MVocoder Audio Effect Plugin

Product Information

Specifications

• Product Name: MVocoder

• Control Buttons: Left Arrow, Right Arrow, Randomize, Panic,
  Settings, WWW

• Mouse Modifiers: Ctrl (constrains randomization), Alt (forces
  full randomization)

• Sleep Indicator: Indicates plugin activity status

Product Usage Instructions

Presets

The MVocoder plugin allows you to load presets using the
following buttons:

• Left Arrow Button: Loads the previous preset.
• Right Arrow Button: Loads the next preset.
• Randomize Button: Loads a random preset.

Randomize

The Randomize button generates random settings for the plugin.
However, the default randomization engine is designed to create
successful changes by learning from existing presets. To modify the
randomization behavior, you can use the following mouse
modifiers:

• No Modifier Keys: Smart randomization engine is used by
  default.

• Ctrl + Click: Constrain randomization to slight modifications
  of existing settings.

• Alt + Click: Force full randomization with extreme settings
  (not applicable to all parameters).

Panic

The Panic button is not described in the provided text
extract.

Settings

The Settings button opens a menu with additional plugin
settings. The menu includes the following items:

• Licence Manager: Activate/deactivate plugins and manage
  subscriptions.

• GUI & Style: Pick the GUI style and main colors for the
  plugin.

• Advanced Settings: Configure processing options for the
  plugin.

• Global System Settings: Change settings for all MeldaProduction
  plugins.

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

• Smart Interpolation: Adjust the interpolation algorithm for
  parameter value changes.

WWW

The WWW button provides access to additional information about
the plugin. The menu includes options for updates, support,
MeldaProduction web page, video tutorials, and social media
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

Q: How can I activate/deactivate plugins and manage

subscriptions?

A: Use the Licence Manager in the Settings menu to
activate/deactivate plugins and manage subscriptions. You can also
drag & drop a licence file onto the plugin for activation.

Q: How can I change the GUI style and colors of the

plugin?

A: In the GUI & Style section of the Settings menu, you can
pick the GUI style and main colors for the plugin, including the
background, title bars, text and graphs area, and highlighting.

Q: What do the Advanced Settings in the Settings menu do?

A: The Advanced Settings allow you to configure several
processing options for the plugin.

Q: How can I change settings for all MeldaProduction

plugins?

A: The Global System Settings in the Settings menu contain some
settings that affect all MeldaProduction plugins. Changing any of
these settings will require restarting your DAW to take effect.

Q: What does the Sleep Indicator mean?

A: The Sleep Indicator informs you whether the plugin is
currently active or in sleep mode. Sleep mode is activated when
there is no input signal and the plugin knows it cannot produce any
signal on its own. This feature helps save CPU. You can disable it
in the Settings menu.

MVocoder
MVocoder is an extremely advanced filter-based vocoder that you can use to generate a robotic voice, make your synth sing etc. Vocoding in general is a voice-modulated synth sound. It works like this: the main input, called the carrier, is filtered into small spectrum regions called bands. You usually use a synth with some rich harmonic content for this input. The side-chain input, called the modulator, is then filtered the same way (or in a different way if you desire). You usually use a voice here. The modulator bands are then processed through an envelope follower, such that the levels of the modulator bands are measured. The levels are then applied to the carrier bands, which are combined to produce the output. Simply put, only frequencies which exist in the modulator are preserved in the carrier. The technology was originally invented for compression in telephones, to save bandwidth and allow more calls to be carried simultaneously, hence the human voice has been the main interest. However it was soon adopted creatively for music, where you can generally use it for any type of signal. Please note however, that the carrier should contain rich harmonic content. It shouldn’t be a pure sine, for example, as in that case there would technically be only one frequency present and the chance that there would be a similar frequency in the modulator is quite slim, and in that case the output would be more or less silent. Besides standard vocoding as described, MVocoder also contains additional modes, which mingle the spectra of the carrier and modulator in different ways. An example would be morphing, which takes the 2 inputs and depending on the Ratio parameter keeps more of the one or the other.
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.
Globals panel

Globals panel contains global parameters such as vocoder mode, volume and panorama.

