugins are available for VST, VST3, AU and AAX interfaces. Installers create both 32-bit and 64-bit versions
of the plugins.

If your host provides multiple plugin interface options, VST3 is usually preferable. If you experience any issues,
contact our support via [e-mail]

Most major hosts such as Cubase or Logic should work without problems. In some other hosts the keyboard
input may be partly non-functional. In that case you need to use the virtual keyboard available for every text
input field. You may also experience various minor graphical glitches, especially during resizing plugin windows.
This unfortunately cannot be avoided since it is caused by disorder in Mac OS X.

Uninstallation on Windows

The Uninstaller is available from the Start menu and Control panel, in the same way as for other applications. If
you don't have any of these for any reason, go to Program files / MeldaProduction / MAudioPlugins and run
setup.exe.

Uninstallation on OSX

The Uninstaller is available from Applications / MeldaProduction / MAudioPlugins / setup.app.

Performance precautions

In order to maximize performance of your computer and minimize CPU usage it is necessary to follow a few
precautions. The most important thing is to keep your buffer sizes (latency) as high as possible. There is
generally no reason to use latency under 256 samples for 44kHz sampling rates (hence 512 for 96kHz etc.).
Increasing buffer sizes (hence also latency) highly decreases required CPU power. In rare cases increasing
buffer sizes may actually increase CPU power, in which case you can assume your audio interface driver is
malfunctioning.

You should also consider using only necessary features. Usually the most CPU demanding features are
upsampling and modulation of certain parameters. You can reduce modulation CPU usage at the cost of lower
audio quality in Settings/Settings/Modulator protection.

Troubleshooting

The plugins are generally very stable, there are known problems however.

GPU compatibility

The software uses hardware acceleration to move some of the processing (mainly GUI related) from your CPU
(processor) to your GPU (graphics processing unit). It is highly recommended to use a new GPU, as it will
provide higher performance improvements, and update your GPU drivers. Older GPUs are slower and may not
even provide required features, so the software will have to perform all calculations in the main CPU. We also

have had extremely bad experiences with GPUs from ATI and despite the fact that software is now probably
bulletproof, it is recommended to use NVidia GPUs as there has not been a single case of a problem with them.

If you experience problems with your GPU (crashing, blank/dysfunctional GUI), and that you cannot disable the
GPU acceleration from the plugin's Settings window itself, download this file:

http://www.meldaproduction.com/download/GPU.zip

And place the GPU.xml included in the zip into

Windows: C:\Users\{username}\AppData\Roaming\MeldaProduction
Mac OS X: ~/Library/Application support/MeldaProduction

Memory limits of 32-bit platform

Most hosts are now 64-bit ready, however some of them are not or users willingly choose 32-bit edition,
because the required plugins are not 64-bit ready yet. All our software is 64-bit ready. Please note that you
must NOT use the 64-bit plugins in 32-bit hosts, even if you have a bridge. If you are stuck with a 32-bit host
for any reason, note that there is a memory limit (about 1.5 GB), which you may not exceed. This can happen if
you load too many samples or different plugins for example. In that case the host may crash. There is no other
solution than to use a 64-bit host.

Updating

You can use "Home/Check for updates" feature in any of the plugins. This will check online if there is a newer
version available and open the download page if necessary.

To install a newer (or even older) version you simply need to download the newest installer and use it. There is
no need to uninstall the previous version, the installer will do that if necessary. You also do not need to worry
about your presets when using the installer. Of course, frequent backup of your work is recommended as usual.

Using touch-screen displays

Touch screen displays are supported on Windows 8 and newer and the GUI has been tweaked to provide a good
workflow. Up to 16 connections/fingers/inputs are supported. Any input device such as touch-screens, mouse,
tablets are supported. These are the main gestures used by the plugins:

- Tap = left click

- Double tap = double click

- Tap & hold and quickly tap next to it with another finger = right click. Tap & hold is a classic right-click
gesture, however that doesn't provide a good workflow, so came up with this method, which is much faster and
does not collide with functionality of some elements.

Purchasing and activation

You can purchase the plugin from our website or any reseller, however purchasing directly from our website is
always the quickest and simplest option. The software is available online only, purchasing is automatic, easy and
instant. After the purchase you will immediately receive a keyfile via email. If you do not receive an e-mail
within a few minutes after your purchase, firstly check your spam folder and if the email is not present there,
contact our support team using [e-mail] so we can send you the licence again.

To activate the software simply drag & drop the licence file onto the plugin. Unfortunately some hosts
(especially on Mac OS X) either do not allow drag & drop, or make it just too clumsy, so you can use
Home/Activate in any of the plugins and follow the instructions. For more information about activation please
check the online video tutorial.

You are allowed to use the software on all your machines, but only you are allowed to operate the software.
The licences are "to-person" as defined in the licence terms, therefore you can use the software on all your
computers, but you are the only person allowed to operate them. MeldaProduction can provide a specialized
licence for facilities such as schools with different licence terms.

