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Emulating our Gold Hardware Standard
The alpha compressor plugin is the software version of our famous analog mastering compressor. It gives you a painstaking emulation of the sound and features the hardware is known and loved for. In cooperation with the code experts from Brainworx, our ultimate flagship was made available as a software plugin version.
The alpha compressor plugin license includes two separate versions optimized for specific tasks. This way, you can always choose the user interface that just works best for you.
The mastering version is the exact representation of the original hardware with all its bells and whistles, including Mid/Side operation, the Niveau Filter section, the Soft Clip limiter and the clever Signal Matrix.
The mixing version focuses on the actual compression section, and by doing so it gives you amazing results without having to spend much thought on the process. Actually, you might want to use this on every single track of your DAW.
Integrated M/S Matrix
M/S technology is commonly known as a variant of stereo microphoning. This technique uses a microphone with cardioid pattern for the middle signal (M) and another one with bi-directional pattern with an offset of 90° for the side signal (S). The main advantage of this technology is its mono compatibility. FM radio stations use M/S technology for transmitting stereo signals exactly for this reason.
To create M/S signals, the left and right channel of the stereo sum are added to generate the mid (M), whereas the side (S) is created by subtracting the right from the left channel:
M = L+R
S = L-R
To decode an M/S signal back into stereo again, M is added to S for the left channel and S is subtracted from M for the right channel:
L = M+S
R = M-S
The integration of an M/S encoder and decoder into a compressor generates new potentials that classic linked stereo compressors can hardly offer. One of the main advantages is the possibility to process the middle and side signals separately. This way you can make the center more compact without corrupting the original stereo spectrum, for example.
Of course it is also possible to enhance the presence of the side signals in an already finished mix. The stereo width can be influenced fast and effectively, too, and it is also possible to compress specific parts of a mix that could not be selected in a stereo mix as precisely as it is possible in M/S mode.
Feed Forward and Feedback
This function makes it possible to switch the feed of the sidechain alternatively behind (feedback) or in front (feed forward) of the actual compressor section.
It has an enormous effect on the character of the compressor: While processing in feedback mode is smooth and even, switching into feed forward mode will result in a clearly stronger and harder kind of compression.
From the technical point of view this function mainly influences the characteristic curve of the ratio value.
In feedback mode it goes up to a moderate ratio of 1:2.5. In contrast, the feed forward mode provides much higher values which also allow limiter settings and even negative ratios (i.e. loud signals will be reduced even more).
Small changes in dynamics can generate a high amount of gain reduction.
The values of the attack controller also change noticeably in feed forward mode, as they are almost twice as high as the values shown on the scale.
Attack and release are very crucial factors 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.
The separate Auto Fast for the release controller behaves in a similar way. The release also often forces the user to accept compromises when searching for the right setting. If it is set it too fast, distortion will occur, if it is too slow, drive and loudness are lost. In Auto Fast mode the compressor adapts to the currently right setting automatically.
This filter specialized in changing the overall sonic character of a track in fine nuances. It features two controllers per channel and is capable of flexibly producing convincing results 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 efficient and elegant solution.
Its main task is changing the proportions between high and low frequencies. It works like a pair of scales: Dependent on the gain setting around a selectable center frequency, the high frequencies are boosted up to +3 dB while the low frequencies are simultaneously attenuated by -5 dB maximum. Turning the gain controller into the other direction will cut the treble and boost the bass instead.
The filter type used for this application is an all pass variant with a flat frequency response that changes its phase according to the setting of the frequency controller. If the signal that went through the all pass is mixed to the original signal, everything in phase will be boosted while out-of-phase signals will be reduced.
Because the filtered signal is mixed to the original, the genuine structure and impulse-response remains almost completely intact. Boosting and cutting the selected frequency-areas at the same time makes it much easier to influence the character of a track (‘bright’ vs. ‘dark’) than with standard equalizers.
The sidechain filter allows frequency-dependent shaping of the compression process by giving specific frequency areas a stronger or weaker influence on the detection circuit.
If the SC gain controller is set to HP (High Pass), the filter will act like a 6 dB high pass and the reaction of the compressor on bass frequencies decreases. The setting LP (Low Pass) turns the filter into a 6 dB low pass and the compressor reacts primarily on low frequencies.
The combination of the sidechain filters, M/S matrix and different attack settings enables you to make very selective changes and – depending on the source material – even allows to process single instruments or voices in finished mixes.