Volume
Volume defines the output volume of the processed signal. Range: silence to 12.0 dB, default 0.00 dB

Ratio
Ratio is an additional parameter for the operation performed on each band and its meaning depends on the Mode parameter. Range: 0.00 to 1.00, default 1.00

Panorama defines the output panorama of the processed signal. Range: 100% left to 100% right, default center

Panorama

Mode
Mode controls the actual operation performed on each band. Vocoder mode performs the typical vocoder processing – the carrier signal is multiplied by the envelope of the modulator. Ratio controls the amount of processed signal. Dual vocoder mode is similar to Vocoder mode, however Ratio controls amount of reverse-processed signal – modulator multiplied by the envelope of the carrier. Morph 1 and Morph 2 modes morph between the carrier and modulator signals in some way. The basic idea is to take the dominant signals from both of them. Ratio controls which one, carrier or modulator, is preferred. Ring modulation mode performs per-band ring modulation, which can often result in highly inharmonic sound and can be used to create special effects. Ratio controls the amount of the processed signal. Exciting mode amplifies the bands dominant in both carrier and modulator signals. Ratio controls the amount of processed signal. Inversion and Dual inversion modes are similar to Vocoder and Dual vocoder modes, but the envelopes are inverted, so instead of keeping the frequencies in carrier which are dominant in modulator they keep the frequencies which are not dominant in modulator.

Carrier & modulator panel

Carrier & modulator panel contains parameters controlling routing and direct carrier & modulator output.
Carrier volume
Carrier volume defines the output volume of the carrier signal (the main input by default). Range: silence to 0.00 dB, default silence
Modulator volume
Modulator volume defines the output volume of the modulator signal (the side- chain input by default). Range: silence to 0.00 dB, default silence
Carrier pan
Carrier pan defines the output panorama of the carrier signal (the main input by default). Range: 100% left to 100% right, default center
Modulator pan

Modulator pan defines the output panorama of the modulator signal (the side- chain input by default). Range: 100% left to 100% right, default center
Swap carrier and modulator
Swap carrier and modulator exchanges the carrier and modulator signals. By default the carrier (usually synth) is the main input and modulator (usually voice) is the side-chain.
LR encoding
LR encoding exchanges the standard side-chain input for input stereo – the left channel becomes the carrier, the right channel the modulator.
Effects panel
Effects panel contains parameters of some effects performed by the vocoder.
Whitening
Whitening controls the amount of special effect applied to the carrier signal, which basically amplifies missing frequencies according to the detector results, hence resulting in a richer sound. Range: 0.00% to 100.0%, default 0.00%
Formant shift
Formant shift lets you manually alter the formant information (in semitones), which generally results in no pitch shifting, but can create the mickey-mouse effect for example. Range: -12.00 to +12.00, default 0
Gate threshold
Gate threshold controls the per-band noise gate threshold. Range: silence to 12.0 dB, default silence
Gate ratio
Gate ratio controls the per-band noise gate ratio. The higher it is, the steeper and rougher the gate is. Range: 1.0 : 1 to 20.0 : 1, default 2.0 : 1
Saturation car
Saturation car controls the amount of saturation performed on the carrier signal. Saturation adds higher harmonics making the spectrum richer, which could be advantageous for vocoding. Range: 0.00% to 100.0%, default 0.00%
Saturation mod
Saturation mod controls the amount of saturation performed on the modulator signal. Saturation adds higher harmonics making the spectrum richer, which could be advantageous for vocoding. Range: 0.00% to 100.0%, default 0.00%

Detector panel
Detector panel contains envelope detector parameters.
Attack
Attack defines the attack time, that is how quickly the level detector increases the measured input level. When the input peak level is higher than the current level measured by the detector, the detector moves into the attack mode, in which the measured level is increased depending on the input signal. The higher the input signal, or the shorter the attack time, the faster the measured level rises. Once the measured level exceeds the Threshold then the dynamics processing (compression, limiting, gating) will start. There must be a reasonable balance between attack and release times. If the attack is too long compared to the release, the detector will tend to keep the measured level low, because the release would cause that level to fall too quickly. In most cases you may expect the attack time to be shorter than the release time. To understand the working of a level detector, it is best to cover the typical cases: In a compressor the attack time controls how quickly the measured level moves above the threshold and the processor begins compressing. As a result, a very short attack time will compress even the beginning transient of a snare drum for example, hence it would remove the punch. With a very long attack time the measured level may not even reach the threshold, so the compressor may not do anything. In a limiter the attack becomes a very sensitive control, defining how much of the signal is limited and how much of it becomes saturated/clipped. If the attack time is very short, limiting starts very quickly and the limiter catches most peaks itself and reduces them, providing lower distortion, but can cause pumping. On the other hand, a higher attack setting (typically above 1ms) will let most peaks through the limiter to the subsequent in-built clipper or saturator, which causes more distortion of the initial transient, but less pumping. In a gate the situation is similar to a compressor – the attack time controls how quickly the measured level can rise above the threshold at which point the gate opens. In this case you will usually need very low attack times, so that the gate reacts quickly enough. The inevitable distortion can then be avoided using look-ahead and hold parameters. In a modulator, the detector is driving other parameters, a filter cut-off frequency for example, and the situation really depends on the target. If you want the detector to react quickly on the input level rising, use a shorter attack time; if you want it to follow the flow of the input signal slowly, use longer attack and release times. Range: 0 ms to 1000 ms, default 1.00 ms
Release
Release defines the release time, that is how quickly the level detector decreases the measured input level. The shorter the release time, the faster the response is. Once the attack stage has been completed, when the input peak level is lower than the current level measured by the detector, the detector moves into the release mode, in which the measured level is decreased depending on the input signal. The lower the input signal, or the shorter the release time, the faster the measured level drops. Once the measured level falls under the Threshold then the dynamics processing (compression, limiting, gating) will stop. There must be a reasonable balance between attack and release times. If the attack is too long compared to release, the detector would tend to keep the level low, because release would cause the level to fall too quickly. Hence in most cases you may expect the attack time to be shorter than the release time. To understand the working of a level detector, it is best to cover the typical cases: In a compressor the release time controls how quickly the measured level falls below the threshold and the compression stops. As a result a very short release time makes the compressor stop quickly, for example, leaving the sustain of a snare drum intact. On the other hand, a very long release keeps the compression working longer, hence it is useful to stabilize the levels. In a limiter the release time keeps the measured level above the limiter threshold causing the gain reduction. Having a very long release time in this case doesn’t make sense as the limiter would be working continuously and the effect would be more or less the same as simply decreasing the input gain manually. However too short a release time lets the limiter stop too quickly, which usually

