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Auto Fast
Originally developed for the alpha compressor, this switchable semi automation ensures a perfect attack on the basis of the value set by the user.

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The attack parameter is a very crucial factor for the operations of a compressor. Choosing the right time settings is very important, but depending on the dynamic progress of the source material this can be a difficult task – no matter if single tracks or complete mixes are processed.

If a very short attack time is chosen, the compressor is able to catch the short peaks, but on the other hand the sustaining signal will also be processed, which might result in audible distortion. Longer settings reduce distortion significantly, but then the compressor is too slow for catching fast impulses.

This is where the Auto Fast function comes into play. For example, if you set the attack to 80 ms and then engage the Auto Fast mode, the attack time will be shortened automatically on fast and loud signal impulses. The compressor reduces the signal quickly and prevents it from slipping through.

Then the attack time directly and automatically returns to its original setting. In Auto Fast mode the compressor can be very fast, but only when it is really needed. This function influences the attack parameter on short and loud impulses only; in all other cases the original setting of the controller has priority.

Anti Log
This alternative characteristic of the release curve follows an antilogarithmic course instead of the standard linear progress and produces a much more audible result.

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It is the time constants and especially the release parameter that decide if the processing of a compressor is obvious or unobtrusive to the ear. Mastering applications, for instance, require a discreet performance as a general rule. Here you will find mostly logarithmic or linear release curves.

It is characteristic of a logarithmic release that the time constant shortens when the amount of gain reduction increases. The advantage of this behavior is that short and loud peaks (e.g. drums) have a fast release time, while the remaining material is processed with a slower release time.

But if intentionally striking and creative compression is the goal, it definitely makes sense to turn things upside down. In the Anti Log mode of the mpressor the curve behaves just the other way round: If the threshold value is passed and compression starts, the release time will be longer at the beginning. If the input signal starts to decline, however, the release time will become faster as a result.

A special circuitry makes this behavior independent from the absolute amount of gain reduction. No matter if the compressor reduces 10, 15 or 20 dB, the curve will always stay the same at the beginning and it will only become faster at the end. With this feature you can create many exceptional compression effects just by the push of a button!

Negative Ratios
The characteristic curve bends and goes back down! Heavy pumping, backward sounds, etc. – perfect for very cool compression effects.

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Negative ratios – what exactly does this mean? To get a better understanding of this function, it makes sense to realize what the ratio control of a ‘normal’ compressor does:

• 1:1 The signal remains linear, there is no compression going on.
• 1:2 After crossing the threshold, an increase of 2 dB at the input will be compressed to an increase of 1 dB at the output.
• 1:∞ After crossing the threshold, the output signal is constantly held at the threshold level without reacting to further increases at the input (limiter).

At a negative ratio, the characteristic curve bends and returns back down after crossing the threshold. The louder the input signal, the lower the output signal – perfect for groovy compression effects. To get a grip on the extreme ‘destruction’ this can cause, engaging the Gain Reduction Limiter is just the right idea.

Beyond infinity – made possible by the mpressor ;-)

Audio Filter
The mpressor plugin of course also features another popular function of the hardware version: the easy yet very flexible Niveau Filter.

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This filter is a specialist in changing the overall sonic character of a signal with ease. It features two controllers per channel and is capable of producing convincing results flexibly in no time at all. Whenever a classic shelving filter would be too limited and a fully parametric filter would be too much, the Niveau Filter is the perfect tool.

Its main function is to change the proportions between high and low frequencies. The principle is quite similar to a pair of scales: Dependent on the gain setting around a variable center frequency, the high frequencies are boosted whereas the low frequencies are attenuated (or vice versa).

By simultaneously boosting and cutting the selected frequency areas, it is much easier to influence the character of a track (‚bright‘ vs. ‚dark‘) compared to using other types of equalizers. The center frequency can be shifted continuously between 26 Hz and 2.2 kHz or between 260 Hz and 22 kHz respectively (when the x10 switch is activated).

The characteristics of the filter change in the extreme positions of the EQ Gain controller: the fully counter-clockwise setting will give you a low pass filter; fully clockwise position will result in a high pass filter. The overall level can drop quite noticeably then, but this can be corrected with the gain controller easily.

External Sidechain
You want to create frequency dependent compression or have it accented by the groove? The sidechain function makes it all possible!

