MeldaProduction MWaveFolderMB Distortion Plug in Instruction Manual

MWaveFolderMB Distortion Plug in

Specifications

• Product Name: MWaveFolderMB
• Screen Types: Easy screen, Edit screen
• Modes: Edit mode

Product Usage Instructions

Preset Navigation

To navigate presets, use the left and right arrow buttons:

• Left arrow button: Loads the previous preset
• Right arrow button: Loads the next preset

Randomization

To load a random preset, click on the “Randomize” button. There
are different ways to use the randomization feature:

• Clicking the “Randomize” button will load a random preset with
  default settings.

• Holding Ctrl (Command on Mac) while clicking the “Randomize”
  button will slightly modify the parameters of the current preset,
  creating small variations.

• Holding Alt (Option on Mac) while clicking the “Randomize”
  button will fully randomize all reasonable automatable parameters,
  potentially resulting in extreme settings. Note that some
  parameters cannot be randomized this way.

Panic Button

The Panic button is not mentioned in the given text extract.
Please refer to the user manual for more information on the Panic
button.

Settings Menu

The Settings button opens a menu with additional plugin
settings. Here are the different sections in the menu:

1. Licence Manager: Allows activation/deactivation of plugins and
   management of subscriptions.

2. GUI & Style: Enables customization of GUI style and colors
   used in the plugin.

3. Advanced Settings: Configures various processing options for
   the plugin.

4. Global System Settings: Contains settings that apply to all
   MeldaProduction plugins.

5. Dry/Wet Affects: Determines which multiband parameters are
   affected by the Global dry/wet control.

6. Smart Interpolation: Adjusts the interpolation algorithm used
   when changing parameter values.

WWW Button

The WWW button opens a menu with additional information about
the plugin. Here are some options available in the menu:

• Check for updates
• Access support
• Visit MeldaProduction web page
• Watch video tutorials
• Access Facebook, Twitter, and YouTube channels

Sleep Indicator

The sleep indicator informs whether the plugin is currently
active or in sleep mode. The plugin automatically switches off to
save CPU when there is no input signal and it cannot produce any
signal on its own. You can disable this feature in the Settings
menu.

FAQ (Frequently Asked Questions)

Q: What is the purpose of the Panic button?

A: The Panic button is not mentioned in the given text extract.
Please refer to the user manual for more information on the Panic
button.

Q: How can I customize the GUI style and colors of the
plugin?

A: Click on the Settings button and select GUI & Style. From
there, you can choose the GUI style and customize the colors used
in the plugin.

Q: Can I modify the randomization behavior?

A: Yes, you can modify the randomization behavior by holding
Ctrl (Command on Mac) or Alt (Option on Mac) while clicking the
Randomize button. Holding Ctrl slightly modifies parameters, while
holding Alt fully randomizes automatable parameters.

Q: How can I access support for the plugin?

A: Click on the WWW button and select the support option. It
will provide you with access to support resources.

MWaveFolderMB
Easy screen vs. Edit screen
The plugin provides 2 user interfaces – an easy screen and an edit screen. Use the Edit button to switch between the two. By default most plugins open on the easy screen (edit button released). This screen is a simplified view of the plugin which provides just a few controls. On the left hand side of the plugin you can see the list of available devices / instruments (previously called ‘active presets’), that is, presets with controls. These controls are actually nothing more than multiparameters (single knobs that can control one or more of the plug-in’s parameters and sometimes known as Macro controls in other plug-ins) and are described in more detail later. Each device may provide different controls and usually is intended for a specific purpose. The easy screen is designed for you to be able to perform common tasks, quickly and easily, without the need to use the advanced settings (that is, those available on the Edit screen). In most cases the devices are highlighted using different text colors. In some cases the colors only mark different types of processing, but in most cases the general rule is that black/white devices are the essential ones designed for general use. Green devices are designed for a specific task or audio materials, e.g. de-essing or processing vocals in a compressor plugin. Red devices usually provide some very special processing or some extreme or creative settings. In a distortion plugin, for example, these may produce an extremely distorted output. Blue devices require an additional input, a side-chain or MIDI input usually. Without these additional inputs these Blue presets usually do not function as intended. Please check your host’s documentation about routing side-chain and MIDI into an effect plugin. To the right of the controls are the meters or time-graphs for the plugin; the standard plugin Toolbar may be to the right of these or at the bottom of the plugin. By clicking the Edit button you can switch the plugin to edit mode (edit button pushed). This mode provides all the of the features that the plugin offers. You lose no settings by toggling between edit mode and the easy screen unless you actually change something. This way you can easily check what is “under the hood” for each device, or start with an device and then tweak the plugin settings further. Devices are factory specified and cannot be modified directly by users, however you can still make your own and store them as normal presets. To do so, configure the plugin as desired, then define each multiparameter and specify its name in its settings. You can then switch to the easy screen and check the user interface that you have created. Once you are satisfied with it, save it as a normal preset while you are on the easy screen. Although your preset will not be displayed or selected in the list of available devices, the functionality will be exactly the same. For more information about multiparameters and devices please check the online video tutorials. If you are an advanced designer, you can also view both the easy and edit screens at the same time. To do that, hold Ctrl key and press the Edit button.

Edit mode
Presets
Presets button shows a window with all available presets. A preset can be loaded from the preset window by double-clicking on it, selecting via the buttons or by using 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. The arrow buttons next to the preset button can be used to switch between presets easily. Holding Ctrl while pressing the button loads a random preset. There must be some presets for this feature to work of course. Presets can be backed up by 3 different methods: A) Using “Backup” and “Restore” buttons in each preset window, which produces a single archive of all presets on the computer. B) Using “Export/Import” buttons, which export a single folder of presets for one plugin. C) By saving the actual preset files, which are found in the following directories (not recommended): Windows: C:Users{username}AppDataRoamingMeldaProduction Mac OS X: /Library/Application support/MeldaProduction Files are named based on the name of the plugin like this: “{pluginname}.presets”, so for example MAutopan.presets or MDynamics.presets. If the directory cannot be found on your computer for some reason, you can just search for the particular file. Please note that prior to version 16 a different format was used and the naming was “{pluginname}presets.xml”. The plugin also supports an online preset exchange. If the computer is connected to the internet, the plugin connects to our server once a week, submits your presets and downloads new ones if available. This feature is manually maintained in order to remove generally unusable presets, so it may take some time before any submitted presets become available. This feature relies on each user so we strongly advise that any submitted presets be named and organised in the same way as the factory presets, otherwise they will be removed.
Left arrow
Left arrow button loads the previous preset.
Right arrow
Right arrow button loads the next preset.
Randomize
Randomize button loads a random preset.
Randomize
Randomize button (with the text ‘Random’) generates random settings. Generally, randomization in plug-ins works by selecting random values for all parameters, but rarely achieves satisfactory results, as the more parameters that change the more likely one will cause an unwanted effect. Our plugins employ a smart randomization engine that learns which settings are suitable for randomization (using the existing presets) and so is much more likely to create successful changes.

In addition, there are some mouse modifiers that assist this process. The smart randomization engine is used by default if no modifier keys are held.
Holding Ctrl while clicking the button constrains the randomization engine so that parameters are only modified slightly rather than completely randomized. This is suitable to create small variations of existing interesting settings.
Holding Alt while clicking the button will force the engine to use full randomization, which sets random values for all reasonable automatable parameters. This can often result in “extreme” settings. Please note that some parameters cannot be randomized this way.
Panic
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
Settings button shows a menu with additional settings of the plugin. Here is a brief description of the separate items.
Licence manager lets you activate/deactivate the plugins and manage subscriptions. While you can simply drag & drop a licence file onto the plugin, in some cases there may be a faster way. For instance, you can enter your user account name and password and the plugin will do all the activating for you.
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.
Global system settings contains some settings for all MeldaProduction plugins. Once you change any of them, restart your DAW if needed, and it will affect all MeldaProduction plugins.
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
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.
Sleep indicator
Sleep indicator informs whether the plugin is currently active or in sleep mode. The plugin can automatically switch itself off to save CPU, when there is no input signal and the plugin knows it cannot produce any signal on its own and it generally makes sense. You can disable this in Settings / Intelligent sleep on silence both for individual instances and globally for all plugins on the system.
Plugin toolbar

Plugin toolbar provides some global features, A-H presets and more.
Oversampling
Oversampling 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. 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 oversampling 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 oversampling even they become highly aliased. Now select 16x oversampling and listen to the difference. If you again select 1x oversampling, 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 oversampling, 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 oversampling. 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 oversampling 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.
Oversampling 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 directly in this popup), which is not very usable in real time applications. Secondly, oversampling also takes much more CPU power, due to both the processing being performed at a higher sampling rate (for 16x oversampling at 44100 Hz, this equates to 706 kHz!), and the complex filtering. Finally, and most importantly, oversampling 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 oversampling 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 oversampling to some heavily distorting processors.
Channel mode
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 up to 8 channels, depending on how many the host supplies. For VST2 plugins you have to first activate surround processing using the Activate surround item in the bottom. 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 all input and output busses will be multi-channel, that includes side- chain for example. For VST3/AU/AAX plugins the activation is not necessary.
First place the plugin 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.
Ambisonics mode provides support for the modern 3D systems (mostly cinema and VR) with up to 64 channels (ambisonics 7th order). Support for this is still quite rare among the DAWs, so this needs to be activated in all DAWs using the Activate ambisonics item in the bottom. This is a global switch for all MeldaProduction plugins, which configures them to report 64in-64out capabilities to the host, on loading. After activation, restart your host to start using the ambisonics capabilities of the plugins. Deactivation is done in the same way. Please note that all input and output busses will be multi- channel, that includes side-chain for example.
First place the plugin on an ambisonics track, supported are all orders from 1st (4 channels) to 7th (64 channels). Then select Ambisonics from the plug- in’s Channel Mode menu. Finally select the Ambisonics settings in the menu and configure the Ambisonics order and other settings if needed. 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.
AGC
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
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
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.
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, oversampling 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
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

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
Copy button copies the current settings to the system clipboard. Other presets, oversampling, 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
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
Undo button reverts the last change. Only changes to automatable or modulatable parameters and global settings (load/randomize) are stored.
Redo
Redo button reverts the last undo operation.
WAV
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.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.
Band editor
Band editor displays the available frequency bands, the crossover frequencies delimiting them, and the input gains and panoramic positions. Use the left mouse button to drag the band boundaries (the vertical lines between bands), the band itself (the central dot in each band) and the input gains (the horizontal bars in each band). The short vertical bars in the bottom of each band control the input panoramic positions (when L+R Channel Mode is selected) or the input Widths (when M+S Channel Mode is selected). Use the right mouse button to open the Band Configuration window where you can manage the bands and crossover filters and the appearance of the analyzer waveforms in the band editor. Buttons to the left-hand side of each band let you mute, solo and bypass the processing in each band. Please note that the Mute and Solo buttons act on the output for each band, that is after the actual band processing. The Collapse button to the right of the Band Editor minimises the editor, releasing space for other editors in the plug-in.