causes distortion as the peaks through the limiter to the subsequent in-built clipper or saturator. Hence release time is used to avoid distortion at the expense of decreasing the output level. In a gate the situation is similar to a compressor – the release time controls how quickly the measured level can fall below the threshold at which point the gate closes. Having a longer release time in a gate is a perfectly acceptable option. The release time will basically control how much of the sound’s sustain will pass. In a modulator, the detector is driving other parameters, a filter cut-off frequency for example, and the situation really depends on the target. If you want the detector to react quickly on the input level falling, use a shorter release time; if you want it to follow the flow of the input signal slowly, use longer attack and release times. Range: 0 ms to 1000 ms, default 10 ms
Peak hold
Peak hold defines the time that signal level detector holds its maximum before the release stage is allowed to start. As an example, you can imagine that when an attack stage ends there can be an additional peak hold stage and the level is not yet falling, before the release stage starts. This is true only when true peak mode is enabled (check the advanced detector settings if available). It is often used in gates to avoid the gated level falling below the threshold too quickly, while having short release times. If you want the gate to close quickly, you need a short release time. But in that case the ending may be too abrupt and even cause some distortion. So you use the peak hold to delay the release stage. It is also used along with look-ahead to avoid distortion in limiters and compressors. If you need a very short attack, the attack stage may be too quick and cause distortions. In limiters this attack time is often 0ms, in which case it becomes a clipper. Setting lookahead and peak hold to the same value will make the detector move ahead in time, so that it can react to attack stages before they actually occur and yet hold the levels for the actual signal to come. Range: 0 ms to 1000 ms, default 1.00 ms
RMS length
RMS length smoothes out the values of the input levels (not the input itself), such that the level detector receives the pre-processed signal without so many fluctuations. When set to its minimum value the detector becomes a so-called “peak detector”, otherwise it is an “RMS detector”. When you look at a typical waveform in any editor, you can see that the signal is constantly changing and contains various transient bursts and separate peaks. This is especially noticeable with rhythmical signals, such as drums. Trying to imagine how a typical attack/release detector works with such a wild signal may be complex, at least. RMS essentially takes the surrounding samples and averages them. The result is a much smoother signal with fewer individual peaks and short noise bursts. RMS length controls how many samples are taken to calculate the average. It stabilizes the levels, but it also causes a slower response time. As such it is great for mastering, when you want to lower the dynamic range in a very subtle way without any instabilities. However, it is not really desirable for processing drums, for example, where the transient bursts may actually be individual drum hits, hence it is usually recommended to use peak detectors for percussive instruments. Note that the RMS detector has 2 modes – a simplified approximation is used by default, and a true RMS is processor can be enabled from the advanced settings (if provided). Both respond differently, neither of them is better than the other, they are simply different. Range: Peak to 100 ms, default 1.0 ms
Filters panel
Filters panel contains advanced parameters regarding the filters being used by the vocoder.
Bands
Bands controls the number of bands used by the vocoder. More bands usually provide higher clarity and higher CPU usage. Range: 4 to 100, default 32
Car resonance
Car resonance controls the filter resonance applied to the carrier signal. A low resonance results in a fuller sound, while a higher