Quick start with your host

In most cases your host will be able to recognize the plugin and be able to open it the same way as it opens
other plugins. If it doesn't, ensure you did installation properly as described above and let your host rescan the
plugins.

Cubase

Click on an empty slot (in mixer or in track inserts for example) and a menu with available plugins will be
displayed. VST2 version is located in MeldaProduction subfolder. However VST3 version is recommended and is
located in the correct folder along with Cubase's factory plugins. For example, dynamic processors are available
from the "Dynamics" subfolder.

To route an audio to the plugin's side-chain (if it has one), you need to use the VST3 version. Enable the side­chain using the arrow button in the Cubase's plugin window title. Then you can route any set of tracks into the
plugin's side-chain either by selecting the plugin as the track output or using sends.

To route MIDI to the plugin, simply create a new MIDI track and select the plugin as its output.

Logic

Choose an empty insert slot on one of your audio tracks (or instrument tracks for example) and select the plugin
from the popup menu. You will find it in the Audio Units / MeldaProduction folder.

To route an audio to the plugin's side-chain (if it has one), a side-chain source should be available in the top of
the plugin's window, so simply select the source track you want to send to the plugin's side-chain.

To route MIDI to the plugin, you need to create a new Instrument track, click on the instrument slot and select
the plugin from AU MIDI-controlled Effects / MeldaProduction. The plugin will receive MIDI from that track.
Then route the audio you want to process with the plugin to this track.

Studio One

Find the plugin in the Effects list and drag & drop it onto the track you would like to insert the plugin to.

To route an audio track to the plugin's side-chain (if it has one), first enable the side-chain using the "Side­chain" button in the Studio One's plugin window title. Then you can route any set of tracks into the plugin's
side-chain from the mixer.

To route MIDI to the plugin, simply create a new MIDI track and select the plugin as its output.

Digital performer

In the Mixing Board, find an empty slot in the track you would like to insert the plugin to. Click on the field and
select the plugin from the effects list.

To route an audio track to the plugin's side-chain (if it has one), choose the track you want to send using Side­chain menu, which appears at the top of the DP's plugin window.

To route MIDI to the plugin, simply create a new MIDI track in the Track view and select the plugin as its
output.

Reaper

Click on an empty slot in the mixer and a window with available plugins will be displayed. Select the plugin you
want to open by double clicking on it or using Ok button.

It is highly recommended to select all MeldaProduction plugins in the plugin window the first time you open it,
click using your right mouse button and enable "Save minimal undo states". This will disable the problematic
Undo feature, which could cause glitches whenever you change certain parameters.

To route an audio track to the plugin's side-chain (if it has one), click on I/O button of the side-chain source
track in the mixer. Routing window will appear, there you click "Add new send" and select the track the plugin is
on. In the created send slot select the channels (after the "=>" mark) for the send, in stereo configuration 3/4
for example. Note that this way the whole track receives the side-chain signal and all plugins with it. It is
possible to send it to a single plugin only, but it is more complicated, please check the Reaper's documentation
about that.

To route MIDI to the plugin, create a new MIDI track and do the same thing as with side-chain, except you
don't need to change output channels.

Live

In Session view, select the track you would like to insert the plugin to. At the left top of Ableton Live's interface,
click on the Plug-in Device Browser icon (third icon from the top). From the plug-ins list choose the plugin (from
MeldaProduction folder), double click on it or drag & drop it into the track.

The X/Y grid usually doesn't provide any parameters of the plugin. This is because the plugins have too many of
them, so you have to select them manually. Check Live's documentation for more information.

To route an audio to the plugin's side-chain (if it has one), select the track you want to send to the side-chain
and in the 'Audio To' menu, choose the audio track that has the plugin on it. Then in the box just below that
select the plugin from the menu.

NOTE: Live does NOT support any interface correctly, it doesn't use the reported buses properly, hence it
doesn't work with surround capable plugins. Therefore you need to use VST version, which reports only stereo
capabilities by default.

To route MIDI to the plugin, create a new MIDI track and in the 'MIDI to' menu, choose the audio track that
has the plugin on it. Note that in Live only the first plug-in on any track can receive MIDI.

ProTools

In the mixer click an empty slot to insert the plugin to and select the plugin from the tree. The plugin may be
present multiple times, once for each channel configuration (mono->stereo etc.). As of now ProTools do not
arrange them in the subfolders, which is a workflow dealbreaker, but we cannot do anything about it. The huge
empty space on top of each plugin window, which occupies so much of the precious display area, is part of
ProTools and every plugin window and again we cannot do anything about it. In some cases you may
experience CPU overload messages, in which case please contact Avid for support. Note that ProTools 10 and
newer is supported. RTAS compatibility for PT9 and older will never be added.