Band menu
Band menu provides features to control the set of bands and copy & paste band settings (Band management section), reset band input gain & panorama (Band gain & panorama section), and to select and customize the crossover (Crossover section) and analyzer options. You can display this menu by right-clicking on the band editor. One of the essential things to control in the band menu is the number of bands. The plugin can either operate as a single bundle plugin. In this case there is no crossover employed of any kind and the first and only band receives all MIDI data if the plugin makes use of it somehow. If there are 2 or more bands however, the plugin somehow produces signals for each band using the crossover, based on the spectrum or level for example, and there’s a change in MIDI behaviour as well – 1st band receives only MIDI channel 1, 2nd receives only MIDI channel 2 etc.
Band management panel
Band management panel contains basic features to create, delete and manipulate bands.
Insert left
Insert left button inserts a new band to the left of the currently-selected band (the last one that you clicked on).
Insert right
Insert right button inserts a new band to the right of the currently-selected band (the last one that you clicked on).
Delete
Delete button deletes the currently-selected band (the last one that you clicked on).

Expand band
Expand band button soloes (or unsoloes) the band that you clicked on and disables the crossover temporarily, so that you can audition what the settings of this band would do to the entire signal, without any of the other bands having any affect.
Auto-set limits by analyzer
Auto-set limits by analyzer button adjusts the band limits using the current analyzer state, so that there’s approximately the same signal level in each band. It is often useful to increase the averaging in the analyzer settings, so that the analysis doesn’t ‘jump’ that quickly.
Create default bands panel
Create default bands panel lets you easily create a predefined set of bands. This is the easiest way to say create default plugin settings with 4 bands.
Clipboard panel
Clipboard panel contains features to transfer band settings via the system clipboard. Note that as always you can paste the settings as text into an email or forum post for example.
Copy
Copy button copies the band settings into the system clipboard. Note that the plugin band parameter settings are not copied; only the band limits, gains and panoramas.
Paste
Paste button loads the band settings from the system clipboard. Note that the plugin band parameter settings are not pasted; only band limits, gains and panoramas.
Reset panel
Reset panel lets you reset some band parameters.
Gain
Gain button resets the input gain of the currently-selected band (the last one that you clicked on) to 0dB.

Gain (all bands)
Gain (all bands) button resets the input gain of all bands to 0dB.
Panorama
Panorama button resets the input panorama of the currently-selected band (the last one that you clicked on) to center.
Panorama (all bands)
Panorama (all bands) button resets the input panorama of all bands to center.
Crossover panel
Crossover panel contains configuration of the crossover used to separate the signals for each band. Crossover is a technical term for an algorithm or device which splits a signal into multiple bands (or signals), which when mixed back together recreate the original signal (meaning that the crossover is transparent). The plugin provides several types of crossover with a flat (or nearly flat) response, which means that whichever crossover you choose and whatever signal you send into the plugin, the output levels of each frequency, after the bands are mixed back together to get the output signal, will be (almost) exactly the same, unless there is some processing applied in the bands themselves. Most of the available crossover types produce bands with different frequency ranges; however there are also a few more creative ones. Analog crossovers have no latency, but they exhibit a phase-shift. That is usually irrelevant unless you are going to mix the output with the input later on. Analog crossovers are based on the classic analog components that you can find in speaker systems for example, however they are perfectly accurate and their slope (band separation) ranges from 6dB/octave to a very steep 120dB/octave. The higher the slope is, the more separated is each band (that is, there is less overlap between bands), but also the bigger is the phase shift. That can reach such an extent that some bassy materials become severely phasey, which may or may not be a good thing. An exception to the rule is the 6dB/oct crossover, which is zero-phase naturally. Its disadvantage is that the separation between bands is rather low, 6dB/oct is often not enough. Analog LP crossover is a linear-phase equivalent to the Analog crossover. It introduces latency as does any linear-phase filter, but it does not cause a phase-shift. This may be especially advantageous for higher filter slopes, which, with classic analog crossovers, would cause severe transient smearing. Please note that the crossover type may not be 100% transparent, especially with small bands in bass spectrum and high slopes. Linear-phase crossover is a fully digital crossover with a high slope (frequency-dependent), which introduces latency, but

exhibits no phase-shift. This crossover mode is designed specifically for mastering.
Hybrid crossover is linear-phase as well, hence it introduces latency, but no phase-shift. However, its slope is more similar to the slopes of the analog crossovers.
Level crossover is a very specialized tool, which doesn’t filter the input signal at all (hence it is not only linear-phase, but also zero-phase). Instead of filtering, it simply performs a gain on each band in such a way that when all the bands are mixed back together, the output is the original signal again. When you select this crossover, the spectrum analyzer graph disappears and the X axis in the band editor changes from frequencies to dB levels. So the band limits are not frequencies anymore, but rather sound levels.
The current level displayed in the graph area is controlled by the Level value below and you are likely to use a modulator, most likely in Follower mode, to control this latter value. The crossover then applies gain to each band depending on how much the current level fits into the band. The Slope parameter controls how quickly each band fades into the adjacent one. This crossover effectively turns the plugin into a very advanced dynamics processor; using a Follower Modulator the band used to process the input audio depends on the audio level. The are many possibilities for this crossover. But the basic principle is to select a spare Modulator, configure it as a Follower and select the Global parameter “Crossover Level value” as its target, with a “Full range” range mode. After configuring the Modulator, you will be able to see the detected value curve in the Modulator’s Level graph. Then if the input signal is strongest, the right most band is processed etc. So if you for example use a delay with 2 bands and set the band limit high enough, the 2nd band will be processing only the loud parts of the signal and vice versa.
Panorama crossover is another specialized tool, similar to the level crossover; it splits the signal into bands according to the panorama. If, for example, you create 3 evenly spaced bands (100%L to 33%L, 33%L to 33%R, 33%R to 100%R), then the leftmost band will contain mainly the signals located in the left speaker, the rightmost band will contain mainly signals from the right speaker and the middle band will contain centred signals. Please note that this doesn’t mean the crossover attempts to analyze the space the signals are coming from and send them to the respective bands, which is probably what your brain would attempt.
This crossover is useful only when processing stereophonic (or surround, in which case the channels from 3 upwards are kept intact) signals and can be used for all kinds of mixing and creative processing. For example, using a multiband compressor with this crossover can be used to effectively control the stereo image as each band would be processing a different part of the stereo image. To mention another example, a multiband delay or reverb can be used to produce a different ambience for different parts of the stereo image.
Mid/side crossover is similar to panorama crossover, but it splits the signal according to their position in mid/side location. In other words, the more to the left a band, the more centred is the signal in it. Similarly the more to the right a band, the more “to the side” is the signal in it. You can think of it as the panorama view folded back on itself, around the center position. If, for example, you create 3 evenly spaced bands (centre to 33% L or R, 33% L or R to 67% L or R, 67% L or R to 100% L or R), then the leftmost band will contain the centred signal, the rightmost band will contain the signals to the extreme left or right and the middle band will contain signals in between. It can be used for similar tasks as the panorama crossover.
Parallel crossover is not a crossover actually, it simply disables the crossover and as a result each band processes the full input signal. In practice this “not really crossover” mode lets you process multiple streams of the input audio signal in parallel. As a consequence there is likely to be an increase in output level, so take care and turn down the output level first. For example, if you use a compressor, this in effect produces an extreme parallel compression. As another example, you can use a reverb to produce several rooms in parallel, potentially leading to a fuller space for example.
Spectrum crossover is the first of the spectral crossovers. It splits the signal into individual frequencies, analyzes their levels and sends the frequencies with the highest level into the highest band etc. It marks each frequency with its level (as you can see on the dB scale on the X axis in the crossover band editor) and puts it into the appropriate band. The crossover is linearphase and fully transparent.
It provides a huge (not only) creative potential as it lets you process the dominant and weak parts of the signal individually. For instance, by compressing the dominant frequencies using MDynamicsMB you can bring more attention to the unsubstantial frequencies in the signal and in a way stabilize it without disrupting the silent parts of it. Note that this is NOT the same thing as using a normal compressor, because this way it treats only the loud frequencies even if the weak frequencies are present at the same time. Another example could be using MDelayMB to generate echoes only to the dominant parts of the signal, such as snare and bass drums in a drum loop.
Transient crossover is also a spectral crossover. It splits the signal into individual frequencies and sends the transient parts for each of them into the highest band etc. It marks each frequency with its “current transientness” (defined by the percentage scale that you can see on the X axis in the crossover band editor) and puts it into the appropriate band. The crossover is linearphase and fully transparent.
It provides a huge (not only) creative potential as it lets you process split the signal into tonal and transient parts (and anything in between) and treat each individually. For instance, by compressing the transients using MDynamicsMB you can easily control the attack of drums. Note that this is NOT the same thing as using a normal compressor, because this way you can treat only the attacks in an already mixed signal without affecting the remaining part of the signal. Another example could be using MDelayMB to generate echoes only for the attacks of each drum.
Serial crossover is not a crossover actually, it simply disables the crossover and processes all bands in series. For instance a multiband compressor can be exploited to perform multiple compressions in series, which is often considered better sounding compared to a single compressor driven hard. Please note that if each band has a latency, the latencies will add up.