resonance ends in a thinner sound. Range: 0.00% to 100.0%, default 0.00%
Mod resonance
Mod resonance controls the filter resonance applied to the modulator signal. A low resonance results in a fuller sound, while a higher resonance ends in a thinner sound. Range: 0.00% to 100.0%, default 0.00%
Car order
Car order controls the filter order applied to the carrier signal. Higher order provides higher clarity and higher CPU usage. Range: 1 to 10, default 3
Mod order
Mod order controls the filter order applied to the modulator signal. Higher order provides higher clarity and higher CPU usage. Range: 1 to 10, default 2
Matrix
Matrix button displays a band matrix, which controls how the bands affect each other.
Band matrix
Band matrix controls how the bands affect each other. By default the graph is a diagonal line from top left to bottom right, which means that modulator band 1 affects carrier band 1, modulator band 2 affects carrier band 2 etc. However you may set this up differently and you may even let some carrier bands be affected by multiple modulator bands, or none of them.

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.
Graphs
Graphs button displays a set of graphs, which you can use to control features of each band, such as the distribution of the bands along the frequency axis, resonances, dynamic properties etc.
Band graphs window
Band graphs window provides a set of graphs, which you can use to control features of each band, such as the distribution of the bands along the frequency axis, resonances, dynamic properties etc.

Equalizer
Equalizer contains volumes for each band.
Collapse
Collapse button minimizes or enlarges the panel to release space for other editors.

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.
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.
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
Collapse button minimizes or enlarges the panel to release space for other editors.
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.
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.
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.Signalgenerator is an incredibly versatile generator of low & high frequency signals. It offers 2 distinct modes – Normal and Harmonics. Normal mode is appropriate for low-frequency oscillators, where the graphical shape is relevant and is used to drive some form of modulation. For example, a tremolo uses this modulation to change the actual signal level in time. Frequencies for such oscillators usually do not exceed 20Hz as this is a sort of limit above which the frequencies become audible. Harmonics mode is designed for high-frequency oscillators, where the actual shape is not as important as the harmonic content of the resulting signal, hence it is especially useful for actual audio signals. Please note that since a shape can contain more harmonics than those available from the harmonic generator, the results may not be exactly the same. As an example, a rectangular wave in normal mode may sound fuller than when converted to the harmonic mode.
Use the arrow-down button to switch from normal mode to harmonics mode or click the Normal and Harmonics buttons
Normal mode
The generator first uses a set of predefined signal shapes (sine, triangle, rectangle…), which you can select directly by right-clicking on the editor and choosing the requested shape from the menu. This menu also provides a link to the modulator shapes preset manager, normalization and randomization. You can also use the Main shape parameter, which generates a combination of adjacent signals to provide a nearly inexhaustible number of basic shapes.
The engine then combines the predefined shape with a Custom shape, which may be anything you can draw using the advanced envelope engine, depending on the level set by the Custom shape control. Use the Edit button to edit the custom shape.
You can also combine those results with a fully featured step sequencer, with variable number of steps and several shapes for each of them, depending on the level set by the Step sequencer control. Use the lower Edit button to edit the step sequence.
Those results may be mixed with a custom sample, which is available from the advanced settings, accessed by clicking the Advanced button.
Smoothness softens any abrupt edges, generated by the step sequencer for example.
Finally there are Advanced features providing more complex transformations, adding harmonics etc. or you can click the Randomize button in the top-left corner to generate a random, but reasonable, modulator shape.
Harmonics mode
Harmonics mode represents the signal as a series of harmonics (that is, multiples of the base frequency). For example, when your oscillator has a frequency of 2Hz (set in the Rate panel), then the harmonics are 2Hz, 4Hz, 6Hz, 8Hz etc. In theory, any signal can be created by mixing a potentially infinite number of these harmonics.
The harmonics mode lets you control the levels and phases of each harmonic. The top graph controls the levels of individual harmonics, while the bottom one controls their phases. Use the left-mouse button to change the values in each graph, the rightmouse button sets the default for the harmonics – 0% level and 0% phase. In both graphs the harmonics of power 2 (that is octaves) are highlighted. Other harmonics may actually sound disharmonic, despite their names.
For example, if you reset all harmonics to the defaults and increase only the first one, you will get a simple sine wave. By adding further harmonics you make the output signal more complex.