Parallel compression, also known as ‘New York‘ compression, is a technique based on mixing a dry signal with a heavily compressed identical signal.
It is thought to maintain the subtleties of a performance while stabilizing the dynamics.
The mix controller of the alpha compressor makes it possible to cross-fade between the unprocessed and the compressed and filtered signals. This allows parallel compression right in the box and supersedes additional routings in favor of a better signal quality.
Now you can use even extreme compression settings without killing a track by winning the loudness war. By mixing just a part of the compressed signal to the original, the major portion of the initial dynamic structure remains intact.
Let’s have a look at the control logic of this interesting feature: If only the compressed button is pushed, you will hear the compressed signal only. Otherwise, if only the direct button is pushed, only the unprocessed signal routed from before the compressor section will be heard. If both buttons are activated, the mix controller will become active, and if none of the buttons is pushed, the channel will be muted.
In practice this lets you switch between unprocessed, compressed or mixed signals very fast without having to change the position of the mix controller.
The left and right channels (stereo mode) or the middle and side channels respectively (M/S mode) can be listened to separately – again you have the choice between the original, the compressed or the mixed signal. To mute the other channel, just deactivate its associated direct and compressed buttons.
Soft Clip Limiter
One of the biggest problems of digital audio technology is setting the right recording level on AD converters. The main challenge is that there is no level reserve beyond the maximum of 0 dBFS which could catch short peaks. In the moment a signal is digitally overdriven it is damaged irrevocably, because the original structure can hardly be reconstructed later.
Depending on the source signal, this type of peaks will produce audible distortion, as the signal is cut off and does not correspond to the original source anymore. In addition, this kind of clipping produces lots of new harmonics that do not always fit into the desired musical context.
The Soft Clip limiter has been developed to solve these specific problems. It is specialized in catching short and transient-like signals reliably. The technical principle is different from a classic ‘brickwall’ design which completely forbids further level increases beyond a certain threshold.
Instead, it is working similar to an analog tape machine driving loud impulses into saturation, acting like a ‘natural’ limiter. The Soft Clip limiter is based on discrete transistors, and just as with tape, its characteristic saturation curve results in rounding peaks instead of cutting them off. Especially when the source material contains variable peak values, the Soft Clip limiter comes in very handy.
Each channel of the original hardware alpha compressor features an additional transformer that can be switched into the signal chain after the mix stage. These are classic output transformers, but they are not used for balancing and galvanic isolation, but as an additional means of sound shaping.
We have decided to approach this in a little different way with the software version. Here you get something similar to the Warm function known from the museq: a slew rate limiter which alters the frequency spectrum, harmonics and transient response.
Because of the mastering approach of the alpha compressor, this is much more a subtle audio shaping feature than a glaring sound effect, but it’s certainly nice to add a little bit of color to a signal which might otherwise seem to be too clean.
Transferring a complex analog hardware into digital code is not exactly trivial, especially if the model is a completely discrete design like the alpha compressor.
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.
The alpha compressor 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 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 alpha compressor plugin uses oversampling according to the following rules:
- Project sample rate lower than 50 kHz: 4x oversamplin
- Project sample rate lower than 100 kHz: 2x oversampling
- Project sample rate higher than 100 kHz: no oversampling
You do not necessarily have to click and drag the controllers of the alpha compressor. Instead, try making your settings with the alternative mousewheel control without clicking on the specific controller first. The following shortcuts provide some further comfort:
VST: Shift + mouse wheel
AU: Shift + mouse wheel
RTAS/TDM: Ctrl/Cmd + mouse wheel
VST: Ctrl/Cmd + mouse click
AU: Alt + mouse click
RTAS/TDM: Alt + mouse click
“I must say wooow what a plugin… makes me wish I really had a chance to try the real unit… I still want the real hardware now more than ever… I must say I never wanted to purchase something so fast before :) Now where can I get the money for the hardware unit?”
“Today the probably most ultimate mastering compressor in the universe has been released in software: the alpha compressor! The elysia guys have done another great job after the surprising release of their creative compressor, the mpressor. This thing sounds divine…”
The alpha compressor plugin is available for MacOS and Windows in 32 and 64 bits. The following formats are supported: AAX DSP, AAX Native, AU, AAX AudioSuite, VST2, VST3 and UAD-2.