Slope
Slope defines the slope of each band transition and is used only by analog crossovers (including the linear-phase versions). It essentially controls the separation between the bands – the higher the slope, the lower the overlap between bands. Higher slopes require more CPU power and exhibit higher phase shift, which may be a problem especially when percussive materials. In these cases it may be necessary to switch to a linear-phase version. Interesting exception to the classic rule are the 6dB/oct crossovers, which are linear- phase by nature (while still being zero latency), because the bands compensate for each other’s phase shift. A side-effect of this is that the signal level in each band is much higher than using other crossovers, so you may expect these crossovers sound considerably different to the other modes.
Level value
Level value is used only with Level crossover and controls the level at which the signal is split into each band. You will probably want to attach this parameter to a modulator in Follower mode for instance.
Level slope
Level slope is used only with some crossover modes (Level, Spectrum and Tonal/Transient) and controls how quickly each band fades into the next one. It’s similar to the Slope parameter used with analog crossovers.
Transient release
Transient release is only used by the Tonal/Transient crossover and controls the release time of each transient. The transients detected by the crossover are naturally very short, so this provides a way to make them longer, hence send more signal to the higher bands of the crossover (receiving transients) and less to the lower bands (receiving the remaining part of the signal).
Transient resolution
Transient resolution is only used by the Tonal/Transient crossover and controls the behaviour of the spectral transient detector. You can use it to adjust the crossover to your audio material and we would recommend a simple trial-and-error approach.
Smoothing
Smoothing is only used by spectral crossovers and controls how frequencies affect their surroundings. Without smoothing the individual bands may sound a bit artificial, because human brain general dislikes separated frequencies. It usually doesn’t matter unless you audition the bands separately, but sometimes when more “brutal” processing is used on each band, it may become audible, which is where the smoothing can provide a solution at the cost of additional CPU and lower separation between bands, because it naturally makes the frequencies “more alike”.
Tone
Tone is only used by spectral crossovers and controls the spectral slope applied by the detector. It is exactly the same feature as the Slope in analyzers and the crossover uses it to determine how to spread the frequencies between the bands. Higher slope gives more energy to higher frequencies and vice versa. Note that whatever the settings are, the crossover still produces signals that perfectly sum to the original input signal, meaning that it is perfectly tranparent and unless the bands are actually doing something, you won’t be able to hear a difference when changing this parameter.
Spectral resolution
Spectral resolution is only used by spectral crossovers and controls the spectral transformation settings. The higher the value is, the higher FFT size and overlap size is used, and therefore more CPU is usually required as well. Whether higher/lower value is good or not depends on the actual signal, the default 50% should work well with most audio materials. Higher values will generally provide better frequency resolution (usually good for less percussive sounds), lower values will provide better time resolution (usually good for more percussive sounds), eventually it is always about a compromise.
Process side-chain
Process side-chain option makes sure the side-chain is processed by the crossover as well as the main input. If you disable this option, main input will be processed of course, but side-chain will not. This may be handy e.g. in a multiband dynamics processor, which should react to the entire signal, but process each bands individually.
Analyzer panel

Analyzer panel lets you configure the fully featured integrated analyzer and sonogram.
Settings
Settings button shows the settings of the spectrum analyzer and the spectrum sonogram.
Analyzer settings

Presets
Presets button displays a window where you can load and manage available presets. Hold Ctrl when clicking to load a random preset instead.

Left arrow
Left arrow button loads the previous preset.

Right arrow
Right arrow button loads the next preset.

Randomize
Randomize button loads a random preset.

Copy
Copy button copies the settings onto the system clipboard.

Paste
Paste button loads the settings from the system clipboard.
Tab selector switches between subsections.
Main settings panel

Tab selector

Main settings panel contains the most useful settings controlling the analyzer behaviour and view.
View
Freeze
Freeze button stops processing temporarily.
Normalize
Normalize button enables or disables the visual normalization, which makes the loudest frequency be displayed at the top of the analyser area (0dB); it does not normalise the sound. This is very useful for comparing frequency levels, however it does hide the actual level. When comparing 2 spectrums you are usually interested mainly in the frequency level differences. In most cases both audio materials will have different overall levels, which would mean that one of the graphs would be “lower” than the other, making the comparison quite difficult. Normalize fixes this and makes the most prominent frequencies of the spectrum reach the top of the analyzer area (or have the most highlighted color in case of sonogram).
Reset
Reset button resets the analyzer state. This is particularly useful when analyzing infinite average and maximum values.
View type
View type controls the way the spectrum is displayed. By default a smooth curve is presented. This view provides the best resolution and detail, but other modes (1/3 octave, 1 octave) may be easier to read.
Opacity
Opacity controls the opacity of all analyzer graphs.
Rainbow opacity
Rainbow opacity controls the opacity of the rainbow graph, if enabled.

Resolution
Resolution defines the vertical range on the display. The human auditory system has a resolution of about 90dB and the relevant range is usually less than 60dB. However you may want to use a higher resolution to check for technical problems – aliasing, distortion etc.
Analysis
Source mode
Source mode defines which audio stages are to be analyzed. By default both input & output are selected and analyzed. However you may want to analyze only the input, or the output (or the external side-chain, where available, on its own or with the input or output).
Channel mode
Channel mode defines which channels are to be analyzed. By default all channels are merged into a mono sum (Mix mode), which is then analyzed. However you may want to analyze separate channels or display both the left and right channels separately. Please note that if two channels (for example: input & output, or input & side-chain) are displayed at the same time then mix mode is used instead of left & right mode. Similarly, when the plug-in is in Surround mode then Mix mode is used. Also please note that when the plug-in is in one of the Mid / Side modes of operation, then you should read ‘Left’ as ‘Mid’ and ‘Right’ as ‘Side’. Different analyser combinations can, of course, be saved as different named presets.
Decay
Decay controls the speed at which the magnitudes return to the minimum value (silence). It is an alternative to averaging, which affects the speed that the frequencies both gain and lose their magnitudes. With a decay of 0% the magnitude goes to the minimum immediately. With 100% it stays the same forever, so it makes it display the maximum.
Slope
Slope makes the analyser increase the magnitude of higher frequencies, since they are typically lower in energy. 3dB per octave is a typical value, which makes pink noise horizontal as pink noise contains equal energy in each octave. Therefore if you set slope to 3dB, the response would be the same for the FFT and 1/3 octave graphs.
Gain
Gain makes all frequencies change magnitude by the specified amount. This has no meaning when normalization is enabled.
Time resolution
Time resolution improves the time resolution, but lowers the spectral resolution. This is typically useful for more scientific analyses, where the signal is moving quickly and you need to follow its movements quickly. This is often advantageous for sonograms with very high FFT sizes.
Deharmonize
Deharmonize tries to remove harmonics in the content and leave only fundamentals. This may help you find the dominant frequencies in the signal.

Super-resolution mode
Super-resolution mode activates a special processing algorithm, which provides high resolution even in the low frequency spectrum. Using standard FFT algorithms you can increase the FFT size to get better bass resolution, but this also slows down the response. Super-resolution mode keeps the quick response in high frequencies as they are naturally quicker, but also highly enhances the bass spectrum resolution. It requires additional CPU power.
Enable when hidden
Enable when hidden causes the analysis engine to continue processing the signal even when the GUI is hidden. Otherwise the sonogram is stopped, therefore will not be immediately available when the GUI is shown again.
Global normalization
Global normalization makes the normalization work based on the maximum of all graphs visible at the time. This means that the levels between the graphs will stay the same, but the maximum level will be 0dB. This is useful for comparing relative levels. If you disable this, all graphs will be normalized separately and will touch 0dB unless they are silent; and this is useful for comparing spectra.
Advanced panel
Advanced panel contains more advanced settings controlling the scientific parameters of the audio analysis.
Peak detection
Peak detection
Peak detection tries to the remove skirts of separate sinusoids letting you view the frequencies contained in your audio material. This may be handy when performing more scientific analyses.
Peak threshold
Peak threshold defines the level below the maximum which is used for peak detection. You can use this to control which peaks get through and to get rid of small insignificant ones.
Scientific settings
Overlapping
Overlapping makes the analyser perform multiple FFT processing on the same data which results in better precision at the cost of higher CPU impact. With higher overlapping the response also speeds up.
FFT size
FFT size defines FFT processing block size. It basically controls the resolution. However for higher resolution in bass content it is recommended to use super-resolution mode instead as it keeps the quick response in higher frequencies.
Window type

Window type defines the type of window used to pre-process the source samples. This has several consequences for the frequency response, but it is a little scientific parameter. If you do not have specific requirements you can just leave this set to its default.
Analytical smoothing
Analytical smoothing switch activates a more complicated smoothing algorithm, which provides more accurate results, however it may require much more CPU power. Unlike normal smoothing this method doesn’t change the proportions of frequencies with higher magnitudes. It is useful mostly for technical analysis and for most musical signals it is often better to use the default smoothing method.
Logarithmic averaging
Logarithmic averaging switch activates averaging in logarithmic mode, hence decibels. If you disable it, linear averaging will be used.
Graphs panel
Graphs panel contains visual settings for the different graphs that you can show in the analyzer.
Average
Copy analysis
Copy analysis button copies the current state of the analysis into the system clipboard so that you can paste it into another analyzer for comparison. Hold ctrl to export the analysis into a CSV file.
Analyzer Fill and Line Color – All Channels
Analyzer Fill and Line Color – All Channels defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in all channel modes – see channel mode help in the Main Settings tab.

Analyzer Fill and Line Color – Left Channel
Analyzer Fill and Line Color – Left Channel defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in the left channel mode only – see channel mode help in the Main Settings tab.

Peaks
Peaks enables detection of frequencies with the highest magnitudes. Frequencies which are at most 20dB lower than the maximum are displayed, and there may be at most 8 of them. Please note that this feature requires additional CPU power.

Line opacity controls the opacity of the graph outline.

Line opacity

Micro-sonogram
Micro-sonogram displays a small single-state sonogram at the bottom of the graph. This may help you compare relevant frequencies, because it is usually easier to compare colors than graph values.

Line width controls the width of the graph online.

Line width

Fill
Fill makes the sonogram (enabled by Show sonogram) fill the whole area.

Fill opacity controls the opacity of the graph interior fill.

Fill opacity

Average (infinite)

Copy analysis
Copy analysis button copies the current state of the analysis into the system clipboard so that you can paste it into another analyzer for comparison. Hold ctrl to export the analysis into a CSV file.
Analyzer Fill and Line Color – All Channels
Analyzer Fill and Line Color – All Channels defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in all channel modes – see channel mode help in the Main Settings tab.
Analyzer Fill and Line Color – Left Channel
Analyzer Fill and Line Color – Left Channel defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in the left channel mode only – see channel mode help in the Main Settings tab.
Maximum

Copy analysis
Copy analysis button copies the current state of the analysis into the system clipboard so that you can paste it into another analyzer for comparison. Hold ctrl to export the analysis into a CSV file.

Analyzer Fill and Line Color – All Channels
Analyzer Fill and Line Color – All Channels defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in all channel modes – see channel mode help in the Main Settings tab.
Analyzer Fill and Line Color – Left Channel
Analyzer Fill and Line Color – Left Channel defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in the left channel mode only – see channel mode help in the Main Settings tab.
Maximum (infinite)
Copy analysis
Copy analysis button copies the current state of the analysis into the system clipboard so that you can paste it into another analyzer for comparison. Hold ctrl to export the analysis into a CSV file.
Analyzer Fill and Line Color – All Channels
Analyzer Fill and Line Color – All Channels defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in all channel modes – see channel mode help in the Main Settings tab.
Analyzer Fill and Line Color – Left Channel
Analyzer Fill and Line Color – Left Channel defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in the left channel mode only – see channel mode help in the Main Settings tab.
Maximum – Average (infinite)
Copy analysis
Copy analysis button copies the current state of the analysis into the system clipboard so that you can paste it into another analyzer for comparison. Hold ctrl to export the analysis into a CSV file.
Analyzer Fill and Line Color – All Channels
Analyzer Fill and Line Color – All Channels defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in all channel modes – see channel mode help in the Main Settings tab.
Analyzer Fill and Line Color – Left Channel
Analyzer Fill and Line Color – Left Channel defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in the left channel mode only – see channel mode help in the Main Settings tab.