Harmonics controls the number of generated harmonics. The higher the number is, the richer the output signal is (unless the levels are 0% of course). This is useful to make the sound cleaner. For example, if you transform a saw-tooth wave to harmonics, it would not sound like a typical saw-tooth wave anymore, but more like a low-passed version of one. The more harmonics you use, the closer you get to the original saw-tooth wave.
Generator is a powerful tool for generating the harmonics, which are otherwise rather clumsy to edit. The generator provides several parameters based upon which it creates the entire series of harmonic levels and phases. These parameters are usually easier to understand than the harmonics themselves. Part of the generator is the randomizer available via the Random seed button, which smartly generates random settings for the generator. This makes the process of getting new sounds as simple as possible.
Signal generation fundamentals
The signal generator produces a periodic signal with specified wave shape. This means that the signal is repeating over and over again. As a result it can only contain multiples of the fundamental frequency. For example, if the generator is producing 100Hz signal, then it can contain 100Hz (fundamental or 1st harmonic), 200Hz (2nd harmonic), 300Hz (3rd harmonic), 400Hz (4th harmonic) etc. However, it can never produce 110Hz. You can then control the level of each harmonic and their relative phases. It does not matter whether you use the normal mode using oscillator shapes, or harmonics mode where you can control the harmonics directly. If both modes result in the same wave shape (such as sine wave vs. 1st harmonic only), then the result is exactly the same.
Sine wave is the simplest of all as it contains the fundamental frequency only. The “sharper” the signal shape is, the more harmonics it contains. The biggest source of higher harmonics is a “discontinuity”, which you can see in both rectangle and saw waves. In theory, these signals have an infinite number of harmonics. However since our hearing is highly limited to less than 20kHz, the number of harmonics which are relevant is actually pretty small. If you generate a 50Hz signal, which is very low, and assuming that you have extremely good ears and you actually hear 20kHz, then the number of harmonics audible for you is 20000 / 50 = 400.
What happens above 20kHz?
Consider the example above again, what happens with harmonics above 400? These either stay there and simply are not audible, disappear if anti-aliasing is used, or get aliased back under 20kHz in which case you get the typical digital dirt.
When you convert a rectangle wave to harmonics mode, only the first 256 harmonics are used, so it basically works like an infinitely steep low-pass filter. What is the limit then? 50 Hz * 256 = 12.8kHz. The harmonic mode will not produce anything above this limit if you are generating a 50Hz signal. Most people do not hear anything above 15kHz, so this is usually enough, but if not, you may need to use the normal mode where you get the “infinite” number of harmonics.
What you see is not always what you get!
Say you want a rectangle wave and play a 440Hz tone(A4). You would expect the output signal to be a really quick rectangle wave, right? Wrong! If you would do that, and actually most synthesizers on the market do that, you would get the infinite number of harmonics. And, since you are working in say 48kHz sampling rate, the maximum frequency that can actually exist in your signal is 24kHz. So everything above it would get aliased below 24kHz, and there would be a lot of aliased dirt.
The “good” synthesizers perform a so-called anti-aliasing. There are several methods, most of them require quite a lot of CPU or have other limitations. The goal is to remove all frequencies above the 24kHz in our case or in reality, it is more about removing all aliased frequencies above 20kHz – this means, that we do not care about frequencies above 20kHz, because we do not hear them anyway. But we will keep it simple. Let’s say we remove everything above 20kHz. You already know that the rectangle wave can be created using an infinite number of harmonics or sine waves. We removed everything above the 45th harmonic (20000 / 440) so our rectangle wave is trying to be formed using just 45 harmonics, so it will not really look like a rectangle wave.
After some additional filtering (like DC removal), the rectangle wave may look completely different than a true rectangle wave, yet it would sound the same! Does it matter? Not really. You simply edit the shape as a rectangle wave and let the synthesizer do the ugly stuff for you. But do not check the output, because it may be very different than what you would expect ;).
How can I generate non-harmonic frequencies?
Ok, so now you are playing a 440Hz (A4) saw wave, it contains 440Hz, 880Hz, 1320Hz etc. Anything generated using the signal generator can contain only these frequencies, the only difference is the levels and phases of each of them. What if you want to make the signal dirty by adding say 500Hz? Well, that is not that simple! Here we are getting into audio synthesizer stuff, so let us just give you a few hints.
The traditional way is to use modulation. One particular method is called frequency modulation (FM). Instead of generating a 440Hz saw wave with your generator, you change the pitch, up and down. You are modulating the frequency, that’s why FM. It is basically a vibrato, but as you increase the speed of the vibrato, it gets so quick that you stop noticing the pitch changes (that’s very simplified but it serves the purpose) and instead it starts producing a very complex spectrum. Will the 500Hz be there? Well, if setup correctly, yes, but there will also be lots of other non-harmonic frequencies.
Another way is possible without any other tools. Let’s say you do not want 440Hz, but 660Hz. Then you may generate 220Hz instead of 440Hz (which is one octave below it) and voila, 660Hz is the 3rd harmonic (3 x 220 is 660)! But you need to shift the saw wave one octave above. Fortunately it is not that hard here – go to the normal mode, select saw tooth, click advanced, and use the harmonics panel to remove the fundamental and leave just the 2nd harmonic, then convert it to harmonic mode. Well, it’s not that