To route an audio to the plugin's side-chain (if it has one), open the plugin, click on the
the plugin title and select the bus you want the audio taken from. You might need to remember the bus
number, unless your ProTools version supports bus renaming. ProTools doesn't support stereo (or surround)
side-chains at all.

To route MIDI to the plugin, create a new MIDI track and in the mixer click the output field for that track and
select the plugin, which should already be in the menu.

No key input

button in

FL Studio

First make sure plugins are scanned, either a full scan through the Plugin Manager or an automatic fast scan
when you open the Plugin Database section of the browser in FL. The scanned plugins will show up in the Plugin
Database > Installed section of the FL browser. The Effects and Generators sections in the Plugin Database will
show all "favorite" plugins. These can be checked and unchecked in the Plugin Manager or added in some other
ways. These favorites also show up in the Add menu, the menu for the "+" button in the channel rack, when
you right click an existing channel button to replace or insert, in the plugin slot menu in the mixer and in the
plugin picker (F8). The menus with favorite plugins also have a "More" choice that will show all scanned plugins.
The full explanation is in our help file, on the page Installing Plugins.

To route an audio to the plugin's side-chain, first set up the mixer: make sure the track you want to receive
audio from is sent to the track the plugin as a sidechain (help). Then set up the plugin wrapper: choose the
desired input on the Processing tab of the wrapper options.

To route MIDI notes to the plugin, first configure the sender: choose a MIDI port for the input device in the
MIDI settings (for a hardware device), or an output port in the wrapper options (for a VST plugin that
produces MIDI). For the receiving plugin, set the input port in the wrapper options to the same value you chose
in step 1.

To route MIDI controllers, the procedure is different. The usual method in FL is to link CC messages to plugin
parameters (help file). VST plugins will also have 128 CC parameters published (through the wrapper) that can
be linkes this way. Those will send the specified CC MIDI message to the plugin, instead of changing a
published parameter.

GUI styles, editor modes and colors

MeldaProduction plugins provide a state of the art styling engine, which lets you change the appearance to your
liking. The first time you run the plugins a style wizard will appear and let you choose the style and other
settings. It may not be available in ProTools and other problematic hosts.

By default each plugin has a certain color scheme, which differs based on what kind of plugin is that. Also,
sections of some plugins are colorized differently, again, based on what kind of section is that (this can be
disabled in global settings). Despite you can change the colors anyhow you want, it is advantageous to keep the
defaults as these are standardized and have predefined meaning, so just by looking at a plugin's color you can
immediately say what kind of plugin and section is that. Same rules apply when designing active presets for
easy screens. This is the current set of colors:

Dynamics = orange
Equalization, filtering = green
Reverb, delay = brown/yellow
Modulation = blue
Distortion, limiting = red
Stereo = cyan/yellow
Time, pitch, unison... = purple/pink
Tools = grey

Special colors:
Synchronization = grey
Detection = blue/green
Side-chain = green
Effects = red
Advanced stuff = grey

About MeldaProduction

The best sound on the market, incredible workflow and versatility beyond your imagination. We create the
deepest and the most powerful audio plugins with unbelievable sound and tons of unique features you cannot
find anywhere else.

Innovative Thinking

At MeldaProduction, we make the most advanced tools for music production and audio processing. We get
inspired by the whole range of tools from the ancient analog gear to the newest digital creations, but we always
push forward.
We've always felt the audio industry is extremely conservative, still relying on the prehistoric equipment making
the job unnecessarily slow and complicated. That's why we invent new technologies, which make audio
processing easier, faster, better sounding and more creative.

Sound Matters

In the world full of audiophiles you just need superb audio quality. And that's why we spend so much time
perfecting audio algorithms until they sound unbeatable. Everything from dynamic filters to spectral dynamic
processing. Our technologies just sound perfect.

Inspiring User Interface

Modern user interfaces must not only be easy and quick to use, but also versatile and the whole visual
appearance should inspire you. MeldaProduction plugins provide the most advanced GUI engine on the market.
It is still the first and only GUI engine, which is freely resizable and stylable. Our plugins can look as an ancient
vintage gear, if you are working on old-school rock music. Or as super-modern futuristic devices if you are
working on modern electronic music.

Easy to Use, Yet Versatile

The only limit is your imagination. Our plugins are with absolutely no doubt the most powerful and versatile
tools on the market. Yet we managed to make the plugins easy to use via the active presets and smart
randomization system. But when you are ready, you are one click away from the endless potential the plugins
provide.

Never-Ending Improvements

Most companies create a plugin, sell it and abandon it. We improve our plugins, add features, optimize... until
there is nothing left to improve and there are no more ideas. Unfortunately that hasn't happened yet :). And the
best thing is that the updates are free-for-life!

MeldaProduction was founded in 2009 by Vojtech Meluzin and is based in Prague, Czech Republic.

www.meldaproduction.com
[e-mail]

MeldaProduction (c) 2017