Comparison
Paste analysis
Paste analysis button pastes an analysis from the system clipboard and displays it as a comparison. This way you can compare your analysis to any other analysis from MeldaProduction plugins.
Analyzer Fill and Line Color – Left Channel
Analyzer Fill and Line Color – Left Channel defines the fill and line colors of the analyzer graph when Rainbow analyzer colors are disabled. This setting is used to set colors in the left channel mode only – see channel mode help in the Main Settings tab.
Sonogram panel
Sonogram panel contains visual settings of the sonogram, mainly the sonogram colors. A sonogram uses a set of colors. When the particular frequency’s level is at the minimum, the first color is used. When it is at the maximum, the last color is used. Otherwise it interpolates the colors in-between.
Presets
Presets button displays a window where you can load and manage available presets. Hold Ctrl when clicking to load a random preset instead.
Left arrow
Left arrow button loads the previous preset.
Right arrow
Right arrow button loads the next preset.
Randomize

Randomize button loads a random preset. Opacity controls the opacity of the sonogram.
Prefiltering panel

Opacity

Prefiltering panel provides the optional prefiltering, which means that level of each frequency is either increased or decreased before analysis. Normally the analyzer shows scientific levels of each frequency. However you can for example use the predefined loudness curves, which makes the analyzer show how the human auditory system responds to the frequencies, so it in fact provides more accurate analysis taking into account the fact that human hearing is more complicated than the mathematical model.
Depth
Depth controls the amount of prefiltering. 100% makes the analyzer follow the prefiltering graph precisely, 0% essentially disables this feature.

Prefiltering
Envelope graph
Envelope graph provides an extremely advanced way to edit any kind of shape that you can imagine. An envelope has a potentially unlimited number of points, connected by several types of curves with adjustable curvature (drag the dot in the middle of each arc) and the surroundings of each point can also be automatically smoothed using the smoothness (horizontal pull rod) control. You can also literally draw the shape in drawing mode (available via the main context menu).
Left mouse button can be used to select points. If there is a point, you can move it (or the entire selection) by dragging it. If there is a curvature circle, you can set up its tension by dragging it. If there is a line, you can drag both edge points of it. If there is a smoothing controller, you can drag its size. Hold Shift to drag more precisely. Hold Ctrl to create a new point and to remove any points above or below.
Left mouse button double click can be used to create a new point. If there is a point, it will be removed instead. If there is a curvature circle, zero tension will be set. If there is a smoothing controller, zero size will be set.
Right mouse button shows a context menu relevant to the object under the cursor or to the entire selection. Hold Ctrl to create or remove any points above or below.
Middle mouse button drag creates a new point and removes any points above or below. It is the same as holding Ctrl and dragging using left mouse button.
Mouse wheel over a point modifies its smoothing controller. If no point is selected, then all points are modified. Ctrl+A selects all points. Delete deletes all selected points.
Presets
Presets button displays a window where you can load and manage available presets. Hold Ctrl when clicking to load a random preset instead.
Left arrow
Left arrow button loads the previous preset.
Right arrow
Right arrow button loads the next preset.
Randomize
Randomize button loads a random preset.
Envelope graph menu

Envelope graph menu provides additional features which are used to edit the graph. Open the menu using right mouse button in the graph. Please note that if you select some points in the graph, or click on a point for example, the menu will be different and will cover only those features related to the selected set of points.
Presets
Presets button displays a window where you can load and manage available presets. Hold Ctrl when clicking to load a random preset instead.
Left arrow
Left arrow button loads the previous preset.
Right arrow
Right arrow button loads the next preset.
Randomize
Randomize button loads a random preset.
Copy
Copy button copies the settings onto the system clipboard.
Paste
Paste button loads the settings from the system clipboard.
Random
Random button generates random settings using the existing presets.
Snap to grid X
Snap to grid X activates the snap to grid feature. Alternatively you can press Alt while dragging a point or selection.
Snap
Snap button activates the snap to grid feature. Alternatively you can press Alt while dragging a point or selection.

Insert point button creates a point at mouse position.

Insert point

Step sequencer
Step sequencer button generates the envelope from step sequencer.

Clear points button deletes all points.
Randomize
Randomize button slightly modifies the Y coordinates.

Clear points

Mirror X
Mirror X button inverts the X coordinates of all points.
Mirror Y
Mirror Y button inverts the Y coordinates of all points.
Export CSV
Export CSV feature lets you export the graph to a CSV file. CSV file is a simple text format, which has multiple lines with X and Y coordinates delimited by ‘;’. For example: 0.275;0.2 0.438;0.5 0.775;0.67
Import CSV
Import CSV feature lets you select a CSV file and imports the graph points from it. CSV file is a simple text format, which has multiple lines with X and Y coordinates delimited by ‘;’. For example: 0.275;0.2 0.438;0.5 0.775;0.67
Expression evaluator
Expression evaluator lets you generate points based on a mathematic formula. The only input variable is ‘x’, so as an example you may write ‘ln(x^3 + 1) – sin(xx)’.
Expression evaluator uses traditional C/C++ style formating, which is natural for most people. It provides arithmetics, logical and conditional operators. Following terms are supported: Constants: pi, e, sqrt2, ln2
Arithmetic operators: -a inverts the sign, e.g. “-x” produces +2 for x=-2 a+b = addition a-b = subtraction ab = multiplication a/b = division a%b = modulo, remainder after division a^b = power, e.g. “2^3” produces 222 = 8
Arithmetic functions: min(a,b) = minimum of both values max(a,b) = maximum of both values limit(a,min,max) = a limited into the interval min..max to01(a,min,max) = converts “a” as min..max to 0..1 from01(a,min,max) = converts “a” as 0..1 to min..max tom11(a,min,max) = converts “a” as min..max to -1..1 fromm11(a,min,max) = converts “a” as -1..1 to min..max
Basic mathematic functions: abs(x) = absolute value, e.g. abs(-3) = 3 sqr(x) = xx sqrt(x) = square root exp(x) = natural exponential e^x ln(x) = natural logarithm log10(x) = logarithm with base 10 log(x, base) = logarithm with specified base inv(x) = 1/x sgn(x) = sign of x, -1 or 0 or +1 depending on xx round(x) = rounding to the nearest value floor(x) = rounding to the nearest lower value, e.g. floor(-2.3) = -3 ceil(x) = rounding to the nearest higher value, e.g. ceil(-2.3) = -2 rand(x) = random value from 0 to x
Functions for specific units: f01(a) = converts “a” as frequency from 20…20000 into log scale 0..1 ffrom01(a) = converts “a” as 0..1 (log scale) to frequency from 20…20000 todb(a) = converts “a” as multiplier to dB value by calculating “20*log10(a)” fromdb(a) = converts “a” as dB value to multiplier by calculating “10^(a/20)”
Trigonometric functions: sin(x), asin(x), cos(x), acos(x), tan(x), atan(x), sinh(x), cosh(x), tanh(x)
Logical operators: a==b = comparison producing 1 if “a” and “b” are equal, 0 otherwise a!=b = comparison producing 1 if “a” and “b” are NOT equal, 0 otherwise a<b = comparison producing 1 if “a” is lower than “b”, 0 otherwise a<=b = comparison producing 1 if “a” is lower or equal to “b”, 0 otherwise a>b = comparison producing 1 if “a” is greater than “b”, 0 otherwise a>=b = comparison producing 1 if “a” is greater or equal to “b”, 0 otherwise !a = logical negation, 0 produces 1, 0 otherwise a&&b = logical AND, produces 1 if both “a” and “b” are nonzero a||b = logical OR, produces 1 if any of “a” and “b” are nonzero a^^b = logical XOR, produces 1 if “a” and “b” are logically different

a ? b : c = if a is nonzero, then the result is b, otherwise it is c
Analyse audio
Analyse audio lets you analyse a portion of an audio file at specified intervals, extract its level envelope and use those levels to construct the graph’s curve.
Curvature
Integral curvature
Integral curvature makes the multi-curvature modes such as rectangles always have an integral number of items, e.g. 1, 2, 3, … rectangles. If you disable this, it will be also possible to have for example 2.3 rectangles, which will however cause a discontinuity.
Smoothing
Lock sides
Lock sides makes the smoothing factor equal on both sides.
Proportional
Proportional makes the smoothing area size defined by the smaller side.
Faster smoothing
Faster smoothing enables slightly faster algorithm, which can however often cause unnecessary curving.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.
Band panel

Band panel contains parameters of a particular band. You can select a band using the band editor above, just click on the band in the graph.
Presets
Presets button shows a window with all available presets. A preset can be loaded from the preset window by double-clicking on it, selecting via the buttons or by using 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. The arrow buttons next to the preset button can be used to switch between presets easily.
Holding Ctrl while pressing the button loads a random preset. There must be some presets for this feature to work of course.
Presets can be backed up by 3 different methods: A) Using “Backup” and “Restore” buttons in each preset window, which produces a single archive of all presets on the computer. B) Using “Export/Import” buttons, which export a single folder of presets for one plugin. C) By saving the actual preset files, which are found in the following directories (not recommended): Windows: C:Users{username}AppDataRoamingMeldaProduction Mac OS X: /Library/Application support/MeldaProduction
Files are named based on the name of the plugin like this: “{pluginname}.presets”, so for example MAutopan.presets or MDynamics.presets. If the directory cannot be found on your computer for some reason, you can just search for the particular file.
Please note that prior to version 16 a different format was used and the naming was “{pluginname}presets.xml”. The plugin also supports an online preset exchange. If the computer is connected to the internet, the plugin connects to our server once a week, submits your presets and downloads new ones if available. This feature is manually maintained in order to remove generally unusable presets, so it may take some time before any submitted presets become available. This feature relies on each user so we strongly advise that any submitted presets be named and organised in the same way as the factory presets, otherwise they will be removed.
Left arrow
Left arrow button loads the previous preset.
Right arrow
Right arrow button loads the next preset.