hard, but it’s not exactly simple either… The only way is, of course, additive synthesis. In that case you do not use one oscillator, but many of them. It lets you generate just about anything. But there is a catch, actually many of them. First, you need to say “ok I want this frequency and that frequency…”, the setup is actually infinitely hard as there may be an infinite number of frequencies :). And the second is, of course, CPU requirements. So is there some ultimate solution? Nope, sorry. The good thing is, you will not probably need it, because while what you see is not always what you get, also what you want is often not what you really want to hear :).
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.
Normal
Normal button switches the generator into the normal mode, which lets you edit the shape of the oscillator. This is especially advantageous for low-frequency oscillators, where the shape matters even though it doesn’t have any physical meaning.
Convert
Convert button converts the current shape into harmonic-based representation. Please note that since the number of harmonics is limited, the result will not perfectly resemble the original shape.
Harmonics
Harmonics button switches the generator into the harmonics mode, which lets you edit the levels and phases of individual harmonics. This is especially advantageous for high-frequency oscillators, hence sound generators.
Signal generator in Normal mode

Signal generator in Normal mode works by generating the oscillator shape using a combination of several curves – a predefined set of standard curves, custom shape, step sequencer and custom sample. It also post-processes the shape using several filters including smoothing to custom transformations. This is especially useful when using the oscillator as an LFO (low- frequencyoscillator), where the harmonic contents does not really matter, but the shape does.
Shape
Shape controls the main shape used by the signal generator. There are several predefined shapes: exponential, triangle, sine power 8, sine power 4, sine square, sine, harmonics, more harmonics, disharmonics, sine square root, sine 4 root, rectangle, rect-saw, saw, noise and mess. You can choose any of them or interpolate between any 2 adjacent shapes using this control.
Custom
Custom controls the amount of the custom shape that is blended into the main shape.
Edit
Edit button shows the custom shape editor.
Signal generator custom shape editor

Signal generator custom shape editor controls the custom shape. You can edit virtually any shape that you can imagine and then blend it with the standard shapes, the step sequencer 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.
Copy
Copy button copies the settings onto the system clipboard.
Paste
Paste button loads the settings from the system clipboard.

Graph editor lets you edit the envelope graph.

Graph editor

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.

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 button generates the envelope from step sequencer.

Step sequencer

Clear points button deletes all points.
Distribute points
Distribute points button makes all points equally spaced.

Clear points

Randomize
Randomize button slightly modifies the Y coordinates.
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.
Step
Step controls the amount of the step sequencer shape that is blended into the main shape (which has already been blended with the custom shape).
Edit
Edit button shows the step sequencer editor.
Signal generator step sequencer editor

Signal generator step sequencer editor controls the step sequencer shape. You can have various numbers of steps each with a different value and shape. Note that for classic rectangular shapes the output can be very rough, hence it may be worth considering using Smoothness parameter to smooth out the resulting shape. This will use additional CPU power of course, but that should be negligible unless you modulate any of the signal generator parameters.
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.
Random values

Random values button generates random sequence of values, but keeps the shape of each step.
Random shapes
Random shapes button generates random sequence of shapes, but keeps the values of each step.
Smooth
Smooth controls the amount of smoothing. Many shapes, especially those produced by the step sequencer, have rough jagged edges, which may be advantageous, but when used to modulate certain parameters, the output may be clicking or causing other artifacts. Smoothness helps it by smoothing the whole signal shape out and removing these rough edges.
Advanced
Advanced button displays an additional window with more advanced settings for post-processing the signal shape, such as harmonics or custom transformations.
Advanced 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.
Random
Random button generates random settings using the existing presets.
Settings panel
Settings panel contains some global settings of the oscillator.
Normalize
Normalize switch enables normalization to -1..+1. It is generally desirable since even if you draw a custom shape, you usually want it to have the full range. You may want to disable it if you want to create some custom shapes, where the level actually matters.
Invert
Invert switch simply inverts the output shape vertically.
Enable crossfading
Enable crossfading enables interpolation between shapes when the shape is changing. This requires more CPU, but can avoid zipper noise when the shape is being modulated for example.
Show position
Show position makes the editor display a position indicator.
Interpolate between 1 and 0
Interpolate between 1 and 0 smoothens the discontinuity between 1 and 0 values, which is inevitable for shapes such as saw or rect for example. However when this is a high frequency oscillator (HFO), this discontinuity is what creates the highest frequencies, so it is actually desirable. When using it as an LFO, you may also want the discontinuity in some extreme cases.
Custom sample panel
Custom sample panel contains parameters of the custom sample that you can load and mix with the other sources. Do NOT confuse this with a sampler, the custom sample is taken as one period of the waveform. It can be used for creative effects and it can be used to import a custom waveform. The custom sample is then stored with limited precision within the settings, so the sample does not need to be kept on the system, but note that these settings may be quite large. To limit the space required by the settings, the sample is stored only if the depth is not 0%, meaning only if the sample is actually used.