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The external sidechain enables the compressor to control its processing totally independent from the audio material running through it. If the SC Extern switch is active, compression will not be controlled by the signals from the regular audio inputs anymore, but by different signals which are fed into the additional sidechain input connectors.

If, for example, a duplicate of the input signal is processed with an equalizer and then fed into the sidechain input, the result will be frequency-dependent compression. Another example is to send the bass drum of a drum machine into the sidechain input in order to achieve nice groovy compression that is pumping in time with the music.

The creative options are almost infinite. Compression can be exactly on time or totally against it, which can of course be varied on the fly. Single instruments can be given more space in a mix according to its rhythm. All of a sudden, static sounds become vivid and sound really interesting!

Important: Your host software must support sidechain functionality in order to use this feature of the mpressor plugin!

Gain Reduction Limiter
This novel limiter is not placed in the audio path as usual, but restricts the virtual control voltage of the plugin instead.

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A specialty of the mpressor is the Gain Reduction Limiter for the virtual control voltage of the plugin. This limiter is not placed in the audio path where you would usually find it, but in the control path of the compressor. When it is activated, it limits the virtual control voltage according to the setting of the GR Limit controller. This means: No matter how high the input level might become – the amount of gain reduction will never exceed the value which you have set.

For comparison, imagine a fader on a mixing console with your hand moving the fader to ‘play compressor’. If now the fader was limited by a piece of duct tape at -10 dB, for example, it could only reduce the signal up to this value. If the input level dropped below this limit, the fader would be moved up correspondingly.

However, if the input signal got even louder, the fader could not be moved down any further because of the duct tape limit, and then the output signal would become louder again in correspondence with the input signal.

Loud parts in an arrangement can keep their dynamics, as they will not be compressed beyond the limit of the Gain Reduction Limiter. Some very nice special effects like ducking or upward compression can be achieved with it easily by only reducing the quieter parts without changing the original dynamics at the same time.

True Emulation
The compression behavior and the sound character of the hardware have been recreated in a complex process to match the analog model as close as possible.

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Transferring a complex analog hardware into digital code is not exactly trivial, especially if the model is a completely discrete design like the mpressor.

The first important task in a project like this is to fragment the electronic circuitry into separate functional blocks. These blocks are translated into software step by step after which they are united to become a functioning prototype.

This first result is measured very accurately and then compared to the hardware, which leads to an extensive and very detailed matching process. The work on the graphical user interface (photography, retouching, rendering) takes place at the same time.

The final stage is the calibration of the behavior of all the controllers in order to give the software the ‘feel’ of the real thing. Finally, the finished code is ported to different plugin interfaces (RTAS/VST/AU/TDM/AAX...) and packed into installation routines.

Oversampling
In order to achieve the best results also on lower sample rates, the mpressor plugin uses an increased internal resolution.

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The mpressor plugin benefits from higher sample rates in two ways: In the first place, it can react to changes in the source signal faster, which is especially important if a short attack time is set. Secondly, the generated virtual control voltage and therefore the compression behavior of the compressor becomes more precise because there are more measuring points available.

The mpressor plugin employs the oversampling technique in order to enjoy these advantages even if lower sample rates are used. This means that the basic sample rate of a project is multiplied by a certain factor inside the plugin without the need to set the complete project to a higher frequency.

This method consumes a certain amount of CPU power, but the acoustic result speaks for itself. The mpressor plugin uses oversampling according to the following rules:

• Project sample rate lower than 50 kHz: 4x oversampling
• Project sample rate lower than 100 kHz: 2x oversampling
• Project sample rate higher than 100 kHz: no oversampling

Mousewheel Support
Setting parameters with the mouse can be pretty annoying – for this reason the controllers of the mpressor can be moved with the mouse wheel easily.

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You do not necessarily have to click and drag the controllers of the mpressor. Instead, try making your settings with the alternative mousewheel control without clicking on the specific controller first. The following shortcuts provide some further comfort:

Fine mode
VST: Shift + mouse wheel
AU: Shift + mouse wheel
RTAS/TDM: Ctrl/Cmd + mouse wheel

Standard setting
VST: Ctrl/Cmd + mouse click
AU: Alt + mouse click
RTAS/TDM: Alt + mouse click

Linear/Circular
VST: Alt