Randomize
Randomize button loads a random preset.
Copy
Copy button copies the settings onto the system clipboard.
Paste
Paste button loads the settings from the system clipboard.
Randomize
Randomize button (with the text ‘Random’) generates random settings. Generally, randomization in plug-ins works by selecting random values for all parameters, but rarely achieves satisfactory results, as the more parameters that change the more likely one will cause an unwanted effect. Our plugins employ a smart randomization engine that learns which settings are suitable for randomization (using the existing presets) and so is much more likely to create successful changes. In addition, there are some mouse modifiers that assist this process. The smart randomization engine is used by default if no modifier keys are held. Holding Ctrl while clicking the button constrains the randomization engine so that parameters are only modified slightly rather than completely randomized. This is suitable to create small variations of existing interesting settings. Holding Alt while clicking the button will force the engine to use full randomization, which sets random values for all reasonable automatable parameters. This can often result in “extreme” settings. Please note that some parameters cannot be randomized this way.
Reset
Reset button loads the default settings.
Link
Link button enables parameter linking between bands. Every parameter change performed with this enabled changes that parameter in all bands. Please note that some more rare parameters, which are not available for assignment and automation, may not be changed. But Pasting settings from the system clipboard does not change the other bands.
Left
Left button selects the previous band. If this is the first band, it selects the last one instead. This way you can easily cycle between the bands if selecting them in the band editor is hard because they are modulated for example.
Right
Right button selects the next band. If this is the last band, it selects the first one instead. This way you can easily cycle between the bands if selecting them in the band editor is hard because they are modulated for example.
Drive
Drive defines the power modification applied to the input signal. The higher the drive, the more processed the signal will be. Range: -24.00 dB to +80.00 dB, default 0.00 dB
Output gain
Output gain defines the power modification applied to the output signal. You can use it to compensate for the added volume, however AGC provides an easier solution. Range: -24.00 dB to +24.00 dB, default 0.00 dB
Dry/Wet
Dry/Wet defines ratio between dry and wet signals. 100% means fully processed, 0% means no processing at all. Range: 0.00% to 100.0%, default 100.0%
Character
Character controls the sound character, 0% being most digital, 100% being most analog-like. Above 100% it gets, well, more than analog :).

Range: 0.00% to 200.0%, default 100.0%
Symmetry
Symmetry offsets the signal vertically making the parts above 0 get processed differently from parts below 0 resulting in various changes in timbre. Range: -100.0% to 100.0%, default 0.00%
Non-linearity
Non-linearity introduces some more complex behaviour such that different levels are processed differently. This results in various changes in timbre. Range: 0.00% to 100.0%, default 0.00%
Pre HP
Pre HP defines the input high-pass filter cut-off frequency which may be used to remove part of the signal before the actual processing, which usually lowers the amount of distortion. Range: Off to 20.0 kHz, default Off
Pre LP
Pre LP defines the input low-pass filter cut-off frequency which may be used to remove part of the signal before the actual processing, which usually lowers the amount of distortion. Range: 20.00 Hz to Off, default Off
Post LP
Post LP defines the output low-pass filter cut-off frequency which may be used to remove part of the signal after the actual processing, hence removing some of the distorted signal. Range: 20.00 Hz to Off, default Off
Oversampling
Oversampling provides an oversampling which may be essential to get truly analog sound without digital artifacts. It requires less CPU than the global oversampling, is designed specifically for this purpose and sounds different. A factor of 8x is usually sufficient unless extreme modulation is used, in which case the resulting extreme sound properties are usually more like a result of the processing than a digital. Range: 1 to 1024, default 1
Automatic gain control
Automatic gain control (AGC) enables automatic adjustment of the output volume so that it matches the input volume. Note that since this is a nonlinear operation, it may cause some distortion of its own. Also note that it takes some time for the AGC to adjust the volume when its setting has been changed.
DC blocker
DC blocker activates the integrated DC blocker that should remove any signal offset.

Oscilloscope
Oscilloscope displays the output signal shape, which may be useful for creative design.
Global parameters panel
Global parameters panel contains global controls, which are usually relevant to global processing performed either before the signal reaches the crossover and gets split into bands, or after the signals are processed and summed back to the master signal.
Dry/Wet
Dry/Wet defines the ratio between dry and wet signals. 100% means fully processed, 0% means no processing at all.
0%
0% button makes the Dry/Wet virtually 0%. You can use it for comparison.
100%
100% button makes the Dry/Wet virtually 100%. You can use it for comparison.
Input gain
Input gain defines the power modification applied to the incoming signal, before it is split into bands.
Output gain
Output gain defines the power modification applied to the output signal, right after it is summed from the bands.

Global meter view
Global meter view provides a powerful metering system. If you do not see it in the plug-in, click the Meters or Meters & Utilities button to the right of the main controls. The display can work as either a classical level indicator or, in time graph mode, show one or more values in time. Use the first button to the left of the display to switch between the 2 modes and to control additional settings, including pause, disable and pop up the display into a floating window. The meter always shows the actual channels being processed, thus in M/S mode, it shows mid and side channels. In the classical level indicators mode each of the meters also shows the recent maximum value. Click on any one of these values boxes to reset them all.
Numbered band meters display the input levels for each band.
In meter indicates the total input level. The input meter shows the audio level before any specific processing (except potential oversampling and other pre-processing). It is always recommended to keep the input level under 0dB. You may need to adjust the previous processing plugins, track levels or gain stages to ensure that it is achieved.
As the levels approach 0dB, that part of the meters is displayed with red bars. And recent peak levels are indicated by single bars.
Out meter indicates the total output level. The output meter is the last item in the processing chain (except potential downsampling and other post- processing). It is always recommended to keep the output under 0dB.
As the levels approach 0dB, that part of the meters is displayed with red bars. And recent peak levels are indicated by single bars.
Width meter shows the stereo width at the output stage. This meter requires at least 2 channels and therefore does not work in mono mode. Stereo width meter basically shows the difference between the mid and side channels. When the value is 0%, the output is monophonic. From 0% to 66% there is a green range, where most audio materials should remain. From 66% to 100% the audio is very stereophonic and the phase coherence may start causing problems. This range is colored blue. You may still want to use this range for wide materials, such as background pads. It is pretty common for mastered tracks to lie on the edge of green and blue zones. Above 100% the side signal exceeds the mid signal, therefore it is too monophonic or the signal is out of phase. This is marked using red color. In this case you should consider rotating the phase of the left or right channels or lowering the side signal, otherwise the audio will be highly mono-incompatible and can cause fatigue even when played back in stereo. For most audio sources the width is fluctuating quickly, so the meter shows a 400ms average. It also shows the temporary maximum above it as a single coloured bar. If you right click on the meter, you can enable/disable loudness pre-filtering, which uses EBU standard filters to simulate human perception. This may be useful to get a more realistic idea about stereo width. However, since humans perceive the bass spectrum as lower than the treble, this may hide phase problems in that bass spectrum.

Time graph
Time graph button switches between the metering view and the time-graphs. The metering view provides an immediate view of the current values including a text representation. The time-graphs provide the same information over a period of time. Since different time-graphs often need different units, only the most important units are provided.
Pause
Pause button pauses the processing.
Popup
Popup button shows a pop-up window and moves the whole metering / time-graph system into it. This is especially useful in cases where you cannot enlarge the meters within the main window or such a task is too complicated. The pop- up window can be arbitrarily resized. In metering mode it is useful for easier reading from a distance for example. In time-graph mode it is useful for getting higher accuracy and a longer time perspective.
Enable
Enable button enables or disables the metering system. You can disable it to save system resources.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.
Utilities
Map
Map button displays all current mappings of modulators, multiparameters and MIDI (whichever subsystems the plugin provides).

Modulator
Modulator button displays settings of the modulator. It also contains a checkbox, to the left, which you can use to enable or disable the modulator. Click on it using your right mouse button or use the menu button to display an additional menu with learning capabilities as described below.
Menu
Menu button shows the smart learn menu. You can also use the right mouse button anywhere on the modulator button.
Learn activates the learning mode and displays “REC” on the button as a reminder, Clear & Learn deletes all parameters currently associated with the modulator, then activates the learning mode as above. After that every parameter you touch will be associated to the modulator along with the range that the parameter was changed. Learning mode is ended by clicking the button again.
In smart learn mode the modulator does not operate but rather records your actions. You can still adjust every automatable parameter and use it normally. When you change a parameter, the plugin associates that parameter with the modulator and also records the range of values that you set.
For example, to associate a frequency slider and make a modulator control it from 100Hz to 1KHz, just enable the smart learn mode, click the slider then move it from 100Hz to 1KHz (you can also edit the range later in the modulator window too). Then disable the learning mode by clicking on the button.

Menu
Menu button displays additional menu containing features for modulator presets and randomization.
Lock
Lock button displays the settings of the global parameter lock. Click on it using your left mouse button to open the Global Parameter Lock window, listing all those parameters that are currently able to be locked. Click on it using your right mouse button or use the menu button to display the menu with learning capabilities – Learn activates the learning mode, Clear & Learn deletes all currently-lockable parameters and then activates the learning mode. After that, every parameter you touch will be added to the lock. Learning mode is ended by clicking the button again. The On/Off button built into the Lock button enables or disables the active locks.

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

Collapse

Multiparameter
Multiparameter button displays settings of the multiparameter. The multiparameter value can be adjusted by dragging it or by pressing Shift and clicking it to enter a new value from the virtual keyboard or from your computer keyboard.
Click on the button using your left mouse button to open the Multiparameter window where all the details of the multiparameter can be set. Click on it using your right mouse button or click on the menu button to the right to display an additional menu with learning capabilities – as described below.
Menu
Menu button shows the smart learn menu. You can also use the right mouse button anywhere on the multiparameter button.
Learn attaches any parameters, including ranges. Click this, then move any parameters through the ranges that you want and click the multiparameter button again to finish. While learning is active, “REC” is displayed on the multiparameter button and learning mode is ended by clicking the button again.
Clear & Learn clears any parameters currently in the list then attaches any parameters, including ranges. Click this, then move any parameters through the ranges that you want and click the multiparameter button again to finish. While learning is active, “REC” is displayed on the multiparameter button and learning mode is ended by clicking the button again.
Reset resets all multiparameter settings to defaults.
Quick Learn clears any parameters currently in the list, attaches one parameter, including its range and assigns its name to the multiparameter. Click this, then move one parameter through the range that you want.
Attach MIDI Controller opens the MIDI Settings window, selects a unused parameter and activates MIDI learn. Click this then move the MIDI controller that you want to assign.
Reorder to … lets you change the order of the multiparameters. This can be useful when creating active-presets. Please note that this feature can cause problems when one multiparameter controls other multiparameters, as these associations will not be preserved and they

will need to be rebuilt.
In learning mode the multiparameter does not operate but rather records your actions. You can still adjust every automatable parameter and use it normally. When you change a parameter, the plugin associates that parameter with the multiparameter and also records the range of values that you set. For example, to associate a frequency slider and make a multiparameter control it from 100Hz to 1KHz, just enable the smart learn mode, click the slider then move it from 100Hz to 1KHz (you can also edit the range later in the Multiparameter window too). Then disable the learning mode by clicking on the button.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.