Depth
Depth controls the amount of custom sample mix. 0% means the sample is not used even if there actually is one loaded. 100% means the sample completely overrides the basic shape, custom shape, step sequencer… However, transformations are still performed on the sample.
Load sample
Load sample button displays a file selection window, which lets you select the custom sample file.
Clear sample
Clear sample button removes the custom sample if it has been loaded.
Shape panel
Shape panel contains parameters performing various transformations on the signal shape. Please note that most transformation require a significant amount of CPU resources, so you should not automate or modulate the signal shape if you are using them.
Harmonics panel
Harmonics panel lets you add separate harmonics of the original signal.
Post-processing panel
Post-processing panel lets you post-process the shape after all the previous generator items.
Transformations

Shape transformation graph
Shape transformation graph lets you perform arbitrary modification of the graph shape. Basically this graph lets you modify the shape “in time”. The Y axis represents the position in the source signal related to the position in the target signal. The best way to check what it does is simply to try it.
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.

Amplitude transformation graph
Amplitude transformation graph lets you perform arbitrary modification of the graph amplitude. Basically this graph lets you modify the shape’s level, vertical axis. The X axis represents the original values, the Y axis defines the resulting values. The best way to check what it does is simply to try it.
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.
Assignable advanced shape parameters
Assignable advanced shape parameters allows you to assign advanced parameters such as step sequencer values to other subsystems such as multiparameters or modulators. By default it is disabled, which removes all the relevant parameters to save valuable resources.
Signal generator in Harmonics mode

Signal generator in Harmonics mode works by generating the oscillator shape using individual harmonics. Essentially a harmonic is a sine wave. The first harmonic, known as the fundamental, fits once in the oscillator time period, hence it is the same as selecting sine wave in the Normal mode. The second harmonic fits twice, the third three times etc. In theory, any shape you create in normal mode can be converted into harmonics. However, this approach to signal generation needs an enormous number of harmonics, which is both inefficient to calculate and mostly hard to edit. Therefore, the harmonic mode can process up to 256 harmonics, which is enough for very complex spectrums, however it is still not enough to generate an accurate square wave for example. If your goal is to create basic shapes, it is better to use the normal mode. It is nearly impossible to say how a particular curve will sound when used as a high-frequency oscillator in a synthesizer, just by looking at its shape. Harmonics mode, on the other hand, is directly related to human hearing and makes this process very simple. In general, the more harmonics you add, the richer the sound will be. The higher the harmonic, the higher the tone. Usually, one leaves the first harmonic enabled too, as this is the fundamental tone, however you may experiment with more dissonant sounds without it. Editing harmonics can be time consuming unless you hear what you want, so a signal generator is also available. This great tool lets you generate a random spectrum by a single click. You can also open the Generator settings and edit its parameters, which basically control the audio properties in a more natural way – using parameters such as complexity, harmonicity etc.
Generator
Generator button shows a powerful harmonics generator, which can create unlimited number of various timbres and even analyze a sample and extract harmonics from it.
Harmonics generator

Harmonics generator is a powerful tool, that can generate various harmonics- based timbres and even analyze a sample file and extract harmonics from it.
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.
Generator panel
Generator panel contains parameters of the harmonics generator. By changing any of the parameters, the harmonics are changed, however only Random seed button changes the structure completely. The other parameters can be used to tweak the results.
Harmonicity
Harmonicity controls the ratio between natural harmonics and those which sound disharmonic (despite the title “harmonics”). Assuming that the 1st harmonic is the fundamental, 2nd harmonic is 1 octave above, 4th is 2 octaves above, both can be considered very natural. 3rd harmonic is 1 octave and a 5th above the fundamental, and is still pretty harmonic, but less than the octaves. 5th harmonic is 2 octaves and a major 3rd above the fundamental. Such a tone may sound very disharmonic, in minor scales for example. Higher harmonics are often very disharmonic and produce typical ringing timbres. When harmonicity parameter is set to 100%, only octaves are allowed. By lowering the value more and more disharmonics are created and with 0% all frequencies are allowed. For values below 0% disharmonics are preferred, hence you can expect more ringing timbres.
Slope
Slope defines the amount of higher harmonics compared to lower ones. When 0%, the higher harmonics have the same levels as lower ones. Typically you use values below 0%, which attenuates the higher harmonics making the resulting sound darker. Similarly values above 0% make the sound brighter.

Fullness
Fullness controls the number of generated harmonics. With values around 0% the resulting timbers will contain only a few harmonics making the sound clear. Higher values increase number of harmonics making the timbre rich.
Fundamental
Fundamental controls the minimum level of the fundamental (the 1st harmonic). Most sounds have a very strong fundamental as it carries the pitch.
Random seed
Random seed button generates a new series of harmonics. Pressing this button will create a whole new timbre.
Post-processor panel
Post-processor panel contains parameters of the harmonics post-processor. The generator and sample analyzer first create a series of harmonics, the timbre. These harmonics are mixed depending on the Sample ratio parameter. After that the postprocessor is engaged, which can further transform the harmonics in several ways.
Sharpen
Sharpen is a sort of soft compression/expanding unit. Values below 0% decrease the level of quiet harmonics, while values above 0% increase their level.
Noise
Noise defines amount of noise added to the timbre. Noise can make the results dirty providing much richer timbres.
Clean
Clean controls the threshold of a gate. It basically attenuates or removes harmonics below this level making the output cleaner.
Compress
Compress reduces the dynamic range of the harmonics by increasing levels of the quiet ones, but keeping the levels of the loud ones.
Harmonize
Harmonize creates additional higher harmonics from existing ones. This is especially useful to transform rich dirty disharmonic timbres into similarly rich but more harmonic timbres.
Sample analyzer panel

Sample analyzer panel contains parameters of the sample analyzer. If there is no sample loaded, the sample analyzer is turned off. The analyzer takes the selected sample and a position within it, analyses one period of the signal waveform and produces the output set of harmonics. You can then combine these harmonics with the output of the generator using Sample ratio parameter. The sample itself is not store with the plugin settings. Instead the path to the target sample file is stored along with the analyzed harmonics. If the sample file is not available, you cannot modify the analysis parameters and the last analyzed harmonics are used. This means that you actually don’t need to have the sample file available on the computer on which you are using the settings.

Load file button lets you select a sample file to analyse.

Load file

Randomize
Randomize button selects random parameters for the harmonics generator, so you can use it to get a random sound character instantly. Hold Ctrl to slightly modify existing generator settings instead of completely changing them.

Magnitudes graph
Magnitudes graph contains the levels of the individual harmonics. The highlighted bars are octaves, thus the 1st, 2nd, 4th, 8th harmonic etc.

Phases graph
Phases graph contains the phases of the individual harmonics. The highlighted bars are octaves, thus the 1st, 2nd, 4th, 8th harmonic etc.
Rate panel

Rate panel contains parameters controlling the speed of the LFO, whether the modulator is set to Normal mode or any other mode while the LFO modulation is used.

Sync
Sync switch turns the modulator into synced mode, where its speed is not defined by frequency, but it uses musical units instead.
Frequency
Frequency defines the modulation speed.
Sync group
Sync group lets you synchronize the modulators with each other and potentially with other parts of the plugin. It can be controlled only when to-host synchronization is disabled, otherwise it is overridden by synchronization from the host. By using the same synchronization group for all modulators you ensure they will always be in-sync even though no other synchronization is used. This can be useful, for example, when you want to modulate different parameters with different shapes or when using some more advanced method, such as using a follower. When the synchronization is enabled, it works on the ‘first is the leader’ basis, hence the first modulator controls the rest of the modulators in the same group.
Synchronization panel

Synchronization panel contains parameters for the to-host synchronization.

Length defines the note length to be used.

Length

Type
Type defines the note type, such as straight notes or triplets, to be used. Together the Length and Type determine the actual time/delay. Example: ‘1/4 Straight’ at 120 bpm = a delay of 500 ms, ‘1/4 Triplet’ at 160 bpm = a delay of 281.25 ms.

Phase
Phase defines the phase offset of the to-host synchronization. Range: 0° (0%) to 360° (100.0%), default 90° (25.0%)

Count
Count defines the number of the units, hence multiplies of the sync length. Range: 1 to 64, default 1

Set frequency
Set frequency button sets the Frequency parameter available for the frequency mode so that it matches the current synchronization. That way you can set the modulator’s frequency to the current synchronization and then change it a little for example.

MIDI reset panel

MIDI reset panel configures the MIDI reset feature, which will reset the oscillator when a MIDI note is received or its MIDI reset parameter is a target of another modulator or multiparameter. This way you can make the oscillator perform “in-sync” with your playing. Please note t

References

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

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