Preset selector
Preset management window provides management for your presets.
Backup
Backup button lets you backup presets for all MeldaProduction software into a single file, so you can transfer it to a different machine and restore the presets there for example.
Restore from backup
Restore from backup button lets you restore presets for all MeldaProduction software from a single file created by the Backup button.
Folders tree

Folders tree lets you organize your presets into any number of folders. Use the buttons at the bottom of the window to create, rename or delete sub- folders. Note that these are not actual files & folders on disk, but are records in the preset database.
Auto-open
Auto-open switch makes the tree automatically open selected items, so that all sub-folders are visible, whenever you select one. This makes it easier to browse through large structures containing many folders. The switch also makes the browser show all presets available in the selected folder including all sub-folders (except when you select the root folder).
Open all
Open all button expands the whole tree, so you can see all of the folders. This may be handy when editing large preset structures.
Close all
Close all button collapses the whole tree except for the root folder. This may be handy when editing large preset structures.
Add
Add button creates a new folder in the tree
Rename
Rename button lets you rename the selected folder.
Delete
Delete button deletes the folder including all the presets and subfolders in it.
Export
Export button lets you export the selected folder including all presets and sub-folders into a file, which you can then transfer to any computer. Or just use as a back-up.
Import
Import button lets you import a file containing presets and sub-folders and add it to the selected folder. The importer will ask you whether to destroy the original contents, so that the new presets replace previous ones, or to keep both.

Presets list
Presets list contains all presets available in the selected folder. Double- click on a preset or use Load button to load a preset. Use the buttons at the bottom of the list to perform additional changes. Please note that these are not actual files & folders on disk, but are records in the preset database.
Favourite
Favourite button toggles the ‘favourite’ indicator for the selected preset.
Show
Show button shows only the favourite presets and hides the others.
Sort
Sort button shows the presets sorted alphabetically.
Random
Random button selects and loads a random preset from the current folder. This way you can quickly browse the presets in the folder in a completely random order.
Previous
Previous button selects and loads the previous preset from the current folder.
Next
Next button selects and loads the next preset from the current folder.
Submit preset
Submit preset button submits the selected preset to the online exchange servers and retrieves all the presets currently in the database. This feature serves as an online database of presets available for all the user community. Please do not submit garbage presets.
Download presets
Download presets button retrieves all the presets currently in the database. This feature serves as an online database of presets available for all the user community. Please consider participating by submitting your presets as well.
Load

Load button loads the specified preset. Please note that you can do the same thing by double-clicking the preset itself or pressing the Enter key.

Add
Add button creates a new preset using the current settings.

Rename
Rename button lets you rename the selected preset.

Replace
Replace button replaces the selected preset by one with current settings.

Delete
Delete button deletes the selected preset.

Search filters the list of available presets to those containing the keywords in name or information.

Search

Clear
Clear button deletes all text in the search field.

Preset information
Preset information field contains optional information about the preset, which you can edit when creating or renaming the preset.

Plugin settings
Plugin settings window offers more advanced settings and is available via the Settings button.

Licence panel
Licence panel lets you manage licences on this computer.
Activate
Activate button lets you activate your licence for the plugin on this computer.
GUI & Style panel

GUI & Style panel lets you configure the plugin’s style (and potentially styles of other plugins) and other GUI properties.

Style button lets you change the style for this particular plugin.

Style

Random style
Random style button selects a random style with random editor mode.

Default style
Default style button reverts to the default style and default size of the GUI. Hold the Ctrl key while clicking to revert all MeldaProduction

software products, not just the current plugin.
Select current style as default
Select current style as default button stores the current style as the default for all MeldaProduction software. This is used for the other plugins that are currently using the default style; that is, those plugins for which you have NOT selected a specific style. Please note that if you have already selected a specific style for a particular plugin, then it won’t be changed until you use the Default style button.
GPU acceleration
GPU acceleration controls how much the GPU is used for visual rendering to save CPU power. Enabled mode provides maximum speed and lets the GPU perform as many drawing operations as possible. Compatibility mode uses the GPU for drawing, but doesn’t use modern technologies for maximum performance. Use it if you experience occasional problems with drawing, the usual case for older ATI graphics cards. With Pro Tools on OSX this mode is always used instead of Enabled mode due to compatibility problems with this host. Disabled mode disables GPU acceleration completely, drawing is then performed by the CPU. Use only if you experience technical difficulties. A known problem may occur when using multiple displays with multiple graphical interfaces. When moving the plugin window from one display to another, it may stop displaying correctly until you move it back to the original display.
Frames per second
Frames per second controls the refresh rate of the visual engine. The higher the number is the smoother everything is, but the more CPU it requires. You might want to lower this value if your computer is running out of CPU power.
Enable high DPI / retina support
Enable high DPI / retina support enables the plugin to use the high resolution on high DPI (Windows) and retina (OSX) devices. It is enabled by default and detected automatically, if the host allows it. If you run into any problems, you can disable it using this option. It may be desired if you use multiple displays where only some of them feature the high resolution making the image on the low resolution ones look ugly.
If you disable this option, on Windows the high DPI device detection will be ignored and the plugin will probably appear very small. You can manually compensate for it by using a bigger style. On OSX disabling this option will disable the high DPI rendering, resulting in the classic blurry look of non- compliant applications. Changes take effect after you restart the host.
Enable colorization
Enable colorization enables the plugin to change the colors of certain elements overriding your style settings. Plugins use that to highlight different parts of the graphics interface for easier workflow. You may want to disable it if you just feel it’s not for you. This particular option is relevant only for controls – knobs, sliders, checkboxes etc.
Enable colorization for panels
Enable colorization for panels enables the plugin to change the colors of certain elements overriding your style settings. Plugins use that to highlight different parts of the graphics interface for easier workflow. You may want to disable it if you just feel it’s not for you. This particular option is relevant only for containers – panels, graphs etc.
Allow default colors by plugin type
Allow default colors by plugin type is on by default and makes the plugin select its default colors depending on the type of the Plugin. Hence for instance equalizer will always be green. This is done by selecting one of the first 8 color presets for the current style, so the actual colors depend on selected style and its presets. You may want to disable this if you for example want all plugins to look the same including the style and colors. It is necessary to restart your host for a change to this option to take effect.
Allow style changes if the editor is too big
Allow style changes if the editor is too big is on by default and makes the plugin change its style, editor mode and other settings if it finds out it is too big to fit the current screen resolution.
Clear window settings cache
Clear window settings cache button deletes stored states of all popup windows on all MeldaProduction software. The window settings mostly contain positions and sizes, but in some cases also the data inside the popup windows. You can use this feature if something goes wrong, a window doesn’t appear at all, problems like that. While this shouldn’t happen and it’s generally better to contract our support, this button provides a potential quick fix.
Dry/Wet affects… panel

Dry/Wet affects… panel controls which multiband parameters are affected by the global dry/wet controls. Dry/Wet generally controls the mix between the processed and input signal, however it is often advantageous to support its function with other parameter. In most cases it is good to enable everything except for global input gain. However note that this ‘default’ is not specified for most plugins for sake of maintaining backwards compatibility.
Band gain
Band gain makes the global dry/wet parameter affect the input gain of each band. In a way you may say that the input gain settings of all bands are applied to wet only when this is enabled. This is suitable in most cases, because that way lowering dry/wet to 0% provides a true “dry” response. If this is not enabled and dry/wet is 0% the output will still be affected by the input gain of each band.
Band panorama
Band panorama makes the global dry/wet parameter affect the input panorama of each band. In a way you may say that the input panorama settings of all bands are applied to wet only when this is enabled. This is suitable in most cases, because that way lowering dry/wet to 0% provides a true “dry” response. If this is not enabled and dry/wet is 0% the output will still be affected by the input panorama of each band.
Global input gain
Global input gain makes the global dry/wet parameter affect the global input gain. This is usually not desired, because the input gain affects the functionality of most plugins, so this would affect the processed signal, which is not necessary, because the dry/wet itself is creating the mix between the dry and wet signals. However this control may be handy in specific creative scenarios.
Global output gain
Global output gain makes the global dry/wet parameter affect the global output gain. In a way you may say that the output gain is applied to wet only when this is enabled. This is suitable in most cases, because that way lowering dry/wet to 0% provides a true “dry” response. If this is not enabled and dry/wet is 0% the output will still be affected by the global output gain.
Plugin settings panel
Plugin settings panel contains settings that control the behaviour of this plugin instance. These are properties that rarely need to be changed, so they have been moved here.
Intelligent sleep on silence
Intelligent sleep on silence option provides a huge CPU saver by automatically disabling the plugin processing if the input is silent and if the plugin doesn’t generate some signal on its own. This makes the plugins take virtually no CPU if there is no need for them to actually process anything. Disable this if you run into any problems with them.
Randomizer loudness compensation
Randomizer loudness compensation enables the automatic detection of loudness after new settings have been generated using the main Random button and using the output gain of the plugin to get some predefined level. This is useful in most cases since normally randomized settings can produce various output levels, so this can mitigate the problem.
Smart bypass

Smart bypass enables the high quality crossfading bypass system, which ensures a smooth transition between the processed and dry signals. You may want to disable it if you are using settings with latency on a plugin, which demands lots of CPU power, which would otherwise need to perform processing even when bypassed, which is pretty much the only downside of the smart bypassing algorithm.
MIDI thru
MIDI thru makes the plugin pass all input MIDI through to its MIDI output. That is often advantageous in DAWs such as Reaper, which naturally pass MIDI from one plugin to the next.
Sample-accurate event processing
Sample-accurate event processing makes the plugin schedule every event such as MIDI or automation to their accurate locations with sample accuracy, if the host allows it. For example, if the block size in your host’s audio settings is 1024 samples, this means the plugin is probably processing blocks of 1024 samples, in 44100 Hz sampling rate it is about 23ms. If this setting is disabled, any change in automation, MIDI, modulation etc. may then be granularized to 23ms (once per block), which means that you will not be able to recognize events that occur say 10ms apart from each other. When this setting is enabled however, the plugin divides processing blocks to sub-blocks and processes the events at their correct positions. This may, of course, require more CPU power.
Latency reporting
Latency reporting makes the plugin report latency to the DAW, if any. Normally this is enabled, but in certain live situations you may want to disable this, so that the DAW stops compensating the latency on other tracks. It has no effect if the plugin is placed on master track.
Custom GUI for devices
Custom GUI for devices enables displaying custom GUIs for the easy screen devices. You can disable it if you like the generic GUI better.
Global system settings panel
Global system settings panel contains settings which are applied to all plugins on this computer.
Intelligent sleep on silence (global)
Intelligent sleep on silence (global) is a global switch, which disables the Auto disable on silence feature in all plugins on the system. It is provided “just in case” something goes wrong.
Right click sets default value
Right click sets default value makes the engine set default value to a parameter when you right click on it. By default, a menu is displayed instead, with an option to set the default value, but potentially with more features. When this is disabled, you can still set a default value by holding ctrl/cmd when right clicking the control.
Tablet mode
Tablet mode enables better support for tablets at the expense of the mouse. Enable this if you are using a tablet to control the plugins and it is behaving incorrectly.
Enable keyboard input
Enable keyboard input enables the keyboard input for the main plugin window. You may want to disable if the plugin intercepts spacebar key (often used by the host for playback enable/disable and your host doesn’t allow for the problem itself.
Collapse plugin toolbar
Collapse plugin toolbar makes all plugins collapse the plugin toolbar containing more advanced features such as channel modes, A-H presets, oversampling, safety limiter etc. It is enabled by default to make the user interfaces cleaner and easier to grasp for beginners.

Set default settings
Set default settings button stores the current plugin settings as the defaults, so that when you open a new instance of the plugin, these settings will be loaded automatically.
Reset default settings
Reset default settings button removes the defaults that you set using Set default settings button, so that when you open a new instance of the plugin, the factory defaults will be loaded.
Advanced global settings panel
Advanced global settings panel contains advanced settings which are applied to all plugins on this computer.
Saturation antialiasing
Saturation antialiasing enables a global support for antialiasing in saturation algorithms available in many of the plugins. These require additional CPU processing, however significantly reduce aliasing artifacts without a need for oversampling.
Forward unused keyboard input to DAW
Forward unused keyboard input to DAW makes the plugin forward unused keyboard events to the DAW from its popups. If this is disabled, pressing say spacebar commonly used to start/stop playback won’t work if a popup window is active. Enabling this makes this work and it is optional just in case your DAW does something unexpected.
Silence when busy
Silence when busy makes all plugins silence the output when something time consuming is being performed in background and the plugin needs to wait for it. For instance, in modular plugins such as MXXX, adding a module requires lots of changes in the entire engine, so it is performed in background and while the plugin is inconsistent state, it is temporarily bypassed. Sometimes however, when performing live, bypassing makes the dry signal go through and that may not be wanted. So you can enable this option, and the plugin will silence the output instead.
Store resampled files
Store resampled files allows the plugins create audio files for sampling rates being used if they differ from the original file sampling rate. It is used only by a few plugins, but it can improve the loading performance a lot at the cost of some additional storage on the hard drive. Disable this option if you are short on free space.
Show confirmations for destructive actions
Show confirmations for destructive actions makes the plugin display a confirmation window whenever you are going to change the plugin settings irreversibly when using a feature, for example: when resetting your settings.
Online check for updates and tutorials
Online check for updates and tutorials lets the plugin ask about once a week if there is a new version or tutorial available so you can be easily kept up to date.
Anonymous online platform reporting
Anonymous online platform reporting helps us maximize compatibility with your operating system and host. If enabled, our plugins will send information about the system and host that you are using. We can use this information to find out which plugins and platforms are used the most and maximize testing and support there. Platform reporting is completely anonymous and requires only minimal internet connection time (a few kB once a week).

CPU benchmark
CPU benchmark button calculates the performance of the plugin with the current settings.
System info
System info button displays some technical information about the build and the machine.
Compatibility settings panel
Compatibility settings panel contains advanced settings you rarely need unless you run into some problems when using multiple versions or old projects.
Storage compatibility mode for V15
Storage compatibility mode for V15 reverts to the older and much slower storage system used by version 15 and older. Use this if you want to open your projects or presets on older version of MeldaProduction plugins.
Automation compatibility mode for V10
Automation compatibility mode for V10 reverts the set of automation parameters back to version 10 and earlier. Use this if you need the plugins to work with projects, which contain autmation, made using version 10 or older. In version 11 the list of automatable parameters have been highly simplified and reorganized and multiparameters are provided for the vast number of hidden parameters. This should speed up loading, improve workflow with the plugins and improve compatibility with various hosts.
Smart interpolation panel
Smart interpolation panel controls the depth of the smart interpolation algorithm, which controls the parameters in order to provide maximum audio quality and lower the chance of zipper noise. Smart interpolation is engaged whenever you change any parameter via the GUI, modulators, multiparameters, MIDI or automation. Many parameters can be automated easily and the plugin responds with sample-accurate results. However, several parameters need exhaustive pre-processing when changed. In these cases, the parameters are not updated every sample, but, for example, once every 32 samples. This highly reduces CPU usage, but affects the output quality. With modulators the situation is more complicated. Besides the updating issue, the modulator itself can perform some pretty advanced processing, hence it is better to perform the processing in blocks. However, the bigger the block, the less often the modulator updates those parameters associated with it and the resulting modulation is less accurate. In a way you can say that the modulator is slower and lazier. This may actually be wanted, so when it comes to modulators it is not true that a better mode always means better output quality. The smart interpolation mode controls the maximum number of samples being processed before the parameters are updated. Minimal mode uses 2048 samples and rarely will do anything unless processing offline. Normal mode uses 256 samples and usually is enough to achieve good quality results. High mode uses 32 samples and provides perfect quality for most cases. It is also a good compromise

between CPU usage and audio quality, so it is the default. Very high mode uses 4 samples and you will rarely need it. Extreme mode uses 1 sample, which means that everything is updated after every single sample. This provides the highest possible accuracy and quality you can ever achieve, however it requires lots of CPU and it is very unlikely that you will ever need it. If you use this mode and still hear audio artifacts, then either what you are hearing is actually CPU overload, or you are doing something that is not physically possible.
The higher the mode, the quicker the parameter updates, but the more the CPU load.
Please note that modulating certain parameters without artifacts is impossible. For example, when modulating a delay very quickly, the physics of such a process just cannot occur in the natural world and the results are appropriately unnatural. These physically impossible processes usually manifest themselves as distortion or zipper noise.

Modulator editor
Modulator is an extremely advanced feature, which lets you change parameters automatically depending on various inputs. You can use this to add movement to your sound, respond to some plugins differently for louder sections, or even follow the pitch of the input. The modulator edit window has two parts: on the left side you can configure the mode of the modulator (the way the modulator works) and on the right side there is a list of parameters to modulate. A modulator can control all automatable parameters (and often more than that) including the parameters of other modulators. Each modulator can control as many parameters as is needed and each of the parameters has its own range and transformation shape. The values and ranges of the first 4 parameters associated with the other modulators can also be modulated/automated. The following modulator modes are available: Normal mode makes the modulator behave like an ordinary low-frequency oscillator (LFO). There are various ways to control its shape as with all oscillators in our plugins. Each modulator can synchronize to the host in the Synchronization panel. Modulators can also synchronize with each other using the Sync groups. Using MIDI reset you can reset the oscillator to any phase using MIDI notes, but obviously to-host synchronization must be disabled in order for this to work. Note that the settings in this mode are used even if the modulator is actually in a different mode by using “LFO modulation”. This basically blends between the actual mode, which may for example detect the input signal level, and give it some additional movement using the LFO depending on the LFO modulation parameter available for each of the remaining modes. Follower mode makes the modulator detect the input signal level. It contains an extremely advanced and accurate level detector taken from our MDynamics plugin. The level follower is an immensely useful feature, yet it may be a little difficult for beginners to comprehend, so we will cover it here in more detail. It is often necessary to adjust the follower slightly for new material. First, it has the standard parameters – attack, release, hold and RMS length. These are fairly standard features and help is available for each of them. Level min and max controls the range of input levels. When the input level is equal to or below the min level, the modulated parameters’ values will be minimal. Similarly, when it reaches the max level, the modulated parameters’ values will be at their maximum. This allows for adjustments to the range of input levels, which are certainly different for any audio material and settings. It can be used creatively too – for example, by using very low values for both limits we can differentiate between silent and non-silent parts, similar to the way a gate effect works. Advanced detector settings provide some extraordinary features, such as psycho-acoustic pre-filtering, which forces the modulator to detect loudness instead of raw input levels, custom input signal pre-filtering using a fully featured 6-band equalizer, and custom attack and release shapes. Band- pass panel pre-filters the level detection signal using a band-pass filter, so this is like a very simplified version of the

equalizer from the advanced detector settings. Side-chain makes the modulator measure side-chain input if the plugin has one. For modular plugins the modulator can also be driven by a feedback signal. The advanced panel provides some further level processing features that you can take advantage of creatively or to further adjust to your actual audio material.
Project onto LFO shape is a more advanced concept, which is available for other modulator modes too. You can easily imagine, that the modulator in any mode generates values for each parameter, we can say it is between 0 and 1, where 0 sets minimum parameter value, and 1 sets the maximum. Project onto LFO shape forces the modulator to use this range in the oscillator shape, which can then be configured in normal mode. The value is basically transformed by the oscillator shape, where the values generated by the modulator are on the horizontal axis (phase) and the output is the actual oscillator value. This feature has no physical meaning and can only be used creatively – to transform the more or less linear results of the level follower into a much more complicated curve.
Let us demonstrate the follower mode with an example – the idea is to apply a delay to a snare drum within a previously mixed drumset. This is commonly used on reggae/dub rhythms for example, however in these cases the snare track is usually available separately. Using the modulators you can get somewhat interesting results even with an already mixed drumset. The idea is to increase the input gain whenever the snare is playing, so that only the snare drum (and potentially other instruments playing at the same moment) are passed into the delay. So first teach the modulator to control input gain parameter of the delay and set it to follower mode, potentially configure some of the parameters to get the desired response. Now the louder the input is, the more delay you get. To make it respond only to snare drum, enable the band-pass and set the filter limits accordingly, e.g. 500Hz to 1k. This makes the input gain increased depending on the input level in this part of the spectrum, which contains the snare drum.
Envelope mode causes the modulator to generate an arbitrary envelope, similar to those from synthesizers. It can either follow MIDI – the envelope starts when a key is pressed, goes though the attack and decay stages, then holds in sustain stage until the key is released when the release stage begins, or it can follow audio – when the audio level exceeds Threshold on it behaves the same way as when a note is pressed in MIDI mode, and then when the input level drops below Threshold off it behaves like a key release. As with most modes there is LFO modulation and LFO projection and the input level can be driven by the side-chain or feedback if available. The envelope shape can be adjusted using several controls (lengths of each stage etc.) and you can even draw your own shape.
Random mode is a smooth random generator. It is very handy if you want some parameters to change over time, but do not actually want them to be periodic like LFOs. A modulator in random mode does not actually generate random values, the results will always be the same at each position in your arrangement in the host. This allows a pseudo synchronization with the host and ensures a “what you hear is what you get” performance. Speed parameter controls the speed of change and any slight change to this parameter will change the whole stream.
Pitch detects the pitch of the input signal assuming it is not polyphonic (here it can work too and will probably detect the lowest note, however it is definitely not suitable for percussive signals, which do not have a pitch). It is very useful, enabling you to tune an oscillator to follow your singing, or allow an equalizer to control separate harmonics of a vocal, use a distortion to get more drive for higher notes in a guitar solo and much more. The pitch detection may be a little tricky to understand, so we will discuss it in more detail.
A pitch detector takes the input signal and tries to approximate the pitch of the fundamental frequency in it. It is physically impossible to detect pitch instantly, as an extreme example, 20Hz takes 50ms for the signal to evolve enough to detect that there is actually a 20Hz frequency in the signal. For this and many other reasons any pitch detector employs several limitations. These are available in the Detector panel. The defaults will work well for most audio material, however, it is useful to understand the parameters, so that you can let the detector adapt better to your particular audio materials if necessary, and also in order to be more creative.
Min and max frequency parameters in the Detector panel control the limits of the frequencies you expect in the input. For example, a female voice is unlikely to sing below 100Hz, so it is customary to set the minimum frequency to 100Hz or even higher. Voice signals contain several artifacts, blows and pops, all of which can temporarily create frequencies below the actual pitch of the voice, so setting these limits is preferable to avoid “jumps” to incorrect pitches. Stabilization and Speed also prevent these jumps by restricting how quickly the pitch can change. These can also be used creatively. Threshold controls the minimum level of the input signal to be considered “not-silent and probably having pitch”. This acts as a form of gate, which prevents the detector from analyzing irrelevant rumble in between actual performances. Shift panel allows the detected pitch to be shifted up or down and Auto-tune panel moves it to the closest note – similar to the automatic pitch changing function from MAutoPitch, except no pitch shifting is actually done and the results are used purely to control some parameters.
Min and max frequency parameters in the top of the editor have a very different meaning than the parameters of the same name in the detector panel. From now on we will assume that the pitch has been detected successfully and are now considering what to do with the results. Again, we may assume the modulator generates values from 0 to 1, where at 0 the modulated parameters’ values become minimal and reach maximum at 1. When the input pitch is equal or below the min frequency parameter, the modulator’s value is 0, hence modulated parameters will have a minimal value as well. Similarly when the pitch reaches max frequency, the modulated parameters will get to the maximum.
Now you may say this makes no sense, because the detected pitch cannot exceed the limits specified in the Detector panel anyway. The reason for this is that most “frequency” parameters of all plugins are limited from 20Hz to 20kHz, whether it is the frequency of a band in an equalizer, or a high-pass frequency in a phaser for example. It is a reasonable solution since physiologically speaking these figures are on or around the range of our hearing limits.
Let us explain the concept with an example. We want to modulate a band of an equalizer, so that it always follows the fundamental frequency, the pitch, of our audio material. All we need to do is to switch the modulator to pitch mode, allow it to control the band frequency parameter and set the range for this parameter to the full range, from 20Hz to 20kHz. The pitch detector may then detect frequencies from 50Hz to 2kHz, but the modulator takes it that the actual limits (converted to 0..1) are 20Hz to 20kHz and that exactly the same range is configured for the band frequency parameter, so you could say that “they understand each other”. We did not need to touch the min and max frequency parameters at all.
Here is one more example, where we would actually want to adjust the min and max frequency parameters. We want to control a drive parameter of a distortion for a guitar so that the higher the guitarist plays the more distortion he gets. Again, we teach a modulator to control the drive parameter, for any range we want, and switch the modulator to pitch mode. Now the modulator will move the drive

parameter, but only slightly, because it assumes the pitch can vary from 20Hz to 20kHz, but the guitar may actually only play from about 100Hz to 1kHz. So we can use the min and max frequency parameters to say “what is high and what is low”, to limit the frequency range. There are no general rules here, you have to experiment, because every instrument and parameter is different. To sum things up, the difference between controlling a frequency parameter and a drive parameter is simply the fact that a frequency parameter is compatible with the pitch. After all, pitch is nothing more than a frequency (strictly speaking it is a logarithmic representation of frequency).
Presets
Presets button displays a window where you can load and manage available presets. Hold Ctrl when clicking to load a random preset instead.
Left arrow
Left arrow button loads the previous preset.
Right arrow
Right arrow button loads the next preset.
Randomize
Randomize button loads a random preset.
Copy
Copy button copies the settings onto the system clipboard.
Paste
Paste button loads the settings from the system clipboard.
Random
Random button generates random settings. Note that unlike copy & paste, presets & randomization do NOT affect the set of parameters being modified, hence it serves to optimize adjustment of the modulator behaviour assuming that you already specified the set of parameters to control. If you hold Shift, the plugin will undo previous randomization.
R
R button enables automation read. This way you can actually automate the modulation value. First you use W button to record the modulator values over time. After that you can modify it in some way and enable automation read to override the normal modulator behaviour. Note that the results may be different when automation is used with potentially lower audio quality and slower response.
W
W button enables automation write. This way you can actually automate the modulation value. Use the button to record the modulator values over time. After that you can modify it in some way and enable automation read to override the normal modulator behaviour. Note that the results may be different when automation is used with potentially lower audio quality and slower response.
Map
Map button displays all current mappings of modulators, multiparameters and MIDI (whichever subsystems the plugin provides).
Parameters panel

Parameters panel contains the list of the parameters that the modulator is controlling, their ranges etc.
Presets
Presets button displays a window where you can load and manage available presets. Hold Ctrl when clicking to load a random preset instead.
Left arrow
Left arrow button loads the previous preset.

Right arrow
Right arrow button loads the next preset.
Randomize
Randomize button loads a random preset.
Parameter list
Add
Add button adds a parameter to the list of controlled parameters. Alternatively you can use the learn feature available by rightclicking the modulator button.

Delete
Delete button deletes the selected parameter from the list of controlled parameters.
Learn
Learn button starts or stops the learning. Click it, then move some parameters in the plugin, then click it again. Learning can also be accessed from the global modulator menu.
Up
Up button moves the selected parameter up one item, if possible. This may be useful when keeping things organized, but please note that if you have some other multiparameter, modulator or another subsystem access the ranges of individual parameters, this function will reorder them, so these connections will no longer be correct.
Down
Down button moves the selected parameter down one item, if possible. This may be useful when keeping things organized, but please note that if you have some other multiparameter, modulator or another subsystem access the ranges of individual parameters, this function will reorder them, so these connections will no longer be correct.
Parameter Settings
Parameter
Parameter defines the target parameter which is being modulated. The set contains all automatable parameters.
Name
Name lets you name the parameter somehow and may be helpful in situations, where there are many parameters being edited without obvious meanings.
Range mode
Range mode defines how the parameter range is selected. While sometimes it is better to specify minimum and maximum, other times it is better to use a nominal center and depth (% of full scale). This control allows you to define which one it will be. Up and down mode makes the values go above and below the selected Value, which is considered the center. The interval is made smaller if necessary. Full range mode is similar, except the range is symmetrically constrained, so the selected Value may not be the center anymore. Up/down only modes goes from the selected value up/down only. Let’s compare these 4 modes. Taking a value of -12dB value, with a depth of 75% and a scale of +/- 24dB. The nominal range is therefore = +/-24 dB 75% = 36dB. With values of 0%, 50% and 100% the outputs are: Up and down: -24, -12, 0 (range constrained to 12 dB either side) Full range: -24, -6, 12 (range limited to minimum, but not constrained) Up only: -12, 6, 24 (range not constrained = +/-24 dB 75% = 36dB) Down only: -12, -18, -24 (range limited to minimum) Interval mode is the most simple one and goes from Value to Maximal value.
Value

Value defines the center of the target parameter’s range or the minimum if the Range mode is set to Interval.
Maximal value
Maximal value defines the upper limit of the target parameter’s range. It is available only if the Range mode is set to Interval. This value can be lower than Value. 0% is always mapped to reference>Value and 100% to reference>Maximal value.
Depth
Depth defines size of the target parameter’s range. It is used only if the Range mode is not set to Interval.
Invert
Invert checkbox inverts the target parameter’s range, so that minimum becomes maximum and vice versa.
Use first parameter’s range
Use first parameter’s range makes the parameter display use the same range as the first parameter in the list. This is often useful if want to control the range in some way and apply the range to multiple parameters.
Transformation shape
Transformation shape button displays the graph editor, which lets you tweak the shape of the curve used to control the selected parameter. The X axis shows the original values, the Y axis defines the results. Note that this takes some CPU, therefore you have to enable it using the enable button in the title.
Restore original values when disabled
Restore original values when disabled makes the modulator restore the original parameter values when it is disabled by automation or modulation. Normally when you manually disable the modulator, the original values are restored as that is usually desired. However when you control the modulator enable state by automation or modulation, you may or may not want this to happen.
Assignable parameter ranges
Assignable parameter ranges allows you to assign parameter ranges of several first parameters to other subsystems such as multiparameters or modulators. By default it is disabled, which removes all the relevant parameters to save valuable resources. This feature is available only if automation compatibility mode for V10 is disabled.
Mode
Mode defines the way in which the modulator works. The modulator is like a black box that generates one number in range 0% to 100% at each moment and then assigns the appropriate value to each of the target parameters. The mode defines what this number will be. Select the particular tab to control the modulator’s behaviour. Normal mode uses a standard low-frequency oscillator (LFO) to drive the parameters. Follower mode uses the level of the input signal. Envelope generates an envelope using MIDI notes or by following input signal level. Random generates randomized output which is however the same every time you render the song. Pitch detects and follows the pitch of the input signal.
Normal mode

Normal mode makes the modulator work as a traditional low-frequency oscillator (LFO). Note that even if the modulator itself is running in a different mode, you can still blend this LFO using the LFO modulation parameter available on each tabbed page. The LFO parameters themselves are available on the first tabbed page only though.
Signal generator

Signal generator defines the modulation LFO shape. It is used by the LFO generator, but also for the Project feature.Si

References

• MeldaProduction
• MeldaProduction
• Linking External Controllers
• Installing Plugins
• Mixer Explained
• Plugin Wrapper
• Plugin Wrapper
• Tutorials: Multiparameters | MeldaProduction

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